diff --git a/CHANGELOG.md b/CHANGELOG.md
new file mode 100644
index 0000000000000000000000000000000000000000..dc04f7720588c21499229f608acd815fe6c71aca
--- /dev/null
+++ b/CHANGELOG.md
@@ -0,0 +1,33 @@
+# Changelog
+All notable changes to this project will be documented in this file.
+
+The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
+and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
+
+# Preamble
+
+This changelog file was started on October 2019. Previous changes were more or less tracked in the *NEWS* file.
+
+## [Unreleased]
+
+### Added
+- RTP bundle mode feature according to https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54 .
+
+### Changed
+- Big internal refactoring of how streams are managed within offer/answer exchanges.
+- ICE now uses all IP addresses detected on the host.
+- Better handling of parameter changes in streams during the session, which avoids unecessary restarts.
+
+### Fixed
+- Internal refactoring of management of locally played tones, in order to fix race conditions.
+
+
+## [4.3.0] - 2019-10-14
+
+### Added
+- New cmake options to make "small" builds of liblinphone, by excluding adavanced IM and DB storage.
+
+### Changed
+- Optimisations in chatrooms loading from Sqlite DB, improving startup time.
+- License changed to GNU GPLv3.
+
diff --git a/CMakeLists.txt b/CMakeLists.txt
index 8053fadbaa489e11c2dd025c231a2c627bc772fc..6ad2adde083361593b2fec4c4cfac58f628662e7 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -97,6 +97,8 @@ include(CMakePushCheckState)
include(GNUInstallDirs)
include(CheckCXXCompilerFlag)
+check_symbol_exists(getifaddrs "sys/types.h;ifaddrs.h" HAVE_GETIFADDRS)
+
if(NOT CMAKE_INSTALL_RPATH AND CMAKE_INSTALL_PREFIX)
set(CMAKE_INSTALL_RPATH ${CMAKE_INSTALL_FULL_LIBDIR})
message(STATUS "Setting install rpath to ${CMAKE_INSTALL_RPATH}")
@@ -284,10 +286,8 @@ if(MSVC)
else()
list(APPEND STRICT_OPTIONS_CPP
"-Wall"
- "-Wcast-align"
"-Wconversion"
"-Werror=return-type"
- "-Wfloat-equal"
"-Winit-self"
"-Wno-error=deprecated-declarations"
"-Wpointer-arith"
diff --git a/ChangeLog.md b/ChangeLog.md
deleted file mode 100644
index aaec8bb0f71d9203b052a3fd68e840a8e222efbf..0000000000000000000000000000000000000000
--- a/ChangeLog.md
+++ /dev/null
@@ -1,25 +0,0 @@
-# Change Log
-All notable changes to this project will be documented in this file.
-
-Group changes to describe their impact on the project, as follows:
-
- Added for new features.
- Changed for changes in existing functionality.
- Deprecated for once-stable features removed in upcoming releases.
- Removed for deprecated features removed in this release.
- Fixed for any bug fixes.
- Security to invite users to upgrade in case of vulnerabilities.
-
-# Preamble
-
-This changelog file was started on October 2019. Previous changes were more or less tracked in the *NEWS* file.
-
-## [4.3.0] - 2019-10-14
-
-### Added
-- New cmake options to make "small" builds of liblinphone, by excluding adavanced IM and DB storage.
-
-### Changed
-- Optimisations in chatrooms loading from Sqlite DB, improving startup time.
-- License changed to GNU GPLv3.
-
diff --git a/build/rpm/liblinphone.spec.cmake b/build/rpm/liblinphone.spec.cmake
index 03fcc579ea757779a6a8dbf11cd7035e00ad6174..e6b799a40ff54c88bfe7447901964609e43e2a2b 100755
--- a/build/rpm/liblinphone.spec.cmake
+++ b/build/rpm/liblinphone.spec.cmake
@@ -89,7 +89,7 @@ rm -rf $RPM_BUILD_ROOT
%files
%defattr(-,root,root)
-%doc ChangeLog.md LICENSE.txt README.md
+%doc CHANGELOG.md LICENSE.txt README.md
%if @ENABLE_DAEMON@ || @ENABLE_CONSOLE_UI@ || @ENABLE_TOOLS@
%{_bindir}/*
%endif
diff --git a/config.h.cmake b/config.h.cmake
index 31a339a26be3b34f69b09fcc00d4ff03fc84f0fb..ab8b6b0b07efe03c6a5a9451f2871427b5d91f05 100644
--- a/config.h.cmake
+++ b/config.h.cmake
@@ -50,3 +50,5 @@
#cmakedefine HAVE_ADVANCED_IM
#cmakedefine HAVE_DB_STORAGE
#cmakedefine ENABLE_UPDATE_CHECK 1
+#cmakedefine HAVE_GETIFADDRS
+
diff --git a/coreapi/bellesip_sal/sal_sdp.c b/coreapi/bellesip_sal/sal_sdp.c
index 2892c77f71fd03f0f3f74fe00ae1580c30ef8766..c87361e8d5f985ea92b64c7666fed999447d0399 100644
--- a/coreapi/bellesip_sal/sal_sdp.c
+++ b/coreapi/bellesip_sal/sal_sdp.c
@@ -192,6 +192,20 @@ static belle_sdp_attribute_t * create_rtcp_xr_attribute(const OrtpRtcpXrConfigur
return BELLE_SDP_ATTRIBUTE(attribute);
}
+static void add_mid_attributes(belle_sdp_media_description_t *media_desc, const SalStreamDescription *stream){
+ if (stream->mid[0] != '\0'){
+ belle_sdp_media_description_add_attribute(media_desc, belle_sdp_attribute_create("mid", stream->mid));
+ }
+ if (stream->mid_rtp_ext_header_id){
+ char *value = bctbx_strdup_printf("%i urn:ietf:params:rtp-hdrext:sdes:mid", stream->mid_rtp_ext_header_id);
+ belle_sdp_media_description_add_attribute(media_desc, belle_sdp_attribute_create("extmap", value));
+ bctbx_free(value);
+ }
+ if (stream->bundle_only){
+ belle_sdp_media_description_add_attribute(media_desc, belle_sdp_attribute_create("bundle-only", NULL));
+ }
+}
+
static void stream_description_to_sdp ( belle_sdp_session_description_t *session_desc, const SalMediaDescription *md, const SalStreamDescription *stream ) {
belle_sdp_mime_parameter_t* mime_param;
belle_sdp_media_description_t* media_desc;
@@ -205,6 +219,7 @@ static void stream_description_to_sdp ( belle_sdp_session_description_t *session
int rtp_port;
int rtcp_port;
bool_t different_rtp_and_rtcp_addr;
+ bool_t stream_enabled = sal_stream_description_enabled(stream);
rtp_addr=stream->rtp_addr;
rtcp_addr=stream->rtcp_addr;
@@ -318,7 +333,8 @@ static void stream_description_to_sdp ( belle_sdp_session_description_t *session
if (stream->rtcp_mux){
belle_sdp_media_description_add_attribute(media_desc, belle_sdp_attribute_create ("rtcp-mux",NULL ) );
}
-
+ add_mid_attributes(media_desc, stream);
+
if (rtp_port != 0) {
different_rtp_and_rtcp_addr = (rtcp_addr[0] != '\0') && (strcmp(rtp_addr, rtcp_addr) != 0);
if ((rtcp_port != (rtp_port + 1)) || (different_rtp_and_rtcp_addr == TRUE)) {
@@ -346,11 +362,11 @@ static void stream_description_to_sdp ( belle_sdp_session_description_t *session
}
}
- if ((rtp_port != 0) && (sal_stream_description_has_avpf(stream) || sal_stream_description_has_implicit_avpf(stream))) {
+ if (stream_enabled && (sal_stream_description_has_avpf(stream) || sal_stream_description_has_implicit_avpf(stream))) {
add_rtcp_fb_attributes(media_desc, md, stream);
}
- if ((rtp_port != 0) && (stream->rtcp_xr.enabled == TRUE)) {
+ if (stream_enabled && (stream->rtcp_xr.enabled == TRUE)) {
char sastr[1024] = {0};
char mastr[1024] = {0};
size_t saoff = 0;
@@ -391,7 +407,25 @@ static void stream_description_to_sdp ( belle_sdp_session_description_t *session
belle_sdp_session_description_add_media_description(session_desc, media_desc);
}
-belle_sdp_session_description_t * media_description_to_sdp ( const SalMediaDescription *desc ) {
+static void bundles_to_sdp(const bctbx_list_t *bundles, belle_sdp_session_description_t *session_desc){
+ const bctbx_list_t *elem;
+ for (elem = bundles; elem != NULL; elem = elem->next){
+ SalStreamBundle *bundle = (SalStreamBundle*) elem->data;
+ const bctbx_list_t * id_iterator;
+ char *attr_value = ms_strdup("BUNDLE");
+ for (id_iterator = bundle->mids; id_iterator != NULL; id_iterator = id_iterator->next){
+ const char *mid = (const char*) id_iterator->data;
+ char *tmp = ms_strdup_printf("%s %s", attr_value, mid);
+ ms_free(attr_value);
+ attr_value = tmp;
+
+ }
+ belle_sdp_session_description_add_attribute(session_desc, belle_sdp_attribute_create("group", attr_value));
+ bctbx_free(attr_value);
+ }
+}
+
+belle_sdp_session_description_t * media_description_to_sdp(const SalMediaDescription *desc) {
belle_sdp_session_description_t* session_desc=belle_sdp_session_description_new();
bool_t inet6;
belle_sdp_origin_t* origin;
@@ -440,6 +474,10 @@ belle_sdp_session_description_t * media_description_to_sdp ( const SalMediaDescr
if (desc->rtcp_xr.enabled == TRUE) {
belle_sdp_session_description_add_attribute(session_desc, create_rtcp_xr_attribute(&desc->rtcp_xr));
}
+
+ if (desc->bundles)
+ bundles_to_sdp(desc->bundles, session_desc);
+
if (desc->custom_sdp_attributes) {
belle_sdp_session_description_t *custom_desc = (belle_sdp_session_description_t *)desc->custom_sdp_attributes;
belle_sip_list_t *l = belle_sdp_session_description_get_attributes(custom_desc);
@@ -807,6 +845,14 @@ static SalStreamDescription * sdp_to_stream_description(SalMediaDescription *md,
}
stream->rtcp_mux = belle_sdp_media_description_get_attribute(media_desc, "rtcp-mux") != NULL;
+ stream->bundle_only = belle_sdp_media_description_get_attribute(media_desc, "bundle-only") != NULL;
+
+ attribute = belle_sdp_media_description_get_attribute(media_desc, "mid");
+ if (attribute){
+ value = belle_sdp_attribute_get_value(attribute);
+ if (value)
+ strncpy(stream->mid, value, sizeof(stream->mid) - 1);
+ }
/* Get media payload types */
sdp_parse_payload_types(media_desc, stream);
@@ -876,18 +922,45 @@ static SalStreamDescription * sdp_to_stream_description(SalMediaDescription *md,
stream->rtcp_xr = md->rtcp_xr; // Use session parameters if no stream parameters are defined
sdp_parse_media_rtcp_xr_parameters(media_desc, &stream->rtcp_xr);
- /* Get the custom attributes */
+ /* Get the custom attributes, and parse some 'extmap'*/
for (custom_attribute_it = belle_sdp_media_description_get_attributes(media_desc); custom_attribute_it != NULL; custom_attribute_it = custom_attribute_it->next) {
belle_sdp_attribute_t *attr = (belle_sdp_attribute_t *)custom_attribute_it->data;
- stream->custom_sdp_attributes = sal_custom_sdp_attribute_append(stream->custom_sdp_attributes, belle_sdp_attribute_get_name(attr), belle_sdp_attribute_get_value(attr));
+ const char *attr_name = belle_sdp_attribute_get_name(attr);
+ const char *attr_value = belle_sdp_attribute_get_value(attr);
+ stream->custom_sdp_attributes = sal_custom_sdp_attribute_append(stream->custom_sdp_attributes, attr_name, attr_value);
+
+ if (strcasecmp(attr_name, "extmap") == 0){
+ char *extmap_urn = (char*)bctbx_malloc0(strlen(attr_value) + 1);
+ int rtp_ext_header_id = 0;
+ if (sscanf(attr_value, "%i %s", &rtp_ext_header_id, extmap_urn) > 0
+ && strcasecmp(extmap_urn, "urn:ietf:params:rtp-hdrext:sdes:mid") == 0){
+ stream->mid_rtp_ext_header_id = rtp_ext_header_id;
+ }
+ bctbx_free(extmap_urn);
+ }
}
md->nb_streams++;
return stream;
}
+static void add_bundles(SalMediaDescription *desc, const char *ids){
+ char *tmp = (char*)ms_malloc0(strlen(ids) + 1);
+ int err;
+ SalStreamBundle *bundle = sal_media_description_add_new_bundle(desc);
+ do{
+ int consumed = 0;
+ err = sscanf(ids, "%s%n", tmp, &consumed);
+ if (err > 0){
+ bundle->mids = bctbx_list_append(bundle->mids, bctbx_strdup(tmp));
+ ids += consumed;
+ }else break;
+ }while( *ids != '\0');
+ ms_free(tmp);
+}
+
-int sdp_to_media_description ( belle_sdp_session_description_t *session_desc, SalMediaDescription *desc ) {
+int sdp_to_media_description( belle_sdp_session_description_t *session_desc, SalMediaDescription *desc ) {
belle_sdp_connection_t* cnx;
belle_sip_list_t* media_desc_it;
belle_sdp_media_description_t* media_desc;
@@ -954,10 +1027,17 @@ int sdp_to_media_description ( belle_sdp_session_description_t *session_desc, S
/* Get session RTCP-XR attributes if any */
sdp_parse_session_rtcp_xr_parameters(session_desc, &desc->rtcp_xr);
- /* Get the custom attributes */
+ /* Get the custom attributes, parse some of them that are relevant */
for (custom_attribute_it = belle_sdp_session_description_get_attributes(session_desc); custom_attribute_it != NULL; custom_attribute_it = custom_attribute_it->next) {
belle_sdp_attribute_t *attr = (belle_sdp_attribute_t *)custom_attribute_it->data;
desc->custom_sdp_attributes = sal_custom_sdp_attribute_append(desc->custom_sdp_attributes, belle_sdp_attribute_get_name(attr), belle_sdp_attribute_get_value(attr));
+
+ if (strcasecmp(belle_sdp_attribute_get_name(attr), "group") == 0){
+ value = belle_sdp_attribute_get_value(attr);
+ if (value && strncasecmp(value, "BUNDLE", strlen("BUNDLE")) == 0){
+ add_bundles(desc, value + strlen("BUNDLE"));
+ }
+ }
}
for ( media_desc_it=belle_sdp_session_description_get_media_descriptions ( session_desc )
diff --git a/coreapi/linphonecore.c b/coreapi/linphonecore.c
index 41142fff623f261524badee0ad6adcc407a4ae3a..ad15917a242bdd23471818c17a593a9b4d54b760 100644
--- a/coreapi/linphonecore.c
+++ b/coreapi/linphonecore.c
@@ -69,6 +69,8 @@
#include "content/content-manager.h"
#include "content/content-type.h"
#include "core/core-p.h"
+#include "conference/session/media-session.h"
+#include "conference/session/media-session-p.h"
// For migration purpose.
#include "address/address-p.h"
@@ -2557,7 +2559,6 @@ static void linphone_core_init(LinphoneCore * lc, LinphoneCoreCbs *cbs, LpConfig
linphone_presence_model_set_basic_status(lc->presence_model, LinphonePresenceBasicStatusOpen);
_linphone_core_read_config(lc);
- linphone_core_add_linphone_spec(lc, "ephemeral");
linphone_core_set_state(lc, LinphoneGlobalReady, "Ready");
if (automatically_start) {
@@ -5590,48 +5591,20 @@ void * linphone_core_get_native_video_window_id(const LinphoneCore *lc){
#ifdef VIDEO_ENABLED
/*case where it was not set but we want to get the one automatically created by mediastreamer2 (desktop versions only)*/
LinphoneCall *call=linphone_core_get_current_call (lc);
+
if (call) {
- VideoStream *vstream = reinterpret_cast<VideoStream *>(linphone_call_get_stream(call, LinphoneStreamTypeVideo));
- if (vstream)
- return video_stream_get_native_window_id(vstream);
+ auto ms = dynamic_pointer_cast<LinphonePrivate::MediaSession>(L_GET_PRIVATE_FROM_C_OBJECT(call)->getActiveSession());
+ if (ms) return ms->getNativeVideoWindowId();
}
#endif
}
return 0;
}
-/* unsets the video id for all calls (indeed it may be kept by filters or videostream object itself by paused calls)*/
-static void unset_video_window_id(LinphoneCore *lc, bool_t preview, void *id){
- if ((id != NULL)
-#ifndef _WIN32
- && ((unsigned long)id != (unsigned long)-1)
-#endif
- ){
- ms_error("Invalid use of unset_video_window_id()");
- return;
- }
- L_GET_PRIVATE_FROM_C_OBJECT(lc)->unsetVideoWindowId(!!preview, id);
-}
void _linphone_core_set_native_video_window_id(LinphoneCore *lc, void *id) {
- if ((id == NULL)
-#ifndef _WIN32
- || ((unsigned long)id == (unsigned long)-1)
-#endif
- ){
- unset_video_window_id(lc,FALSE,id);
- }
+ L_GET_PRIVATE_FROM_C_OBJECT(lc)->setVideoWindowId(false, id);
lc->video_window_id=id;
-#ifdef VIDEO_ENABLED
- {
- LinphoneCall *call=linphone_core_get_current_call(lc);
- if (call) {
- VideoStream *vstream = reinterpret_cast<VideoStream *>(linphone_call_get_stream(call, LinphoneStreamTypeVideo));
- if (vstream)
- video_stream_set_native_window_id(vstream,id);
- }
- }
-#endif
}
void linphone_core_set_native_video_window_id(LinphoneCore *lc, void *id) {
@@ -5650,10 +5623,10 @@ void * linphone_core_get_native_preview_window_id(const LinphoneCore *lc){
/*case where we want the id automatically created by mediastreamer2 (desktop versions only)*/
#ifdef VIDEO_ENABLED
LinphoneCall *call=linphone_core_get_current_call(lc);
+
if (call) {
- VideoStream *vstream = reinterpret_cast<VideoStream *>(linphone_call_get_stream(call, LinphoneStreamTypeVideo));
- if (vstream)
- return video_stream_get_native_preview_window_id(vstream);
+ auto ms = dynamic_pointer_cast<LinphonePrivate::MediaSession>(L_GET_PRIVATE_FROM_C_OBJECT(call)->getActiveSession());
+ if (ms) return ms->getNativePreviewVideoWindowId();
}
if (lc->previewstream)
return video_preview_get_native_window_id(lc->previewstream);
@@ -5663,24 +5636,11 @@ void * linphone_core_get_native_preview_window_id(const LinphoneCore *lc){
}
void _linphone_core_set_native_preview_window_id(LinphoneCore *lc, void *id) {
- if ((id == NULL)
-#ifndef _WIN32
- || ((unsigned long)id == (unsigned long)-1)
-#endif
- ) {
- unset_video_window_id(lc,TRUE,id);
- }
+ L_GET_PRIVATE_FROM_C_OBJECT(lc)->setVideoWindowId(true, id);
lc->preview_window_id=id;
#ifdef VIDEO_ENABLED
- {
- LinphoneCall *call=linphone_core_get_current_call(lc);
- if (call) {
- VideoStream *vstream = reinterpret_cast<VideoStream *>(linphone_call_get_stream(call, LinphoneStreamTypeVideo));
- if (vstream)
- video_stream_set_native_preview_window_id(vstream,id);
- }else if (lc->previewstream){
- video_preview_set_native_window_id(lc->previewstream,id);
- }
+ if (lc->previewstream){
+ video_preview_set_native_window_id(lc->previewstream,id);
}
#endif
}
@@ -6132,7 +6092,11 @@ void sip_config_uninit(LinphoneCore *lc)
for(elem=config->proxies;elem!=NULL;elem=bctbx_list_next(elem)){
LinphoneProxyConfig *cfg=(LinphoneProxyConfig*)(elem->data);
_linphone_proxy_config_unpublish(cfg); /* to unpublish without changing the stored flag enable_publish */
- _linphone_proxy_config_unregister(cfg); /* to unregister without changing the stored flag enable_register */
+
+ /* Do not unregister when push notifications are allowed, otherwise this clears tokens from the SIP server.*/
+ if (!linphone_proxy_config_is_push_notification_allowed(cfg)){
+ _linphone_proxy_config_unregister(cfg); /* to unregister without changing the stored flag enable_register */
+ }
}
ms_message("Unregistration started.");
@@ -7146,7 +7110,7 @@ void linphone_core_set_media_encryption_mandatory(LinphoneCore *lc, bool_t m) {
}
void linphone_core_init_default_params(LinphoneCore*lc, LinphoneCallParams *params) {
- L_GET_CPP_PTR_FROM_C_OBJECT(params)->initDefault(L_GET_CPP_PTR_FROM_C_OBJECT(lc));
+ L_GET_CPP_PTR_FROM_C_OBJECT(params)->initDefault(L_GET_CPP_PTR_FROM_C_OBJECT(lc), LinphoneCallOutgoing);
}
void linphone_core_set_device_identifier(LinphoneCore *lc,const char* device_id) {
@@ -7336,6 +7300,14 @@ bool_t linphone_core_video_multicast_enabled(const LinphoneCore *lc) {
return lc->rtp_conf.video_multicast_enabled;
}
+bool_t linphone_core_rtp_bundle_enabled(const LinphoneCore *lc){
+ return linphone_config_get_bool(lc->config, "rtp", "bundle", FALSE);
+}
+
+void linphone_core_enable_rtp_bundle(LinphoneCore *lc, bool_t value){
+ linphone_config_set_bool(lc->config, "rtp", "bundle", value);
+}
+
void linphone_core_set_video_preset(LinphoneCore *lc, const char *preset) {
lp_config_set_string(lc->config, "video", "preset", preset);
}
diff --git a/coreapi/misc.c b/coreapi/misc.c
index bcf3082b4df51b69e66c5b7cf3f56f5147fe128e..c7815cca6082f9d2e354fd66d3093c3dad1ad570 100644
--- a/coreapi/misc.c
+++ b/coreapi/misc.c
@@ -45,10 +45,6 @@
#undef snprintf
#include <mediastreamer2/stun.h>
-#ifdef HAVE_GETIFADDRS
-#include <net/if.h>
-#include <ifaddrs.h>
-#endif
#include <math.h>
#if _MSC_VER
#define snprintf _snprintf
diff --git a/coreapi/offeranswer.c b/coreapi/offeranswer.c
index 871361ab883d7731d6f049c9216026afbe7516f3..406ebe49e337e59086db517480b5d888013c3e41 100644
--- a/coreapi/offeranswer.c
+++ b/coreapi/offeranswer.c
@@ -336,7 +336,7 @@ static SalStreamDir compute_dir_incoming(SalStreamDir local, SalStreamDir offere
static void initiate_outgoing(MSFactory* factory, const SalStreamDescription *local_offer,
const SalStreamDescription *remote_answer,
SalStreamDescription *result){
- if (remote_answer->rtp_port!=0)
+ if (sal_stream_description_enabled(remote_answer))
result->payloads=match_payloads(factory, local_offer->payloads,remote_answer->payloads,TRUE,FALSE);
else {
ms_message("Local stream description [%p] rejected by peer",local_offer);
@@ -411,7 +411,18 @@ static void initiate_outgoing(MSFactory* factory, const SalStreamDescription *lo
result->dir=compute_dir_outgoing(local_offer->dir,remote_answer->dir);
}
-
+ if (remote_answer->mid[0] != '\0'){
+ if (local_offer->mid[0] != '\0'){
+ strncpy(result->mid, remote_answer->mid, sizeof(result->mid) - 1);
+ result->mid_rtp_ext_header_id = remote_answer->mid_rtp_ext_header_id;
+ result->bundle_only = remote_answer->bundle_only;
+ result->rtcp_mux = TRUE; /* RTCP mux must be enabled in bundle mode. */
+ }else{
+ ms_error("The remote has set a mid in an answer while we didn't offered it.");
+ }
+ }else{
+ result->rtcp_mux = remote_answer->rtcp_mux && local_offer->rtcp_mux;
+ }
if (result->payloads && !only_telephone_event(result->payloads)){
strcpy(result->rtp_addr,remote_answer->rtp_addr);
@@ -422,13 +433,14 @@ static void initiate_outgoing(MSFactory* factory, const SalStreamDescription *lo
result->ptime=remote_answer->ptime;
result->maxptime=remote_answer->maxptime;
}else{
- result->rtp_port=0;
+ sal_stream_description_disable(result);
}
if (sal_stream_description_has_srtp(result) == TRUE) {
/* verify crypto algo */
memset(result->crypto, 0, sizeof(result->crypto));
- if (!match_crypto_algo(local_offer->crypto, remote_answer->crypto, &result->crypto[0], &result->crypto_local_tag, FALSE))
- result->rtp_port = 0;
+ if (!match_crypto_algo(local_offer->crypto, remote_answer->crypto, &result->crypto[0], &result->crypto_local_tag, FALSE)){
+ sal_stream_description_disable(result);
+ }
}
result->rtp_ssrc=local_offer->rtp_ssrc;
strncpy(result->rtcp_cname,local_offer->rtcp_cname,sizeof(result->rtcp_cname));
@@ -446,19 +458,19 @@ static void initiate_outgoing(MSFactory* factory, const SalStreamDescription *lo
result->dtls_fingerprint[0] = '\0';
result->dtls_role = SalDtlsRoleInvalid;
}
- result->rtcp_mux = remote_answer->rtcp_mux && local_offer->rtcp_mux;
result->implicit_rtcp_fb = local_offer->implicit_rtcp_fb && remote_answer->implicit_rtcp_fb;
}
static void initiate_incoming(MSFactory *factory, const SalStreamDescription *local_cap,
const SalStreamDescription *remote_offer,
- SalStreamDescription *result, bool_t one_matching_codec){
+ SalStreamDescription *result, bool_t one_matching_codec, const char *bundle_owner_mid){
result->payloads=match_payloads(factory, local_cap->payloads,remote_offer->payloads, FALSE, one_matching_codec);
result->proto=remote_offer->proto;
result->type=local_cap->type;
result->dir=compute_dir_incoming(local_cap->dir,remote_offer->dir);
- if (!result->payloads || only_telephone_event(result->payloads) || remote_offer->rtp_port==0){
+
+ if (!result->payloads || only_telephone_event(result->payloads) || !sal_stream_description_enabled(remote_offer)){
result->rtp_port=0;
return;
}
@@ -488,12 +500,28 @@ static void initiate_incoming(MSFactory *factory, const SalStreamDescription *lo
result->ptime=local_cap->ptime;
result->maxptime=local_cap->maxptime;
}
+
+ /* Handle RTP bundle negociation */
+ if (remote_offer->mid[0] != '\0' && bundle_owner_mid){
+ strncpy(result->mid, remote_offer->mid, sizeof(result->mid) - 1);
+ result->mid_rtp_ext_header_id = remote_offer->mid_rtp_ext_header_id;
+
+ if (strcmp(bundle_owner_mid, remote_offer->mid) != 0){
+ /* The stream is a secondary one part of a bundle.
+ * In this case it must set the bundle-only attribute, and set port to zero.*/
+ result->bundle_only = TRUE;
+ result->rtp_port = 0;
+ }
+ result->rtcp_mux = TRUE; /* RTCP mux must be enabled in bundle mode. */
+ }else {
+ result->rtcp_mux = remote_offer->rtcp_mux && local_cap->rtcp_mux;
+ }
if (sal_stream_description_has_srtp(result) == TRUE) {
/* select crypto algo */
memset(result->crypto, 0, sizeof(result->crypto));
if (!match_crypto_algo(local_cap->crypto, remote_offer->crypto, &result->crypto[0], &result->crypto_local_tag, TRUE)) {
- result->rtp_port = 0;
+ sal_stream_description_disable(result);
ms_message("No matching crypto algo for remote stream's offer [%p]",remote_offer);
}
@@ -528,7 +556,6 @@ static void initiate_incoming(MSFactory *factory, const SalStreamDescription *lo
result->dtls_fingerprint[0] = '\0';
result->dtls_role = SalDtlsRoleInvalid;
}
- result->rtcp_mux = remote_offer->rtcp_mux && local_cap->rtcp_mux;
result->implicit_rtcp_fb = local_cap->implicit_rtcp_fb && remote_offer->implicit_rtcp_fb;
}
@@ -567,6 +594,16 @@ int offer_answer_initiate_outgoing(MSFactory *factory, const SalMediaDescription
if ((local_offer->rtcp_xr.enabled == TRUE) && (remote_answer->rtcp_xr.enabled == FALSE)) {
result->rtcp_xr.enabled = FALSE;
}
+ /* TODO: check that the bundle answer is compliant with our offer.
+ * For now, just check the presence of a bundle response. */
+ if (local_offer->bundles){
+ if (remote_answer->bundles){
+ /* Copy the bundle offering to the result media description. */
+ result->bundles = bctbx_list_copy_with_data(remote_answer->bundles, (bctbx_list_copy_func) sal_stream_bundle_clone);
+ }
+ }else if (remote_answer->bundles){
+ ms_error("Remote answerer is proposing bundles, which we did not offer.");
+ }
return 0;
}
@@ -582,11 +619,23 @@ int offer_answer_initiate_incoming(MSFactory *factory, const SalMediaDescription
int i;
const SalStreamDescription *ls=NULL,*rs;
+ if (remote_offer->bundles && local_capabilities->accept_bundles){
+ /* Copy the bundle offering to the result media description. */
+ result->bundles = bctbx_list_copy_with_data(remote_offer->bundles, (bctbx_list_copy_func) sal_stream_bundle_clone);
+ }
+
for(i=0;i<remote_offer->nb_streams;++i){
rs = &remote_offer->streams[i];
ls = &local_capabilities->streams[i];
if (ls && rs->type == ls->type && rs->proto == ls->proto){
- initiate_incoming(factory, ls,rs,&result->streams[i],one_matching_codec);
+ const char *bundle_owner_mid = NULL;
+ if (local_capabilities->accept_bundles){
+ int owner_index = sal_media_description_get_index_of_transport_owner(remote_offer, rs);
+ if (owner_index != -1){
+ bundle_owner_mid = remote_offer->streams[owner_index].mid;
+ }
+ }
+ initiate_incoming(factory, ls,rs,&result->streams[i],one_matching_codec, bundle_owner_mid);
// Handle global RTCP FB attributes
result->streams[i].rtcp_fb.generic_nack_enabled = rs->rtcp_fb.generic_nack_enabled;
result->streams[i].rtcp_fb.tmmbr_enabled = rs->rtcp_fb.tmmbr_enabled;
diff --git a/coreapi/private_functions.h b/coreapi/private_functions.h
index 5384d3ce8ed35f011ec3b04702f93b86a75089a5..5d2ae99a83df43864bf68916438feb162cc0788a 100644
--- a/coreapi/private_functions.h
+++ b/coreapi/private_functions.h
@@ -240,8 +240,6 @@ LINPHONE_PUBLIC void linphone_core_enable_short_turn_refresh(LinphoneCore *lc, b
LINPHONE_PUBLIC void linphone_call_stats_fill(LinphoneCallStats *stats, MediaStream *ms, OrtpEvent *ev);
void linphone_call_stats_update(LinphoneCallStats *stats, MediaStream *stream);
LinphoneCallStats *_linphone_call_stats_new(void);
-void _linphone_call_stats_uninit(LinphoneCallStats *stats);
-void _linphone_call_stats_clone(LinphoneCallStats *dst, const LinphoneCallStats *src);
void _linphone_call_stats_set_ice_state (LinphoneCallStats *stats, LinphoneIceState state);
void _linphone_call_stats_set_type (LinphoneCallStats *stats, LinphoneStreamType type);
void _linphone_call_stats_set_received_rtcp (LinphoneCallStats *stats, mblk_t *m);
diff --git a/coreapi/private_structs.h b/coreapi/private_structs.h
index 0148991a0ad220889811bd490b7b6afd6efc3af8..5e6fbb49f3bd4c9e3407385771884ce4f350a652 100644
--- a/coreapi/private_structs.h
+++ b/coreapi/private_structs.h
@@ -31,8 +31,8 @@
struct _LinphoneQualityReporting{
reporting_session_report_t * reports[3]; /**Store information on audio and video media streams (RFC 6035) */
- bool_t was_video_running; /*Keep video state since last check in order to detect its (de)activation*/
LinphoneQualityReportingReportSendCb on_report_sent;
+ bool_t was_video_running; /*Keep video state since last check in order to detect its (de)activation*/
};
struct _LinphoneCallLog{
diff --git a/coreapi/proxy.c b/coreapi/proxy.c
index 35a36c38c1fe774521561b3ab34fe686ca679425..53d133fa9ec8add54a1e08e21cedf6466dba902d 100644
--- a/coreapi/proxy.c
+++ b/coreapi/proxy.c
@@ -165,7 +165,7 @@ static void linphone_proxy_config_init(LinphoneCore* lc, LinphoneProxyConfig *cf
cfg->avpf_rr_interval = lc ? !!lp_config_get_default_int(lc->config, "proxy", "avpf_rr_interval", 5) : 5;
cfg->publish_expires= lc ? lp_config_get_default_int(lc->config, "proxy", "publish_expires", -1) : -1;
cfg->publish = lc ? !!lp_config_get_default_int(lc->config, "proxy", "publish", FALSE) : FALSE;
- cfg->push_notification_allowed = lc ? !!lp_config_get_default_int(lc->config, "proxy", "push_notification_allowed", TRUE) : TRUE;
+ cfg->push_notification_allowed = lc ? !!lp_config_get_default_int(lc->config, "proxy", "push_notification_allowed", FALSE) : FALSE;
cfg->refkey = refkey ? ms_strdup(refkey) : NULL;
if (nat_policy_ref) {
LinphoneNatPolicy *policy = linphone_config_create_nat_policy_from_section(lc->config,nat_policy_ref);
@@ -589,8 +589,7 @@ static LinphoneAddress *guess_contact_for_register (LinphoneProxyConfig *cfg) {
void _linphone_proxy_config_unregister(LinphoneProxyConfig *obj) {
if (obj->op && (obj->state == LinphoneRegistrationOk ||
- (obj->state == LinphoneRegistrationProgress && obj->expires != 0)) &&
- !linphone_proxy_config_is_push_notification_allowed(obj)) {
+ (obj->state == LinphoneRegistrationProgress && obj->expires != 0))) {
obj->op->unregister();
}
}
@@ -1815,6 +1814,7 @@ void linphone_proxy_config_set_conference_factory_uri(LinphoneProxyConfig *cfg,
cfg->conference_factory_uri = bctbx_strdup(uri);
if (cfg->lc) {
linphone_core_add_linphone_spec(cfg->lc, "groupchat");
+ linphone_core_add_linphone_spec(cfg->lc, "ephemeral");
}
} else if (cfg->lc) {
bool_t remove = TRUE;
@@ -1831,6 +1831,7 @@ void linphone_proxy_config_set_conference_factory_uri(LinphoneProxyConfig *cfg,
}
if (remove) {
linphone_core_remove_linphone_spec(cfg->lc, "groupchat");
+ linphone_core_remove_linphone_spec(cfg->lc, "ephemeral");
}
}
}
diff --git a/coreapi/quality_reporting.c b/coreapi/quality_reporting.c
index 8e6973ce0af3e317577b26de867cdaf69e353012..3d036045ca38a4a2e86dcc4e933a89ee68e57d72 100644
--- a/coreapi/quality_reporting.c
+++ b/coreapi/quality_reporting.c
@@ -169,8 +169,11 @@ static bool_t media_report_enabled(LinphoneCall * call, int stats_type){
if (!quality_reporting_enabled(call))
return FALSE;
- if (stats_type == LINPHONE_CALL_STATS_VIDEO && !linphone_call_params_video_enabled(linphone_call_get_current_params(call)))
- return FALSE;
+ if (stats_type == LINPHONE_CALL_STATS_VIDEO){
+ if (!(L_GET_CPP_PTR_FROM_C_OBJECT(call)->getLog()->reporting.was_video_running
+ || linphone_call_params_video_enabled(linphone_call_get_current_params(call))) )
+ return FALSE;
+ }
if (stats_type == LINPHONE_CALL_STATS_TEXT && !linphone_call_params_realtime_text_enabled(linphone_call_get_current_params(call)))
return FALSE;
@@ -413,7 +416,7 @@ static const SalStreamDescription * get_media_stream_for_desc(const SalMediaDesc
int count;
if (smd != NULL) {
for (count = 0; count < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; ++count) {
- if (sal_stream_description_active(&smd->streams[count]) && smd->streams[count].type == sal_stream_type) {
+ if (sal_stream_description_enabled(&smd->streams[count]) && smd->streams[count].type == sal_stream_type) {
return &smd->streams[count];
}
}
@@ -480,7 +483,7 @@ static void qos_analyzer_on_action_suggested(void *user_data, int datac, const c
}
}
- appendbuf=ms_strdup_printf("%s%d;", report->qos_analyzer.timestamp?report->qos_analyzer.timestamp:"", ms_time(0));
+ appendbuf=ms_strdup_printf("%s%llu;", report->qos_analyzer.timestamp?report->qos_analyzer.timestamp:"", (unsigned long long)ms_time(0));
STR_REASSIGN(report->qos_analyzer.timestamp,appendbuf);
STR_REASSIGN(report->qos_analyzer.input_leg, ms_strdup_printf("%s aenc_ptime aenc_br a_dbw a_ubw venc_br v_dbw v_ubw tenc_br t_dbw t_ubw", datav[0]));
@@ -741,10 +744,10 @@ void linphone_reporting_call_state_updated(LinphoneCall *call){
}
}
linphone_reporting_update_ip(call);
- if (!media_report_enabled(call, LINPHONE_CALL_STATS_VIDEO) && log->reporting.was_video_running){
+ if (media_report_enabled(call, LINPHONE_CALL_STATS_VIDEO) && log->reporting.was_video_running){
send_report(call, log->reporting.reports[LINPHONE_CALL_STATS_VIDEO], "VQSessionReport");
}
- log->reporting.was_video_running=media_report_enabled(call, LINPHONE_CALL_STATS_VIDEO);
+ log->reporting.was_video_running = linphone_call_params_video_enabled(linphone_call_get_current_params(call));
break;
}
case LinphoneCallEnd:{
diff --git a/coreapi/tester_utils.h b/coreapi/tester_utils.h
index c8037577dad1599b45d2be141b3e7dd8e7d9c909..7b06acc49261c45727e522753536019d2dbfc948 100644
--- a/coreapi/tester_utils.h
+++ b/coreapi/tester_utils.h
@@ -112,6 +112,8 @@ LINPHONE_PUBLIC bool_t linphone_call_params_get_update_call_when_ice_completed(c
LINPHONE_PUBLIC int _linphone_call_stats_get_updated(const LinphoneCallStats *stats);
LINPHONE_PUBLIC bool_t _linphone_call_stats_rtcp_received_via_mux(const LinphoneCallStats *stats);
LINPHONE_PUBLIC mblk_t *_linphone_call_stats_get_received_rtcp (const LinphoneCallStats *stats);
+LINPHONE_PUBLIC bool_t _linphone_call_stats_has_received_rtcp(const LinphoneCallStats *stats);
+LINPHONE_PUBLIC bool_t _linphone_call_stats_has_sent_rtcp(const LinphoneCallStats *stats);
LINPHONE_PUBLIC LinphoneQualityReporting *linphone_call_log_get_quality_reporting(LinphoneCallLog *call_log);
LINPHONE_PUBLIC reporting_session_report_t **linphone_quality_reporting_get_reports(LinphoneQualityReporting *qreporting);
@@ -214,6 +216,8 @@ LINPHONE_PUBLIC void linphone_account_creator_cbs_set_confirmation_key(LinphoneA
LINPHONE_PUBLIC void linphone_core_delete_local_encryption_db(const LinphoneCore *lc);
LINPHONE_PUBLIC void linphone_core_set_network_reachable_internal(LinphoneCore *lc, bool_t is_reachable);
+LINPHONE_PUBLIC bctbx_list_t *linphone_fetch_local_addresses(void);
+
#ifndef __cplusplus
LINPHONE_PUBLIC Sal *linphone_core_get_sal(const LinphoneCore *lc);
LINPHONE_PUBLIC SalOp *linphone_proxy_config_get_sal_op(const LinphoneProxyConfig *cfg);
diff --git a/include/linphone/call_params.h b/include/linphone/call_params.h
index 1f29a2da86139de2d876f955d3bd53a522fbc5e2..55381ee8a748d5e3615f9982ccc1c18b51ff76cb 100644
--- a/include/linphone/call_params.h
+++ b/include/linphone/call_params.h
@@ -520,6 +520,27 @@ LINPHONE_PUBLIC bctbx_list_t* linphone_call_params_get_custom_contents (const Li
**/
LINPHONE_PUBLIC void linphone_call_params_add_custom_content (LinphoneCallParams *params, LinphoneContent *content);
+/**
+ * Indicates whether RTP bundle mode (also known as Media Multiplexing) is enabled.
+ * See https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54 for more information.
+ * @param[in] params the #LinphoneCallParams
+ * @return a boolean indicating the enablement of rtp bundle mode.
+ * @ingroup media_parameters
+ */
+LINPHONE_PUBLIC bool_t linphone_call_params_rtp_bundle_enabled(const LinphoneCallParams *params);
+
+/**
+ * Enables or disables RTP bundle mode (Media Multiplexing).
+ * See https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54 for more information about the feature.
+ * When enabled, liblinphone will try to negociate the use of a single port for all streams.
+ * It automatically enables rtcp-mux.
+ * @param[in] params the #LinphoneCallParams
+ * @param[in] value a boolean to indicate whether the feature is to be enabled.
+ * @ingroup media_parameters
+ */
+LINPHONE_PUBLIC void linphone_call_params_enable_rtp_bundle(LinphoneCallParams *params, bool_t value);
+
+
/*******************************************************************************
* DEPRECATED *
******************************************************************************/
diff --git a/include/linphone/core.h b/include/linphone/core.h
index 692002be9f3b08fb10027c685bec3494803252e7..2c08fb6348bbf300180262de90463be2cd028b1e 100644
--- a/include/linphone/core.h
+++ b/include/linphone/core.h
@@ -5151,6 +5151,27 @@ LINPHONE_PUBLIC void linphone_core_enable_video_multicast(LinphoneCore *core, bo
**/
LINPHONE_PUBLIC bool_t linphone_core_video_multicast_enabled(const LinphoneCore *core);
+/**
+ * Returns whether RTP bundle mode (also known as Media Multiplexing) is enabled.
+ * See https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54 for more information.
+ * @param[in] lc the #LinphoneCore
+ * @return a boolean indicating the enablement of rtp bundle mode.
+ * @ingroup media_parameters
+ */
+LINPHONE_PUBLIC bool_t linphone_core_rtp_bundle_enabled(const LinphoneCore *lc);
+
+/**
+ * Enables or disables RTP bundle mode (Media Multiplexing).
+ * See https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54 for more information about the feature.
+ * When enabled, liblinphone will try to negociate the use of a single port for all streams when doing an outgoing call.
+ * It automatically enables rtcp-mux.
+ * This feature can also be enabled per-call using #LinphoneCallParams.
+ * @param[in] lc the #LinphoneCore
+ * @param[in] value a boolean to indicate whether the feature is to be enabled.
+ * @ingroup media_parameters
+ */
+LINPHONE_PUBLIC void linphone_core_enable_rtp_bundle(LinphoneCore *lc, bool_t value);
+
/**
* @brief Set the network simulator parameters.
*
diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt
index c4038e576657d82355b0dfb72d8bcac15c10bde0..bfb14125994e427e343872800b4f72ae49766f38 100644
--- a/src/CMakeLists.txt
+++ b/src/CMakeLists.txt
@@ -164,8 +164,11 @@ set(LINPHONE_CXX_OBJECTS_PRIVATE_HEADER_FILES
conference/session/call-session-p.h
conference/session/call-session.h
conference/session/media-session.h
+ conference/session/streams.h
conference/session/port-config.h
conference/session/tone-manager.h
+ conference/session/ms2-streams.h
+ conference/session/media-description-renderer.h
containers/lru-cache.h
content/content-disposition.h
content/content-manager.h
@@ -210,7 +213,7 @@ set(LINPHONE_CXX_OBJECTS_PRIVATE_HEADER_FILES
event-log/events.h
hacks/hacks.h
logger/logger.h
- nat/ice-agent.h
+ nat/ice-service.h
nat/stun-client.h
object/app-data-container.h
object/base-object-p.h
@@ -231,6 +234,7 @@ set(LINPHONE_CXX_OBJECTS_PRIVATE_HEADER_FILES
utils/background-task.h
utils/general-internal.h
utils/payload-type-handler.h
+ utils/if-addrs.h
variant/variant.h
)
@@ -333,6 +337,14 @@ set(LINPHONE_CXX_OBJECTS_SOURCE_FILES
conference/session/call-session.cpp
conference/session/media-session.cpp
conference/session/tone-manager.cpp
+ conference/session/media-description-renderer.cpp
+ conference/session/stream.cpp
+ conference/session/streams-group.cpp
+ conference/session/ms2-stream.cpp
+ conference/session/audio-stream.cpp
+ conference/session/video-stream.cpp
+ conference/session/rtt-stream.cpp
+ conference/session/media-session.cpp
content/content-disposition.cpp
content/content-manager.cpp
content/content-type.cpp
@@ -366,7 +378,7 @@ set(LINPHONE_CXX_OBJECTS_SOURCE_FILES
event-log/event-log.cpp
hacks/hacks.cpp
logger/logger.cpp
- nat/ice-agent.cpp
+ nat/ice-service.cpp
nat/stun-client.cpp
object/app-data-container.cpp
object/base-object.cpp
@@ -388,6 +400,7 @@ set(LINPHONE_CXX_OBJECTS_SOURCE_FILES
utils/general.cpp
utils/payload-type-handler.cpp
utils/utils.cpp
+ utils/if-addrs.cpp
variant/variant.cpp
)
diff --git a/src/c-wrapper/api/c-call-params.cpp b/src/c-wrapper/api/c-call-params.cpp
index d6b0fe669299d55f64d907c5100f00833b513d51..03983c9a7a0f003e9c323eb53b65f7dd742c129b 100644
--- a/src/c-wrapper/api/c-call-params.cpp
+++ b/src/c-wrapper/api/c-call-params.cpp
@@ -508,6 +508,14 @@ void linphone_call_params_add_custom_content (LinphoneCallParams *params, Linpho
L_GET_CPP_PTR_FROM_C_OBJECT(params)->addCustomContent(*cppContent);
}
+bool_t linphone_call_params_rtp_bundle_enabled(const LinphoneCallParams *params){
+ return (bool_t)L_GET_CPP_PTR_FROM_C_OBJECT(params)->rtpBundleEnabled();
+}
+
+void linphone_call_params_enable_rtp_bundle(LinphoneCallParams *params, bool_t value){
+ L_GET_CPP_PTR_FROM_C_OBJECT(params)->enableRtpBundle(!!value);
+}
+
// =============================================================================
// Reference and user data handling functions.
// =============================================================================
@@ -536,7 +544,7 @@ void linphone_call_params_unref (LinphoneCallParams *cp) {
LinphoneCallParams *linphone_call_params_new (LinphoneCore *core) {
LinphoneCallParams *params = L_INIT(CallParams);
L_SET_CPP_PTR_FROM_C_OBJECT(params, new LinphonePrivate::MediaSessionParams());
- L_GET_CPP_PTR_FROM_C_OBJECT(params)->initDefault(L_GET_CPP_PTR_FROM_C_OBJECT(core));
+ L_GET_CPP_PTR_FROM_C_OBJECT(params)->initDefault(L_GET_CPP_PTR_FROM_C_OBJECT(core), LinphoneCallOutgoing);
return params;
}
diff --git a/src/c-wrapper/api/c-call-stats.cpp b/src/c-wrapper/api/c-call-stats.cpp
index 45f32311dba6fad848a2c97a6236118674e49aa7..88ef08cfccaf5786b508ae22e05c3bce4aafc7f8 100644
--- a/src/c-wrapper/api/c-call-stats.cpp
+++ b/src/c-wrapper/api/c-call-stats.cpp
@@ -23,7 +23,8 @@
// =============================================================================
-void _linphone_call_stats_clone (LinphoneCallStats *dst, const LinphoneCallStats *src);
+static void _linphone_call_stats_clone (LinphoneCallStats *dst, const LinphoneCallStats *src);
+static void _linphone_call_stats_uninit (LinphoneCallStats *stats);
/**
* The LinphoneCallStats objects carries various statistic informations regarding quality of audio or video streams.
@@ -53,8 +54,8 @@ struct _LinphoneCallStats {
rtp_stats_t rtp_stats; /**< RTP stats */
int rtp_remote_family; /**< Ip adress family of the remote destination */
int clockrate; /*RTP clockrate of the stream, provided here for easily converting timestamp units expressed in RTCP packets in milliseconds*/
- bool_t rtcp_received_via_mux; /*private flag, for non-regression test only*/
float estimated_download_bandwidth; /**<Estimated download bandwidth measurement of received stream, expressed in kbit/s, including IP/UDP/RTP headers*/
+ bool_t rtcp_received_via_mux; /*private flag, for non-regression test only*/
};
BELLE_SIP_DECLARE_VPTR_NO_EXPORT(LinphoneCallStats);
@@ -62,7 +63,7 @@ BELLE_SIP_DECLARE_VPTR_NO_EXPORT(LinphoneCallStats);
BELLE_SIP_DECLARE_NO_IMPLEMENTED_INTERFACES(LinphoneCallStats);
BELLE_SIP_INSTANCIATE_VPTR(LinphoneCallStats, belle_sip_object_t,
- NULL, // destroy
+ _linphone_call_stats_uninit, // destroy
_linphone_call_stats_clone, // clone
NULL, // marshal
FALSE
@@ -77,7 +78,7 @@ LinphoneCallStats *_linphone_call_stats_new () {
return stats;
}
-void _linphone_call_stats_uninit (LinphoneCallStats *stats) {
+static void _linphone_call_stats_uninit (LinphoneCallStats *stats) {
if (stats->received_rtcp) {
freemsg(stats->received_rtcp);
stats->received_rtcp=NULL;
@@ -88,16 +89,26 @@ void _linphone_call_stats_uninit (LinphoneCallStats *stats) {
}
}
-void _linphone_call_stats_clone (LinphoneCallStats *dst, const LinphoneCallStats *src) {
- /*
- * Save the belle_sip_object_t part, copy the entire structure and restore the belle_sip_object_t part
- */
- belle_sip_object_t tmp = dst->base;
- memcpy(dst, src, sizeof(LinphoneCallStats));
- dst->base = tmp;
-
- dst->received_rtcp = NULL;
- dst->sent_rtcp = NULL;
+static void _linphone_call_stats_clone (LinphoneCallStats *dst, const LinphoneCallStats *src) {
+ dst->type = src->type;
+ dst->jitter_stats = src->jitter_stats;
+ dst->received_rtcp = src->received_rtcp ? dupmsg(src->received_rtcp) : nullptr;
+ dst->sent_rtcp = src->sent_rtcp ? dupmsg(src->sent_rtcp) : nullptr;
+ dst->round_trip_delay = src->round_trip_delay;
+ dst->ice_state = src->ice_state;
+ dst->upnp_state = src->upnp_state;
+ dst->download_bandwidth = src->download_bandwidth;
+ dst->upload_bandwidth = src->upload_bandwidth;
+ dst->local_late_rate = src->local_late_rate;
+ dst->local_loss_rate = src->local_loss_rate;
+ dst->updated = src->updated;
+ dst->rtcp_download_bandwidth = src->rtcp_download_bandwidth;
+ dst->rtcp_upload_bandwidth = src->rtcp_upload_bandwidth;
+ dst->rtp_stats = src->rtp_stats;
+ dst->rtp_remote_family = src->rtp_remote_family;
+ dst->clockrate = src->clockrate;
+ dst->rtcp_received_via_mux = src->rtcp_received_via_mux;
+ dst->estimated_download_bandwidth = src->estimated_download_bandwidth;
}
void _linphone_call_stats_set_ice_state (LinphoneCallStats *stats, LinphoneIceState state) {
@@ -160,6 +171,14 @@ bool_t _linphone_call_stats_rtcp_received_via_mux (const LinphoneCallStats *stat
return stats->rtcp_received_via_mux;
}
+bool_t _linphone_call_stats_has_received_rtcp(const LinphoneCallStats *stats){
+ return stats->received_rtcp != NULL;
+}
+
+bool_t _linphone_call_stats_has_sent_rtcp(const LinphoneCallStats *stats){
+ return stats->sent_rtcp != NULL;
+}
+
// =============================================================================
// Public functions
// =============================================================================
@@ -226,8 +245,10 @@ LinphoneStreamType linphone_call_stats_get_type (const LinphoneCallStats *stats)
float linphone_call_stats_get_sender_loss_rate (const LinphoneCallStats *stats) {
const report_block_t *srb = NULL;
- if (!stats || !stats->sent_rtcp)
+ if (!stats->sent_rtcp){
+ ms_warning("linphone_call_stats_get_sender_loss_rate(): there is no RTCP packet sent.");
return 0.0;
+ }
/* Perform msgpullup() to prevent crashes in rtcp_is_SR() or rtcp_is_RR() if the RTCP packet is composed of several mblk_t structure */
if (stats->sent_rtcp->b_cont != NULL)
msgpullup(stats->sent_rtcp, (size_t)-1);
@@ -248,8 +269,10 @@ float linphone_call_stats_get_sender_loss_rate (const LinphoneCallStats *stats)
float linphone_call_stats_get_receiver_loss_rate (const LinphoneCallStats *stats) {
const report_block_t *rrb = NULL;
- if (!stats || !stats->received_rtcp)
+ if (!stats->received_rtcp){
+ ms_warning("linphone_call_stats_get_receiver_loss_rate(): there is no RTCP packet received.");
return 0.0;
+ }
/* Perform msgpullup() to prevent crashes in rtcp_is_SR() or rtcp_is_RR() if the RTCP packet is composed of several mblk_t structure */
if (stats->received_rtcp->b_cont != NULL)
msgpullup(stats->received_rtcp, (size_t)-1);
@@ -278,8 +301,10 @@ float linphone_call_stats_get_local_late_rate (const LinphoneCallStats *stats) {
float linphone_call_stats_get_sender_interarrival_jitter (const LinphoneCallStats *stats) {
const report_block_t *srb = NULL;
- if (!stats || !stats->sent_rtcp)
+ if (!stats->sent_rtcp){
+ ms_warning("linphone_call_stats_get_sender_interarrival_jitter(): there is no RTCP packet sent.");
return 0.0;
+ }
/* Perform msgpullup() to prevent crashes in rtcp_is_SR() or rtcp_is_RR() if the RTCP packet is composed of several mblk_t structure */
if (stats->sent_rtcp->b_cont != NULL)
msgpullup(stats->sent_rtcp, (size_t)-1);
@@ -297,8 +322,10 @@ float linphone_call_stats_get_sender_interarrival_jitter (const LinphoneCallStat
float linphone_call_stats_get_receiver_interarrival_jitter (const LinphoneCallStats *stats) {
const report_block_t *rrb = NULL;
- if (!stats || !stats->received_rtcp)
+ if (!stats->received_rtcp){
+ ms_warning("linphone_call_stats_get_receiver_interarrival_jitter(): there is no RTCP packet received.");
return 0.0;
+ }
/* Perform msgpullup() to prevent crashes in rtcp_is_SR() or rtcp_is_RR() if the RTCP packet is composed of several mblk_t structure */
if (stats->received_rtcp->b_cont != NULL)
msgpullup(stats->received_rtcp, (size_t)-1);
@@ -364,3 +391,4 @@ float linphone_call_stats_get_estimated_download_bandwidth(const LinphoneCallSta
void linphone_call_stats_set_estimated_download_bandwidth(LinphoneCallStats *stats, float estimated_value) {
stats->estimated_download_bandwidth = estimated_value;
}
+
diff --git a/src/c-wrapper/api/c-call.cpp b/src/c-wrapper/api/c-call.cpp
index 102244410c25480f9062734daab57c5916ac5ab1..d09bc90e14612254dffd74ca90aa52a949fc70a1 100644
--- a/src/c-wrapper/api/c-call.cpp
+++ b/src/c-wrapper/api/c-call.cpp
@@ -29,6 +29,7 @@
#include "call/remote-conference-call.h"
#include "chat/chat-room/real-time-text-chat-room.h"
#include "conference/params/media-session-params-p.h"
+#include "conference/session/ms2-streams.h"
#include "core/core-p.h"
// =============================================================================
@@ -77,7 +78,7 @@ void linphone_call_init_media_streams (LinphoneCall *call) {
/*This function is not static because used internally in linphone-daemon project*/
void _post_configure_audio_stream (AudioStream *st, LinphoneCore *lc, bool_t muted) {
- L_GET_PRIVATE_FROM_C_OBJECT(lc)->postConfigureAudioStream(st, !!muted);
+ LinphonePrivate::MS2AudioStream::postConfigureAudioStream(st, lc, !!muted);
}
void linphone_call_stop_media_streams (LinphoneCall *call) {
diff --git a/src/c-wrapper/internal/c-sal.cpp b/src/c-wrapper/internal/c-sal.cpp
index e82bb534934d08545b26cd5becfb7f8265d80a0b..8f6397e9e1ce3bd007733b4b638c6e1927bd4070 100644
--- a/src/c-wrapper/internal/c-sal.cpp
+++ b/src/c-wrapper/internal/c-sal.cpp
@@ -66,6 +66,28 @@ SalTransport sal_transport_parse(const char* param) {
return SalTransportUDP;
}
+
+SalStreamBundle *sal_stream_bundle_new(void){
+ return ms_new0(SalStreamBundle, 1);
+}
+
+void sal_stream_bundle_add_stream(SalStreamBundle *bundle, SalStreamDescription *stream, const char *mid){
+ strncpy(stream->mid, mid ? mid : "", sizeof(stream->mid));
+ stream->mid[sizeof(stream->mid) -1] = '\0';
+ bundle->mids = bctbx_list_append(bundle->mids, ms_strdup(mid));
+}
+
+void sal_stream_bundle_destroy(SalStreamBundle *bundle){
+ bctbx_list_free_with_data(bundle->mids, (void (*)(void*)) ms_free);
+ ms_free(bundle);
+}
+
+SalStreamBundle *sal_stream_bundle_clone(const SalStreamBundle *bundle){
+ SalStreamBundle *ret = sal_stream_bundle_new();
+ ret->mids = bctbx_list_copy_with_data(bundle->mids, (bctbx_list_copy_func)bctbx_strdup);
+ return ret;
+}
+
SalMediaDescription *sal_media_description_new(){
SalMediaDescription *md=ms_new0(SalMediaDescription,1);
int i;
@@ -79,6 +101,65 @@ SalMediaDescription *sal_media_description_new(){
return md;
}
+SalStreamBundle * sal_media_description_add_new_bundle(SalMediaDescription *md){
+ SalStreamBundle *bundle = sal_stream_bundle_new();
+ md->bundles = bctbx_list_append(md->bundles, bundle);
+ return bundle;
+}
+
+int sal_stream_bundle_has_mid(const SalStreamBundle *bundle, const char *mid){
+ const bctbx_list_t *elem;
+ for (elem = bundle->mids; elem != NULL; elem = elem->next){
+ const char *m = (const char *) elem->data;
+ if (strcmp(m, mid) == 0) return TRUE;
+ }
+ return FALSE;
+}
+
+
+int sal_media_description_lookup_mid(const SalMediaDescription *md, const char *mid){
+ int index;
+ for (index = 0 ; index < md->nb_streams; ++index){
+ const SalStreamDescription * sd = &md->streams[index];
+ if (strcmp(sd->mid, mid) == 0){
+ return index;
+ }
+ }
+ return -1;
+}
+
+const SalStreamBundle *sal_media_description_get_bundle_from_mid(const SalMediaDescription *md, const char *mid){
+ const bctbx_list_t *elem;
+ for (elem = md->bundles; elem != NULL; elem = elem->next){
+ SalStreamBundle *bundle = (SalStreamBundle *)elem->data;
+ if (sal_stream_bundle_has_mid(bundle, mid)) return bundle;
+ }
+ return NULL;
+}
+
+const char *sal_stream_bundle_get_mid_of_transport_owner(const SalStreamBundle *bundle){
+ return (const char*)bundle->mids->data; /* the first one is the transport owner*/
+}
+
+int sal_media_description_get_index_of_transport_owner(const SalMediaDescription *md, const SalStreamDescription *sd){
+ const SalStreamBundle *bundle;
+ const char *master_mid;
+ int index;
+ if (sd->mid[0] == '\0') return -1; /* not part of any bundle */
+ /* lookup the mid in the bundle descriptions */
+ bundle = sal_media_description_get_bundle_from_mid(md, sd->mid);
+ if (!bundle) {
+ ms_warning("Orphan stream with mid '%s'", sd->mid);
+ return -1;
+ }
+ master_mid = sal_stream_bundle_get_mid_of_transport_owner(bundle);
+ index = sal_media_description_lookup_mid(md, master_mid);
+ if (index == -1){
+ ms_error("Stream with mid '%s' has no transport owner (mid '%s') !", sd->mid, master_mid);
+ }
+ return index;
+}
+
static void sal_media_description_destroy(SalMediaDescription *md){
int i;
for(i=0;i<SAL_MEDIA_DESCRIPTION_MAX_STREAMS;i++){
@@ -88,6 +169,7 @@ static void sal_media_description_destroy(SalMediaDescription *md){
md->streams[i].already_assigned_payloads=NULL;
sal_custom_sdp_attribute_free(md->streams[i].custom_sdp_attributes);
}
+ bctbx_list_free_with_data(md->bundles, (void (*)(void*)) sal_stream_bundle_destroy);
sal_custom_sdp_attribute_free(md->custom_sdp_attributes);
ms_free(md);
}
@@ -108,7 +190,7 @@ SalStreamDescription *sal_media_description_find_stream(SalMediaDescription *md,
int i;
for(i=0;i<SAL_MEDIA_DESCRIPTION_MAX_STREAMS;++i){
SalStreamDescription *ss=&md->streams[i];
- if (!sal_stream_description_active(ss)) continue;
+ if (!sal_stream_description_enabled(ss)) continue;
if (ss->proto==proto && ss->type==type) return ss;
}
return NULL;
@@ -118,7 +200,7 @@ unsigned int sal_media_description_nb_active_streams_of_type(SalMediaDescription
unsigned int i;
unsigned int nb = 0;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; ++i) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (md->streams[i].type == type) nb++;
}
return nb;
@@ -127,7 +209,7 @@ unsigned int sal_media_description_nb_active_streams_of_type(SalMediaDescription
SalStreamDescription * sal_media_description_get_active_stream_of_type(SalMediaDescription *md, SalStreamType type, unsigned int idx) {
unsigned int i;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; ++i) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (md->streams[i].type == type) {
if (idx-- == 0) return &md->streams[i];
}
@@ -160,7 +242,7 @@ void sal_media_description_set_dir(SalMediaDescription *md, SalStreamDir stream_
int i;
for(i=0;i<SAL_MEDIA_DESCRIPTION_MAX_STREAMS;++i){
SalStreamDescription *ss=&md->streams[i];
- if (!sal_stream_description_active(ss)) continue;
+ if (!sal_stream_description_enabled(ss)) continue;
ss->dir=stream_dir;
}
}
@@ -169,7 +251,7 @@ int sal_media_description_get_nb_active_streams(const SalMediaDescription *md) {
int i;
int nb = 0;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (sal_stream_description_active(&md->streams[i])) nb++;
+ if (sal_stream_description_enabled(&md->streams[i])) nb++;
}
return nb;
}
@@ -185,7 +267,7 @@ static bool_t has_dir(const SalMediaDescription *md, SalStreamDir stream_dir){
/* we are looking for at least one stream with requested direction, inactive streams are ignored*/
for(i=0;i<SAL_MEDIA_DESCRIPTION_MAX_STREAMS;++i){
const SalStreamDescription *ss=&md->streams[i];
- if (!sal_stream_description_active(ss)) continue;
+ if (!sal_stream_description_enabled(ss)) continue;
if (ss->dir==stream_dir) {
return TRUE;
}
@@ -213,8 +295,16 @@ bool_t sal_media_description_has_dir(const SalMediaDescription *md, SalStreamDir
return FALSE;
}
-bool_t sal_stream_description_active(const SalStreamDescription *sd) {
- return (sd->rtp_port > 0);
+bool_t sal_stream_description_enabled(const SalStreamDescription *sd) {
+ /* When the bundle-only attribute is present, a 0 rtp port doesn't mean that the stream is disabled.*/
+ return sd->rtp_port > 0 || sd->bundle_only;
+}
+
+void sal_stream_description_disable(SalStreamDescription *sd){
+ sd->rtp_port = 0;
+ /* Remove potential bundle parameters. A disabled stream is moved out of the bundle. */
+ sd->mid[0] = '\0';
+ sd->bundle_only = FALSE;
}
/*these are switch case, so that when a new proto is added we can't forget to modify this function*/
@@ -280,7 +370,7 @@ bool_t sal_media_description_has_avpf(const SalMediaDescription *md) {
int i;
if (md->nb_streams == 0) return FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (sal_stream_description_has_avpf(&md->streams[i]) != TRUE) return FALSE;
}
return TRUE;
@@ -290,7 +380,7 @@ bool_t sal_media_description_has_implicit_avpf(const SalMediaDescription *md) {
int i;
if (md->nb_streams == 0) return FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (sal_stream_description_has_implicit_avpf(&md->streams[i]) != TRUE) return FALSE;
}
return TRUE;
@@ -300,17 +390,17 @@ bool_t sal_media_description_has_srtp(const SalMediaDescription *md) {
int i;
if (md->nb_streams == 0) return FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
- if (sal_stream_description_has_srtp(&md->streams[i]) != TRUE) return FALSE;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
+ if (sal_stream_description_has_srtp(&md->streams[i])) return TRUE;
}
- return TRUE;
+ return FALSE;
}
bool_t sal_media_description_has_dtls(const SalMediaDescription *md) {
int i;
if (md->nb_streams == 0) return FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (sal_stream_description_has_dtls(&md->streams[i]) != TRUE) return FALSE;
}
return TRUE;
@@ -320,7 +410,7 @@ bool_t sal_media_description_has_zrtp(const SalMediaDescription *md) {
int i;
if (md->nb_streams == 0) return FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (sal_stream_description_has_zrtp(&md->streams[i]) != TRUE) return FALSE;
}
return TRUE;
@@ -330,7 +420,7 @@ bool_t sal_media_description_has_ipv6(const SalMediaDescription *md){
int i;
if (md->nb_streams == 0) return FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md->streams[i])) continue;
if (md->streams[i].rtp_addr[0] != '\0'){
if (!sal_stream_description_has_ipv6(&md->streams[i])) return FALSE;
}else{
@@ -411,6 +501,7 @@ int sal_stream_description_equals(const SalStreamDescription *sd1, const SalStre
if (strcmp(sd1->rtp_addr, sd2->rtp_addr) != 0) result |= SAL_MEDIA_DESCRIPTION_NETWORK_CHANGED;
if (sd1->rtp_addr[0]!='\0' && sd2->rtp_addr[0]!='\0' && ms_is_multicast(sd1->rtp_addr) != ms_is_multicast(sd2->rtp_addr))
result |= SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED;
+ if (sd1->multicast_role != sd2->multicast_role) result |= SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED;
if (sd1->rtp_port != sd2->rtp_port) {
if ((sd1->rtp_port == 0) || (sd2->rtp_port == 0)) result |= SAL_MEDIA_DESCRIPTION_CODEC_CHANGED;
else result |= SAL_MEDIA_DESCRIPTION_NETWORK_CHANGED;
@@ -480,7 +571,7 @@ int sal_media_description_equals(const SalMediaDescription *md1, const SalMediaD
int i;
for(i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; ++i){
- if (!sal_stream_description_active(&md1->streams[i]) && !sal_stream_description_active(&md2->streams[i])) continue;
+ if (!sal_stream_description_enabled(&md1->streams[i]) && !sal_stream_description_enabled(&md2->streams[i])) continue;
result |= sal_stream_description_equals(&md1->streams[i], &md2->streams[i]);
}
return result;
diff --git a/src/c-wrapper/internal/c-sal.h b/src/c-wrapper/internal/c-sal.h
index 998d9ed54b8da4ec7688722777f81893df710d07..98c542d63d146b66565ba080956e8ef13a7119bb 100644
--- a/src/c-wrapper/internal/c-sal.h
+++ b/src/c-wrapper/internal/c-sal.h
@@ -243,6 +243,12 @@ typedef enum {
SalOpSDPSimulateRemove /** Will simulate no SDP in the op */
} SalOpSDPHandling;
+#define SAL_STREAM_DESCRIPTION_PORT_TO_BE_DETERMINED 65536
+
+typedef struct SalStreamBundle{
+ bctbx_list_t *mids; /* List of mids corresponding to streams associated in the bundle. The first one is the "tagged" one. */
+} SalStreamBundle;
+
typedef struct SalStreamDescription{
char name[16]; /*unique name of stream, in order to ease offer/answer model algorithm*/
SalMediaProto proto;
@@ -264,7 +270,9 @@ typedef struct SalStreamDescription{
SalSrtpCryptoAlgo crypto[SAL_CRYPTO_ALGO_MAX];
unsigned int crypto_local_tag;
int max_rate;
+ bool_t bundle_only;
bool_t implicit_rtcp_fb;
+ bool_t pad[2]; /* Use me */
OrtpRtcpFbConfiguration rtcp_fb;
OrtpRtcpXrConfiguration rtcp_xr;
SalCustomSdpAttribute *custom_sdp_attributes;
@@ -272,14 +280,15 @@ typedef struct SalStreamDescription{
SalIceRemoteCandidate ice_remote_candidates[SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES];
char ice_ufrag[SAL_MEDIA_DESCRIPTION_MAX_ICE_UFRAG_LEN];
char ice_pwd[SAL_MEDIA_DESCRIPTION_MAX_ICE_PWD_LEN];
+ char mid[32]; /* Media line identifier for RTP bundle mode */
+ int mid_rtp_ext_header_id; /* Identifier for the MID field in the RTP extension header */
bool_t ice_mismatch;
bool_t set_nortpproxy; /*Formely set by ICE to indicate to the proxy that it has nothing to do*/
bool_t rtcp_mux;
- bool_t pad[1];
+ uint8_t haveZrtpHash; /**< flag for zrtp hash presence */
+ uint8_t zrtphash[128];
char dtls_fingerprint[256];
SalDtlsRole dtls_role;
- uint8_t zrtphash[128];
- uint8_t haveZrtpHash; /**< flag for zrtp hash presence */
int ttl; /*for multicast -1 to disable*/
SalMulticastRole multicast_role;
} SalStreamDescription;
@@ -313,9 +322,11 @@ typedef struct SalMediaDescription{
OrtpRtcpXrConfiguration rtcp_xr;
char ice_ufrag[SAL_MEDIA_DESCRIPTION_MAX_ICE_UFRAG_LEN];
char ice_pwd[SAL_MEDIA_DESCRIPTION_MAX_ICE_PWD_LEN];
+ bctbx_list_t *bundles; /* list of SalStreamBundle */
bool_t ice_lite;
bool_t set_nortpproxy;
- bool_t pad[2];
+ bool_t accept_bundles; /* Set to TRUE if RTP bundles can be accepted during offer answer. This field has no appearance on the SDP.*/
+ bool_t pad[1];
} SalMediaDescription;
typedef struct SalMessage{
@@ -348,7 +359,12 @@ SalStreamDescription * sal_media_description_find_secure_stream_of_type(SalMedia
SalStreamDescription * sal_media_description_find_best_stream(SalMediaDescription *md, SalStreamType type);
void sal_media_description_set_dir(SalMediaDescription *md, SalStreamDir stream_dir);
int sal_stream_description_equals(const SalStreamDescription *sd1, const SalStreamDescription *sd2);
-bool_t sal_stream_description_active(const SalStreamDescription *sd);
+
+
+/* Enabled means that the stream exists and is accepted as part of the session: the port value is non-zero or the stream has bundle-only attribute.
+ *However, it may be marked with a=inactive, which is unrelated to the return value of this function.*/
+bool_t sal_stream_description_enabled(const SalStreamDescription *sd);
+void sal_stream_description_disable(SalStreamDescription *sd);
bool_t sal_stream_description_has_avpf(const SalStreamDescription *sd);
bool_t sal_stream_description_has_implicit_avpf(const SalStreamDescription *sd);
bool_t sal_stream_description_has_srtp(const SalStreamDescription *sd);
@@ -362,6 +378,18 @@ bool_t sal_media_description_has_zrtp(const SalMediaDescription *md);
bool_t sal_media_description_has_ipv6(const SalMediaDescription *md);
int sal_media_description_get_nb_active_streams(const SalMediaDescription *md);
+SalStreamBundle * sal_media_description_add_new_bundle(SalMediaDescription *md);
+/* Add stream to the bundle. The SalStreamDescription must be part of the SalMediaDescription in which the SalStreamBundle is added. */
+void sal_stream_bundle_add_stream(SalStreamBundle *bundle, SalStreamDescription *stream, const char *mid);
+void sal_stream_bundle_destroy(SalStreamBundle *bundle);
+SalStreamBundle *sal_stream_bundle_clone(const SalStreamBundle *bundle);
+int sal_stream_bundle_has_mid(const SalStreamBundle *bundle, const char *mid);
+const char *sal_stream_bundle_get_mid_of_transport_owner(const SalStreamBundle *bundle);
+
+int sal_media_description_lookup_mid(const SalMediaDescription *md, const char *mid);
+int sal_media_description_get_index_of_transport_owner(const SalMediaDescription *md, const SalStreamDescription *sd);
+
+
#ifdef __cplusplus
}
#endif
diff --git a/src/call/call-p.h b/src/call/call-p.h
index c2f52fba1f320d177d2a86bb2245617a11c1251d..93971d5fde3aded2fb0648878d32782f924bdfde 100644
--- a/src/call/call-p.h
+++ b/src/call/call-p.h
@@ -55,6 +55,7 @@ public:
unsigned int getAudioStartCount () const;
unsigned int getVideoStartCount () const;
unsigned int getTextStartCount () const;
+ // don't make new code relying on this method.
MediaStream *getMediaStream (LinphoneStreamType type) const;
SalCallOp *getOp () const;
@@ -96,7 +97,7 @@ private:
void onIncomingCallSessionStarted (const std::shared_ptr<CallSession> &session) override;
void onIncomingCallSessionTimeoutCheck (const std::shared_ptr<CallSession> &session, int elapsed, bool oneSecondElapsed) override;
void onInfoReceived (const std::shared_ptr<CallSession> &session, const LinphoneInfoMessage *im) override;
- void onNoMediaTimeoutCheck (const std::shared_ptr<CallSession> &session, bool oneSecondElapsed) override;
+ void onLossOfMediaDetected (const std::shared_ptr<CallSession> &session) override;
void onEncryptionChanged (const std::shared_ptr<CallSession> &session, bool activated, const std::string &authToken) override;
void onCallSessionStateChangedForReporting (const std::shared_ptr<CallSession> &session) override;
void onRtcpUpdateForReporting (const std::shared_ptr<CallSession> &session, SalStreamType type) override;
diff --git a/src/call/call.cpp b/src/call/call.cpp
index a5925302e3d648bc49153ff5b7cc11601f4a258f..c23326c81a3cd5b9bdd83a2fe1c3e37da332352f 100644
--- a/src/call/call.cpp
+++ b/src/call/call.cpp
@@ -47,8 +47,10 @@ LinphoneProxyConfig *CallPrivate::getDestProxy () const {
return getActiveSession()->getPrivate()->getDestProxy();
}
+/* This a test-only method.*/
IceSession *CallPrivate::getIceSession () const {
return static_pointer_cast<MediaSession>(getActiveSession())->getPrivate()->getIceSession();
+ return nullptr;
}
unsigned int CallPrivate::getAudioStartCount () const {
@@ -64,7 +66,27 @@ unsigned int CallPrivate::getTextStartCount () const {
}
MediaStream *CallPrivate::getMediaStream (LinphoneStreamType type) const {
- return static_pointer_cast<MediaSession>(getActiveSession())->getPrivate()->getMediaStream(type);
+ auto ms = static_pointer_cast<MediaSession>(getActiveSession())->getPrivate();
+ StreamsGroup & sg = ms->getStreamsGroup();
+ MS2Stream *s = nullptr;
+ switch(type){
+ case LinphoneStreamTypeAudio:
+ s = sg.lookupMainStreamInterface<MS2Stream>(SalAudio);
+ break;
+ case LinphoneStreamTypeVideo:
+ s = sg.lookupMainStreamInterface<MS2Stream>(SalVideo);
+ break;
+ case LinphoneStreamTypeText:
+ s = sg.lookupMainStreamInterface<MS2Stream>(SalText);
+ break;
+ default:
+ break;
+ }
+ if (!s){
+ lError() << "CallPrivate::getMediaStream() : no stream with type " << type;
+ return nullptr;
+ }
+ return s->getMediaStream();
}
SalCallOp * CallPrivate::getOp () const {
@@ -129,7 +151,7 @@ shared_ptr<Call> CallPrivate::startReferredCall (const MediaSessionParams *param
if (params)
msp = *params;
else {
- msp.initDefault(q->getCore());
+ msp.initDefault(q->getCore(), LinphoneCallOutgoing);
msp.enableAudio(q->getCurrentParams()->audioEnabled());
msp.enableVideo(q->getCurrentParams()->videoEnabled());
}
@@ -155,7 +177,7 @@ void CallPrivate::createPlayer () const {
// -----------------------------------------------------------------------------
void CallPrivate::initializeMediaStreams () {
- static_pointer_cast<MediaSession>(getActiveSession())->getPrivate()->initializeStreams();
+
}
void CallPrivate::stopMediaStreams () {
@@ -172,13 +194,12 @@ void CallPrivate::startRemoteRing () {
return;
MSSndCard *ringCard = lc->sound_conf.lsd_card ? lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
- int maxRate = static_pointer_cast<MediaSession>(getActiveSession())->getPrivate()->getLocalDesc()->streams[0].max_rate;
- if (maxRate > 0)
- ms_snd_card_set_preferred_sample_rate(ringCard, maxRate);
- // We release sound before playing ringback tone
- AudioStream *as = reinterpret_cast<AudioStream *>(getMediaStream(LinphoneStreamTypeAudio));
- if (as)
- audio_stream_unprepare_sound(as);
+ SalMediaDescription *md = static_pointer_cast<MediaSession>(getActiveSession())->getPrivate()->getLocalDesc();
+ if (md){
+ int maxRate = md->streams[0].max_rate;
+ if (maxRate > 0)
+ ms_snd_card_set_preferred_sample_rate(ringCard, maxRate);
+ }
if (lc->sound_conf.remote_ring) {
ms_snd_card_set_stream_type(ringCard, MS_SND_CARD_STREAM_VOICE);
lc->ringstream = ring_start(lc->factory, lc->sound_conf.remote_ring, 2000, ringCard);
@@ -363,16 +384,8 @@ void CallPrivate::onInfoReceived (const shared_ptr<CallSession> &session, const
linphone_call_notify_info_message_received(L_GET_C_BACK_PTR(q), im);
}
-void CallPrivate::onNoMediaTimeoutCheck (const shared_ptr<CallSession> &session, bool oneSecondElapsed) {
- L_Q();
- int disconnectTimeout = linphone_core_get_nortp_timeout(q->getCore()->getCCore());
- bool disconnected = false;
- AudioStream *as = reinterpret_cast<AudioStream *>(getMediaStream(LinphoneStreamTypeAudio));
- if (((q->getState() == CallSession::State::StreamsRunning) || (q->getState() == CallSession::State::PausedByRemote))
- && oneSecondElapsed && as && (as->ms.state == MSStreamStarted) && (disconnectTimeout > 0))
- disconnected = !audio_stream_alive(as, disconnectTimeout);
- if (disconnected)
- terminateBecauseOfLostMedia();
+void CallPrivate::onLossOfMediaDetected (const shared_ptr<CallSession> &session) {
+ terminateBecauseOfLostMedia();
}
void CallPrivate::onEncryptionChanged (const shared_ptr<CallSession> &session, bool activated, const string &authToken) {
diff --git a/src/call/call.h b/src/call/call.h
index 19946f99dc641f6cfd3c6ccc5306ee0ac7161bf1..667a1e651ba04d84d0925ae778bba54cbea6d6f3 100644
--- a/src/call/call.h
+++ b/src/call/call.h
@@ -40,6 +40,7 @@ class LINPHONE_PUBLIC Call : public Object, public CoreAccessor {
friend class ChatMessagePrivate;
friend class CorePrivate;
friend class MediaSessionPrivate;
+ friend class Stream;
public:
L_OVERRIDE_SHARED_FROM_THIS(Call);
diff --git a/src/conference/params/call-session-params.cpp b/src/conference/params/call-session-params.cpp
index 20b94bee3381968239988de6d99d02839d8c9c2f..4c7a3b786172167c8e5c39fccaf3bdcd9e8038f1 100644
--- a/src/conference/params/call-session-params.cpp
+++ b/src/conference/params/call-session-params.cpp
@@ -87,7 +87,7 @@ CallSessionParams &CallSessionParams::operator= (const CallSessionParams &other)
// -----------------------------------------------------------------------------
-void CallSessionParams::initDefault (const std::shared_ptr<Core> &core) {
+void CallSessionParams::initDefault (const std::shared_ptr<Core> &core, LinphoneCallDir dir) {
L_D();
d->inConference = false;
d->privacy = LinphonePrivacyDefault;
diff --git a/src/conference/params/call-session-params.h b/src/conference/params/call-session-params.h
index 23c1855880cd3377db98a1a0196265da3a9e63d9..23da09de094893d8de8277d4db8904ad47e2ebda 100644
--- a/src/conference/params/call-session-params.h
+++ b/src/conference/params/call-session-params.h
@@ -50,7 +50,7 @@ public:
CallSessionParams &operator= (const CallSessionParams &other);
- virtual void initDefault (const std::shared_ptr<Core> &core);
+ virtual void initDefault (const std::shared_ptr<Core> &core, LinphoneCallDir dir);
const std::string& getSessionName () const;
void setSessionName (const std::string &sessionName);
diff --git a/src/conference/params/media-session-params-p.h b/src/conference/params/media-session-params-p.h
index 603b0b7e791bc6b42bd6d64fc6c31927cd65a676..8fd29cbf043a89dbb77944e8e265377e9b399d5d 100644
--- a/src/conference/params/media-session-params-p.h
+++ b/src/conference/params/media-session-params-p.h
@@ -55,8 +55,12 @@ public:
void setDownPtime (int value) { downPtime = value; }
int getUpPtime () const { return upPtime; }
void setUpPtime (int value) { upPtime = value; }
- bool getUpdateCallWhenIceCompleted () const { return updateCallWhenIceCompleted; }
- void setUpdateCallWhenIceCompleted (bool value) { updateCallWhenIceCompleted = value; }
+ bool getUpdateCallWhenIceCompleted () const;
+ void setUpdateCallWhenIceCompleted(bool value){
+ /* apply to both case when set explicitely */
+ updateCallWhenIceCompleted = value;
+ updateCallWhenIceCompletedWithDTLS = value;
+ }
void setReceivedFps (float value) { receivedFps = value; }
void setReceivedVideoDefinition (LinphoneVideoDefinition *value);
@@ -102,6 +106,8 @@ public:
LinphoneMediaEncryption encryption = LinphoneMediaEncryptionNone;
bool mandatoryMediaEncryptionEnabled = false;
+
+ bool rtpBundle = false;
private:
bool _implicitRtcpFbEnabled = false;
@@ -110,6 +116,7 @@ private:
int downPtime = 0;
int upPtime = 0;
bool updateCallWhenIceCompleted = true;
+ bool updateCallWhenIceCompletedWithDTLS = false;
SalCustomSdpAttribute *customSdpAttributes = nullptr;
SalCustomSdpAttribute *customSdpMediaAttributes[LinphoneStreamTypeUnknown];
diff --git a/src/conference/params/media-session-params.cpp b/src/conference/params/media-session-params.cpp
index bdfd07346b9a6d5b2fd2c37dd8cccef312b47e18..4df4dca583b350d744042493148b6dfa8fab3b59 100644
--- a/src/conference/params/media-session-params.cpp
+++ b/src/conference/params/media-session-params.cpp
@@ -71,6 +71,7 @@ void MediaSessionParamsPrivate::clone (const MediaSessionParamsPrivate *src) {
if (src->customSdpMediaAttributes[i])
customSdpMediaAttributes[i] = sal_custom_sdp_attribute_clone(src->customSdpMediaAttributes[i]);
}
+ rtpBundle = src->rtpBundle;
}
void MediaSessionParamsPrivate::clean () {
@@ -197,6 +198,13 @@ void MediaSessionParamsPrivate::setCustomSdpMediaAttributes (LinphoneStreamType
customSdpMediaAttributes[lst] = sal_custom_sdp_attribute_clone(csa);
}
+bool MediaSessionParamsPrivate::getUpdateCallWhenIceCompleted() const{
+ if (encryption == LinphoneMediaEncryptionDTLS){
+ return updateCallWhenIceCompletedWithDTLS;
+ }
+ return updateCallWhenIceCompleted;
+}
+
// =============================================================================
MediaSessionParams::MediaSessionParams () : CallSessionParams(*new MediaSessionParamsPrivate) {
@@ -225,15 +233,20 @@ MediaSessionParams &MediaSessionParams::operator= (const MediaSessionParams &oth
// -----------------------------------------------------------------------------
-void MediaSessionParams::initDefault (const std::shared_ptr<Core> &core) {
+void MediaSessionParams::initDefault (const std::shared_ptr<Core> &core, LinphoneCallDir dir) {
L_D();
- CallSessionParams::initDefault(core);
+ CallSessionParams::initDefault(core, dir);
LinphoneCore *cCore = core->getCCore();
d->audioEnabled = true;
- d->videoEnabled = linphone_core_video_enabled(cCore) && cCore->video_policy.automatically_initiate;
- if (!linphone_core_video_enabled(cCore) && cCore->video_policy.automatically_initiate) {
+ if (dir == LinphoneCallOutgoing){
+ d->videoEnabled = cCore->video_policy.automatically_initiate;
+ }else{
+ d->videoEnabled = cCore->video_policy.automatically_accept;
+ }
+ if (!linphone_core_video_enabled(cCore) && d->videoEnabled) {
lError() << "LinphoneCore has video disabled for both capture and display, but video policy is to start the call with video. "
"This is a possible mis-use of the API. In this case, video is disabled in default LinphoneCallParams";
+ d->videoEnabled = false;
}
d->realtimeTextEnabled = !!linphone_core_realtime_text_enabled(cCore);
d->realtimeTextKeepaliveInterval = linphone_core_realtime_text_get_keepalive_interval(cCore);
@@ -247,7 +260,16 @@ void MediaSessionParams::initDefault (const std::shared_ptr<Core> &core) {
d->audioMulticastEnabled = !!linphone_core_audio_multicast_enabled(cCore);
d->videoMulticastEnabled = !!linphone_core_video_multicast_enabled(cCore);
d->updateCallWhenIceCompleted = !!lp_config_get_int(linphone_core_get_config(cCore), "sip", "update_call_when_ice_completed", true);
+ /*
+ * At the time of WebRTC/JSSIP interoperability tests, it was found that the ICE re-INVITE was breaking communication.
+ * The update_call_when_ice_completed_with_dtls property is hence set to false.
+ * If this is no longer the case it should be changed to true.
+ * Otherwise an application may decide to set to true as ICE reINVITE is mandatory per ICE RFC and unless from this WebRTC interoperability standpoint
+ * there is no problem in having the ICE re-INVITE to be done when SRTP-DTLS is used.
+ */
+ d->updateCallWhenIceCompletedWithDTLS = linphone_config_get_bool(linphone_core_get_config(cCore), "sip", "update_call_when_ice_completed_with_dtls", false);
d->mandatoryMediaEncryptionEnabled = !!linphone_core_is_media_encryption_mandatory(cCore);
+ d->rtpBundle = linphone_core_rtp_bundle_enabled(cCore);
}
// -----------------------------------------------------------------------------
@@ -265,8 +287,8 @@ bool MediaSessionParams::audioMulticastEnabled () const {
void MediaSessionParams::enableAudio (bool value) {
L_D();
d->audioEnabled = value;
- if (d->audioEnabled && (getAudioDirection() == LinphoneMediaDirectionInactive))
- setAudioDirection(LinphoneMediaDirectionSendRecv);
+ //if (d->audioEnabled && (getAudioDirection() == LinphoneMediaDirectionInactive))
+ // setAudioDirection(LinphoneMediaDirectionSendRecv);
}
void MediaSessionParams::enableAudioMulticast (bool value) {
@@ -309,8 +331,8 @@ void MediaSessionParams::setAudioDirection (LinphoneMediaDirection direction) {
void MediaSessionParams::enableVideo (bool value) {
L_D();
d->videoEnabled = value;
- if (d->videoEnabled && (getVideoDirection() == LinphoneMediaDirectionInactive))
- setVideoDirection(LinphoneMediaDirectionSendRecv);
+ //if (d->videoEnabled && (getVideoDirection() == LinphoneMediaDirectionInactive))
+ // setVideoDirection(LinphoneMediaDirectionSendRecv);
}
void MediaSessionParams::enableVideoMulticast (bool value) {
@@ -529,4 +551,14 @@ const char * MediaSessionParams::getCustomSdpMediaAttribute (LinphoneStreamType
return sal_custom_sdp_attribute_find(d->customSdpMediaAttributes[lst], attributeName.c_str());
}
+void MediaSessionParams::enableRtpBundle(bool value){
+ L_D();
+ d->rtpBundle = value;
+}
+
+bool MediaSessionParams::rtpBundleEnabled()const{
+ L_D();
+ return d->rtpBundle;
+}
+
LINPHONE_END_NAMESPACE
diff --git a/src/conference/params/media-session-params.h b/src/conference/params/media-session-params.h
index 8bb3c2793aabf520c3c0c20d35d7143a6d6a813c..e57cc2a3e75edc0970335b9cfc3a49d37f97e7af 100644
--- a/src/conference/params/media-session-params.h
+++ b/src/conference/params/media-session-params.h
@@ -35,6 +35,10 @@ class MediaSessionParamsPrivate;
class MediaSessionParams : public CallSessionParams {
friend class MediaSession;
friend class MediaSessionPrivate;
+ friend class MS2Stream;
+ friend class MS2AudioStream;
+ friend class MS2VideoStream;
+ friend class MS2RTTStream;
public:
MediaSessionParams ();
@@ -47,7 +51,7 @@ public:
MediaSessionParams &operator= (const MediaSessionParams &other);
- void initDefault (const std::shared_ptr<Core> &core) override;
+ void initDefault (const std::shared_ptr<Core> &core, LinphoneCallDir dir) override;
bool audioEnabled () const;
bool audioMulticastEnabled () const;
@@ -109,6 +113,9 @@ public:
void addCustomSdpMediaAttribute (LinphoneStreamType lst, const std::string &attributeName, const std::string &attributeValue);
void clearCustomSdpMediaAttributes (LinphoneStreamType lst);
const char * getCustomSdpMediaAttribute (LinphoneStreamType lst, const std::string &attributeName) const;
+
+ void enableRtpBundle(bool value);
+ bool rtpBundleEnabled()const;
private:
L_DECLARE_PRIVATE(MediaSessionParams);
diff --git a/src/conference/session/audio-stream.cpp b/src/conference/session/audio-stream.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..57205fdeb2bda509eae7928372a5e3cca1c90f47
--- /dev/null
+++ b/src/conference/session/audio-stream.cpp
@@ -0,0 +1,713 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+#include "bctoolbox/defs.h"
+
+#include "ms2-streams.h"
+#include "media-session.h"
+#include "media-session-p.h"
+#include "core/core.h"
+#include "c-wrapper/c-wrapper.h"
+#include "call/call.h"
+#include "call/call-p.h"
+#include "conference/participant.h"
+#include "conference/params/media-session-params-p.h"
+#include "nat/ice-service.h"
+
+#include "mediastreamer2/msfileplayer.h"
+#include "mediastreamer2/msvolume.h"
+
+#include "linphone/core.h"
+
+#include <cmath>
+
+using namespace::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+/*
+ * MS2AudioStream implementation.
+ */
+
+MS2AudioStream::MS2AudioStream(StreamsGroup &sg, const OfferAnswerContext ¶ms) : MS2Stream(sg, params){
+ string bindIp = getBindIp();
+ mStream = audio_stream_new2(getCCore()->factory, bindIp.empty() ? nullptr : bindIp.c_str(), mPortConfig.rtpPort, mPortConfig.rtcpPort);
+
+ /* Initialize zrtp even if we didn't explicitely set it, just in case peer offers it */
+ if (linphone_core_media_encryption_supported(getCCore(), LinphoneMediaEncryptionZRTP)) {
+ LinphoneCallLog *log = getMediaSession().getLog();
+ const LinphoneAddress *peerAddr = linphone_call_log_get_remote_address(log);
+ const LinphoneAddress *selfAddr = linphone_call_log_get_local_address(log);
+ char *peerUri = ms_strdup_printf("%s:%s@%s" , linphone_address_get_scheme(peerAddr)
+ , linphone_address_get_username(peerAddr)
+ , linphone_address_get_domain(peerAddr));
+ char *selfUri = ms_strdup_printf("%s:%s@%s" , linphone_address_get_scheme(selfAddr)
+ , linphone_address_get_username(selfAddr)
+ , linphone_address_get_domain(selfAddr));
+
+ MSZrtpParams zrtpParams;
+ zrtpCacheAccess zrtpCacheInfo = linphone_core_get_zrtp_cache_access(getCCore());
+
+ memset(&zrtpParams, 0, sizeof(MSZrtpParams));
+ /* media encryption of current params will be set later when zrtp is activated */
+ zrtpParams.zidCacheDB = zrtpCacheInfo.db;
+ zrtpParams.zidCacheDBMutex = zrtpCacheInfo.dbMutex;
+ zrtpParams.peerUri = peerUri;
+ zrtpParams.selfUri = selfUri;
+ /* Get key lifespan from config file, default is 0:forever valid */
+ zrtpParams.limeKeyTimeSpan = bctbx_time_string_to_sec(lp_config_get_string(linphone_core_get_config(getCCore()), "sip", "lime_key_validity", "0"));
+ setZrtpCryptoTypesParameters(&zrtpParams, params.remoteStreamDescription ? params.remoteStreamDescription->haveZrtpHash : false);
+ audio_stream_enable_zrtp(mStream, &zrtpParams);
+ if (peerUri)
+ ms_free(peerUri);
+ if (selfUri)
+ ms_free(selfUri);
+ }
+ initializeSessions((MediaStream*)mStream);
+}
+
+void MS2AudioStream::setZrtpCryptoTypesParameters(MSZrtpParams *params, bool haveRemoteZrtpHash) {
+ const MSCryptoSuite *srtpSuites = linphone_core_get_srtp_crypto_suites(getCCore());
+ if (srtpSuites) {
+ for(int i = 0; (srtpSuites[i] != MS_CRYPTO_SUITE_INVALID) && (i < SAL_CRYPTO_ALGO_MAX) && (i < MS_MAX_ZRTP_CRYPTO_TYPES); i++) {
+ switch (srtpSuites[i]) {
+ case MS_AES_128_SHA1_32:
+ params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES1;
+ params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS32;
+ break;
+ case MS_AES_128_NO_AUTH:
+ params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES1;
+ break;
+ case MS_NO_CIPHER_SHA1_80:
+ params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS80;
+ break;
+ case MS_AES_128_SHA1_80:
+ params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES1;
+ params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS80;
+ break;
+ case MS_AES_CM_256_SHA1_80:
+ lWarning() << "Deprecated crypto suite MS_AES_CM_256_SHA1_80, use MS_AES_256_SHA1_80 instead";
+ BCTBX_NO_BREAK;
+ case MS_AES_256_SHA1_80:
+ params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES3;
+ params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS80;
+ break;
+ case MS_AES_256_SHA1_32:
+ params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES3;
+ params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS32;
+ break;
+ case MS_CRYPTO_SUITE_INVALID:
+ break;
+ }
+ }
+ }
+
+ /* linphone_core_get_srtp_crypto_suites is used to determine sensible defaults; here each can be overridden */
+ MsZrtpCryptoTypesCount ciphersCount = linphone_core_get_zrtp_cipher_suites(getCCore(), params->ciphers); /* if not present in config file, params->ciphers is not modified */
+ if (ciphersCount != 0) /* Use zrtp_cipher_suites config only when present, keep config from srtp_crypto_suite otherwise */
+ params->ciphersCount = ciphersCount;
+ params->hashesCount = linphone_core_get_zrtp_hash_suites(getCCore(), params->hashes);
+ MsZrtpCryptoTypesCount authTagsCount = linphone_core_get_zrtp_auth_suites(getCCore(), params->authTags); /* If not present in config file, params->authTags is not modified */
+ if (authTagsCount != 0)
+ params->authTagsCount = authTagsCount; /* Use zrtp_auth_suites config only when present, keep config from srtp_crypto_suite otherwise */
+ params->sasTypesCount = linphone_core_get_zrtp_sas_suites(getCCore(), params->sasTypes);
+ params->keyAgreementsCount = linphone_core_get_zrtp_key_agreement_suites(getCCore(), params->keyAgreements);
+
+ params->autoStart = (getMediaSessionPrivate().getParams()->getMediaEncryption() != LinphoneMediaEncryptionZRTP) && (haveRemoteZrtpHash == false) ;
+}
+
+void MS2AudioStream::configureAudioStream(){
+ MSSndCard *playcard = getCCore()->sound_conf.lsd_card ? getCCore()->sound_conf.lsd_card : getCCore()->sound_conf.play_sndcard;
+ if (playcard) {
+ // Set the stream type immediately, as on iOS AudioUnit is instanciated very early because it is
+ // otherwise too slow to start.
+ ms_snd_card_set_stream_type(playcard, MS_SND_CARD_STREAM_VOICE);
+ }
+
+ if (linphone_core_echo_limiter_enabled(getCCore())) {
+ string type = lp_config_get_string(linphone_core_get_config(getCCore()), "sound", "el_type", "mic");
+ if (type == "mic")
+ audio_stream_enable_echo_limiter(mStream, ELControlMic);
+ else if (type == "full")
+ audio_stream_enable_echo_limiter(mStream, ELControlFull);
+ }
+
+ // Equalizer location in the graph: 'mic' = in input graph, otherwise in output graph.
+ // Any other value than mic will default to output graph for compatibility.
+ string location = lp_config_get_string(linphone_core_get_config(getCCore()), "sound", "eq_location", "hp");
+ mStream->eq_loc = (location == "mic") ? MSEqualizerMic : MSEqualizerHP;
+ lInfo() << "Equalizer location: " << location;
+
+ audio_stream_enable_gain_control(mStream, true);
+ if (linphone_core_echo_cancellation_enabled(getCCore())) {
+ int len = lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "ec_tail_len", 0);
+ int delay = lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "ec_delay", 0);
+ int framesize = lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "ec_framesize", 0);
+ audio_stream_set_echo_canceller_params(mStream, len, delay, framesize);
+ if (mStream->ec) {
+ char *statestr=static_cast<char *>(ms_malloc0(ecStateMaxLen));
+ if (lp_config_relative_file_exists(linphone_core_get_config(getCCore()), ecStateStore)
+ && (lp_config_read_relative_file(linphone_core_get_config(getCCore()), ecStateStore, statestr, ecStateMaxLen) == 0)) {
+ ms_filter_call_method(mStream->ec, MS_ECHO_CANCELLER_SET_STATE_STRING, statestr);
+ }
+ ms_free(statestr);
+ }
+ }
+ audio_stream_enable_automatic_gain_control(mStream, linphone_core_agc_enabled(getCCore()));
+ bool_t enabled = !!lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "noisegate", 0);
+ audio_stream_enable_noise_gate(mStream, enabled);
+ audio_stream_set_features(mStream, linphone_core_get_audio_features(getCCore()));
+}
+
+bool MS2AudioStream::prepare(){
+ MSSndCard *playcard = getCCore()->sound_conf.lsd_card ? getCCore()->sound_conf.lsd_card : getCCore()->sound_conf.play_sndcard;
+ if (playcard) {
+ // Set the stream type immediately, as on iOS AudioUnit is instanciated very early because it is
+ // otherwise too slow to start.
+ ms_snd_card_set_stream_type(playcard, MS_SND_CARD_STREAM_VOICE);
+ }
+
+ if (!getCCore()->use_files){
+ audio_stream_prepare_sound(mStream, getCCore()->sound_conf.play_sndcard, getCCore()->sound_conf.capt_sndcard);
+ }else if (getIceService().isActive()){
+ audio_stream_prepare_sound(mStream, nullptr, nullptr);
+ }
+ MS2Stream::prepare();
+ return false;
+}
+
+void MS2AudioStream::sessionConfirmed(const OfferAnswerContext &ctx){
+ if (mStartZrtpLater){
+ lInfo() << "Starting zrtp late";
+ startZrtpPrimaryChannel(ctx);
+ mStartZrtpLater = false;
+ }
+}
+
+void MS2AudioStream::finishPrepare(){
+ MS2Stream::finishPrepare();
+ audio_stream_unprepare_sound(mStream);
+}
+
+MediaStream *MS2AudioStream::getMediaStream()const{
+ return &mStream->ms;
+}
+
+void MS2AudioStream::setupMediaLossCheck(){
+ int disconnectTimeout = linphone_core_get_nortp_timeout(getCCore());
+ mMediaLostCheckTimer = getCore().createTimer( [this, disconnectTimeout]() -> bool{
+ if (!audio_stream_alive(mStream, disconnectTimeout)){
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+ listener->onLossOfMediaDetected(getMediaSession().getSharedFromThis());
+ }
+ return true;
+ }, 1000, "Audio stream alive check");
+}
+
+void MS2AudioStream::render(const OfferAnswerContext ¶ms, CallSession::State targetState){
+ const SalStreamDescription *stream = params.resultStreamDescription;
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+
+ bool basicChangesHandled = handleBasicChanges(params, targetState);
+
+ if (basicChangesHandled) {
+ if (getState() == Running) {
+ bool muted = mMuted;
+ MS2Stream::render(params, targetState); // MS2Stream::render() may decide to unmute.
+ if (muted && !mMuted) {
+ lInfo() << "Early media finished, unmuting audio input...";
+ enableMic(micEnabled());
+ }
+ }
+ return;
+ }
+
+ int usedPt = -1;
+ string onHoldFile = "";
+ RtpProfile *audioProfile = makeProfile(params.resultMediaDescription, stream, &usedPt);
+ if (usedPt == -1){
+ lError() << "No payload types configured for this stream !";
+ stop();
+ return;
+ }
+
+ bool ok = true;
+ if (isMain()){
+ getMediaSessionPrivate().getCurrentParams()->getPrivate()->setUsedAudioCodec(rtp_profile_get_payload(audioProfile, usedPt));
+ }
+ MSSndCard *playcard = getCCore()->sound_conf.lsd_card ? getCCore()->sound_conf.lsd_card : getCCore()->sound_conf.play_sndcard;
+ if (!playcard)
+ lWarning() << "No card defined for playback!";
+ MSSndCard *captcard = getCCore()->sound_conf.capt_sndcard;
+ if (!captcard)
+ lWarning() << "No card defined for capture!";
+ string playfile = L_C_TO_STRING(getCCore()->play_file);
+ string recfile = L_C_TO_STRING(getCCore()->rec_file);
+ /* Don't use file or soundcard capture when placed in recv-only mode */
+ if ((stream->rtp_port == 0) || (stream->dir == SalStreamRecvOnly) || (stream->multicast_role == SalMulticastReceiver)) {
+ captcard = nullptr;
+ playfile = "";
+ }
+
+ if (targetState == CallSession::State::Paused) {
+ // In paused state, we never use soundcard
+ playcard = captcard = nullptr;
+ recfile = "";
+ // And we will eventually play "playfile" if set by the user
+ }
+ if (listener && listener->isPlayingRingbackTone(getMediaSession().getSharedFromThis())) {
+ captcard = nullptr;
+ playfile = ""; /* It is setup later */
+ if (lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "send_ringback_without_playback", 0) == 1) {
+ playcard = nullptr;
+ recfile = "";
+ }
+ }
+ // If playfile are supplied don't use soundcards
+ bool useRtpIo = !!lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "rtp_io", false);
+ bool useRtpIoEnableLocalOutput = !!lp_config_get_int(linphone_core_get_config(getCCore()), "sound", "rtp_io_enable_local_output", false);
+ if (getCCore()->use_files || (useRtpIo && !useRtpIoEnableLocalOutput)) {
+ captcard = playcard = nullptr;
+ }
+ if (getMediaSessionPrivate().getParams()->getPrivate()->getInConference()) {
+ // First create the graph without soundcard resources
+ captcard = playcard = nullptr;
+ }
+ if (listener && !listener->areSoundResourcesAvailable(getMediaSession().getSharedFromThis())) {
+ lInfo() << "Sound resources are used by another CallSession, not using soundcard";
+ captcard = playcard = nullptr;
+ }
+
+ if (playcard) {
+ ms_snd_card_set_stream_type(playcard, MS_SND_CARD_STREAM_VOICE);
+ }
+ configureAudioStream();
+ bool useEc = captcard && linphone_core_echo_cancellation_enabled(getCCore());
+ audio_stream_enable_echo_canceller(mStream, useEc);
+ if (playcard && (stream->max_rate > 0))
+ ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
+ if (captcard && (stream->max_rate > 0))
+ ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
+
+ if (!getMediaSessionPrivate().getParams()->getPrivate()->getInConference() && !getMediaSessionPrivate().getParams()->getRecordFilePath().empty()) {
+ audio_stream_mixed_record_open(mStream, getMediaSessionPrivate().getParams()->getRecordFilePath().c_str());
+ getMediaSessionPrivate().getCurrentParams()->setRecordFilePath(getMediaSessionPrivate().getParams()->getRecordFilePath());
+ }
+
+ MS2Stream::render(params, targetState);
+ RtpAddressInfo dest;
+ getRtpDestination(params, &dest);
+ /* Now start the stream */
+ MSMediaStreamIO io = MS_MEDIA_STREAM_IO_INITIALIZER;
+ if (useRtpIo) {
+ if (useRtpIoEnableLocalOutput) {
+ io.input.type = MSResourceRtp;
+ io.input.session = createRtpIoSession();
+ if (playcard) {
+ io.output.type = MSResourceSoundcard;
+ io.output.soundcard = playcard;
+ } else {
+ io.output.type = MSResourceFile;
+ io.output.file = recfile.empty() ? nullptr : recfile.c_str();
+ }
+ } else {
+ io.input.type = io.output.type = MSResourceRtp;
+ io.input.session = io.output.session = createRtpIoSession();
+ }
+ if (!io.input.session)
+ ok = false;
+ } else {
+ if (playcard) {
+ io.output.type = MSResourceSoundcard;
+ io.output.soundcard = playcard;
+ } else {
+ io.output.type = MSResourceFile;
+ io.output.file = recfile.empty() ? nullptr : recfile.c_str();
+ }
+ if (captcard) {
+ io.input.type = MSResourceSoundcard;
+ io.input.soundcard = captcard;
+ } else {
+ io.input.type = MSResourceFile;
+ onHoldFile = playfile;
+ io.input.file = nullptr; /* We prefer to use the remote_play api, that allows to play multimedia files */
+ }
+ }
+ if (ok) {
+ VideoStream *vs = getPeerVideoStream();
+ if (vs) audio_stream_link_video(mStream, vs);
+ mCurrentCaptureCard = ms_media_resource_get_soundcard(&io.input);
+ mCurrentPlaybackCard = ms_media_resource_get_soundcard(&io.output);
+
+ int err = audio_stream_start_from_io(mStream, audioProfile, dest.rtpAddr.c_str(), dest.rtpPort,
+ dest.rtcpAddr.c_str(), dest.rtcpPort, usedPt, &io);
+ if (err == 0)
+ postConfigureAudioStream((mMuted || mMicMuted) && (listener && !listener->isPlayingRingbackTone(getMediaSession().getSharedFromThis())));
+ mStartCount++;
+ }
+
+ if ((targetState == CallSession::State::Paused) && !captcard && !playfile.empty()) {
+ int pauseTime = 500;
+ ms_filter_call_method(mStream->soundread, MS_FILE_PLAYER_LOOP, &pauseTime);
+ }
+ if (listener && listener->isPlayingRingbackTone(getMediaSession().getSharedFromThis()))
+ setupRingbackPlayer();
+ if (getMediaSessionPrivate().getParams()->getPrivate()->getInConference() && listener) {
+ // Transform the graph to connect it to the conference filter
+ bool mute = (stream->dir == SalStreamRecvOnly);
+ listener->onCallSessionConferenceStreamStarting(getMediaSession().getSharedFromThis(), mute);
+ }
+ getMediaSessionPrivate().getCurrentParams()->getPrivate()->setInConference(getMediaSessionPrivate().getParams()->getPrivate()->getInConference());
+ getMediaSessionPrivate().getCurrentParams()->enableLowBandwidth(getMediaSessionPrivate().getParams()->lowBandwidthEnabled());
+
+ // Start ZRTP engine if needed : set here or remote have a zrtp-hash attribute
+ if (linphone_core_media_encryption_supported(getCCore(), LinphoneMediaEncryptionZRTP) && isMain()) {
+ getMediaSessionPrivate().performMutualAuthentication();
+ LinphoneMediaEncryption requestedMediaEncryption = getMediaSessionPrivate().getParams()->getMediaEncryption();
+ //Start zrtp if remote has offered it or if local is configured for zrtp and is the offerrer. If not, defered when ACK is received
+ if ((requestedMediaEncryption == LinphoneMediaEncryptionZRTP && params.localIsOfferer)
+ || (params.remoteStreamDescription->haveZrtpHash == 1)) {
+ startZrtpPrimaryChannel(params);
+ }else if (requestedMediaEncryption == LinphoneMediaEncryptionZRTP && !params.localIsOfferer){
+ mStartZrtpLater = true;
+ }
+ }
+
+ getGroup().addPostRenderHook([this, onHoldFile] {
+ /* The on-hold file is to be played once both audio and video are ready */
+ if (!onHoldFile.empty() && !getMediaSessionPrivate().getParams()->getPrivate()->getInConference()) {
+ MSFilter *player = audio_stream_open_remote_play(mStream, onHoldFile.c_str());
+ if (player) {
+ int pauseTime = 500;
+ ms_filter_call_method(player, MS_PLAYER_SET_LOOP, &pauseTime);
+ ms_filter_call_method_noarg(player, MS_PLAYER_START);
+ }
+ }
+ });
+
+ if (targetState == CallSession::State::StreamsRunning){
+ setupMediaLossCheck();
+ }
+
+ return;
+}
+
+void MS2AudioStream::stop(){
+ if (mMediaLostCheckTimer) {
+ getCore().destroyTimer(mMediaLostCheckTimer);
+ mMediaLostCheckTimer = nullptr;
+ }
+ MS2Stream::stop();
+ if (mStream->ec) {
+ char *stateStr = nullptr;
+ ms_filter_call_method(mStream->ec, MS_ECHO_CANCELLER_GET_STATE_STRING, &stateStr);
+ if (stateStr) {
+ lInfo() << "Writing echo canceler state, " << (int)strlen(stateStr) << " bytes";
+ lp_config_write_relative_file(linphone_core_get_config(getCCore()), ecStateStore, stateStr);
+ }
+ }
+ VideoStream *vs = getPeerVideoStream();
+ if (vs) audio_stream_unlink_video(mStream, vs);
+
+ audio_stream_stop(mStream);
+ /* In mediastreamer2, stop actually stops and destroys. We immediately need to recreate the stream object for later use, keeping the
+ * sessions (for RTP, SRTP, ZRTP etc) that were setup at the beginning. */
+ mStream = audio_stream_new_with_sessions(getCCore()->factory, &mSessions);
+ getMediaSessionPrivate().getCurrentParams()->getPrivate()->setUsedAudioCodec(nullptr);
+
+
+ mCurrentCaptureCard = nullptr;
+ mCurrentPlaybackCard = nullptr;
+}
+
+//To give a chance for auxilary secret to be used, primary channel (I.E audio) should be started either on 200ok if ZRTP is signaled by a zrtp-hash or when ACK is received in case calling side does not have zrtp-hash.
+void MS2AudioStream::startZrtpPrimaryChannel(const OfferAnswerContext ¶ms) {
+ const SalStreamDescription *remote = params.remoteStreamDescription;
+ audio_stream_start_zrtp(mStream);
+ if (remote->haveZrtpHash == 1) {
+ int retval = ms_zrtp_setPeerHelloHash(mSessions.zrtp_context, (uint8_t *)remote->zrtphash, strlen((const char *)(remote->zrtphash)));
+ if (retval != 0)
+ lError() << "ZRTP hash mismatch 0x" << hex << retval;
+ }
+}
+
+void MS2AudioStream::forceSpeakerMuted (bool muted) {
+ if (muted)
+ audio_stream_set_spk_gain(mStream, 0);
+ else
+ audio_stream_set_spk_gain_db(mStream, getCCore()->sound_conf.soft_play_lev);
+}
+
+void MS2AudioStream::setRoute(LinphoneAudioRoute route){
+ audio_stream_set_audio_route(mStream, (MSAudioRoute)route);
+}
+
+void MS2AudioStream::parameterizeEqualizer(AudioStream *as, LinphoneCore *lc) {
+ LinphoneConfig *config = linphone_core_get_config(lc);
+ const char *eqActive = lp_config_get_string(config, "sound", "eq_active", nullptr);
+ if (eqActive)
+ lWarning() << "'eq_active' linphonerc parameter has no effect anymore. Please use 'mic_eq_active' or 'spk_eq_active' instead";
+ const char *eqGains = lp_config_get_string(config, "sound", "eq_gains", nullptr);
+ if(eqGains)
+ lWarning() << "'eq_gains' linphonerc parameter has no effect anymore. Please use 'mic_eq_gains' or 'spk_eq_gains' instead";
+ if (as->mic_equalizer) {
+ MSFilter *f = as->mic_equalizer;
+ bool enabled = !!lp_config_get_int(config, "sound", "mic_eq_active", 0);
+ ms_filter_call_method(f, MS_EQUALIZER_SET_ACTIVE, &enabled);
+ const char *gains = lp_config_get_string(config, "sound", "mic_eq_gains", nullptr);
+ if (enabled && gains) {
+ bctbx_list_t *gainsList = ms_parse_equalizer_string(gains);
+ for (bctbx_list_t *it = gainsList; it; it = bctbx_list_next(it)) {
+ MSEqualizerGain *g = reinterpret_cast<MSEqualizerGain *>(bctbx_list_get_data(it));
+ lInfo() << "Read microphone equalizer gains: " << g->frequency << "(~" << g->width << ") --> " << g->gain;
+ ms_filter_call_method(f, MS_EQUALIZER_SET_GAIN, g);
+ }
+ if (gainsList)
+ bctbx_list_free_with_data(gainsList, ms_free);
+ }
+ }
+ if (as->spk_equalizer) {
+ MSFilter *f = as->spk_equalizer;
+ bool enabled = !!lp_config_get_int(config, "sound", "spk_eq_active", 0);
+ ms_filter_call_method(f, MS_EQUALIZER_SET_ACTIVE, &enabled);
+ const char *gains = lp_config_get_string(config, "sound", "spk_eq_gains", nullptr);
+ if (enabled && gains) {
+ bctbx_list_t *gainsList = ms_parse_equalizer_string(gains);
+ for (bctbx_list_t *it = gainsList; it; it = bctbx_list_next(it)) {
+ MSEqualizerGain *g = reinterpret_cast<MSEqualizerGain *>(bctbx_list_get_data(it));
+ lInfo() << "Read speaker equalizer gains: " << g->frequency << "(~" << g->width << ") --> " << g->gain;
+ ms_filter_call_method(f, MS_EQUALIZER_SET_GAIN, g);
+ }
+ if (gainsList)
+ bctbx_list_free_with_data(gainsList, ms_free);
+ }
+ }
+}
+
+void MS2AudioStream::postConfigureAudioStream(AudioStream *as, LinphoneCore *lc, bool muted){
+ float micGain = lc->sound_conf.soft_mic_lev;
+ if (muted)
+ audio_stream_set_mic_gain(as, 0);
+ else
+ audio_stream_set_mic_gain_db(as, micGain);
+ float recvGain = lc->sound_conf.soft_play_lev;
+ if (static_cast<int>(recvGain)){
+ if (as->volrecv)
+ ms_filter_call_method(as->volrecv, MS_VOLUME_SET_DB_GAIN, &recvGain);
+ else
+ lWarning() << "Could not apply playback gain: gain control wasn't activated";
+ }
+ LinphoneConfig *config = linphone_core_get_config(lc);
+ float ngThres = lp_config_get_float(config, "sound", "ng_thres", 0.05f);
+ float ngFloorGain = lp_config_get_float(config, "sound", "ng_floorgain", 0);
+ if (as->volsend) {
+ int dcRemoval = lp_config_get_int(config, "sound", "dc_removal", 0);
+ ms_filter_call_method(as->volsend, MS_VOLUME_REMOVE_DC, &dcRemoval);
+ float speed = lp_config_get_float(config, "sound", "el_speed", -1);
+ float thres = lp_config_get_float(config, "sound", "el_thres", -1);
+ float force = lp_config_get_float(config, "sound", "el_force", -1);
+ int sustain = lp_config_get_int(config, "sound", "el_sustain", -1);
+ float transmitThres = lp_config_get_float(config, "sound", "el_transmit_thres", -1);
+ if (static_cast<int>(speed) == -1)
+ speed = 0.03f;
+ if (static_cast<int>(force) == -1)
+ force = 25;
+ MSFilter *f = as->volsend;
+ ms_filter_call_method(f, MS_VOLUME_SET_EA_SPEED, &speed);
+ ms_filter_call_method(f, MS_VOLUME_SET_EA_FORCE, &force);
+ if (static_cast<int>(thres) != -1)
+ ms_filter_call_method(f, MS_VOLUME_SET_EA_THRESHOLD, &thres);
+ if (static_cast<int>(sustain) != -1)
+ ms_filter_call_method(f, MS_VOLUME_SET_EA_SUSTAIN, &sustain);
+ if (static_cast<int>(transmitThres) != -1)
+ ms_filter_call_method(f, MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD, &transmitThres);
+ ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_THRESHOLD, &ngThres);
+ ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_FLOORGAIN, &ngFloorGain);
+ }
+ if (as->volrecv) {
+ /* Parameters for a limited noise-gate effect, using echo limiter threshold */
+ float floorGain = (float)(1 / pow(10, micGain / 10));
+ int spkAgc = lp_config_get_int(config, "sound", "speaker_agc_enabled", 0);
+ MSFilter *f = as->volrecv;
+ ms_filter_call_method(f, MS_VOLUME_ENABLE_AGC, &spkAgc);
+ ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_THRESHOLD, &ngThres);
+ ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_FLOORGAIN, &floorGain);
+ }
+ parameterizeEqualizer(as, lc);
+}
+
+void MS2AudioStream::postConfigureAudioStream(bool muted) {
+ postConfigureAudioStream(mStream, getCCore(), muted);
+ forceSpeakerMuted(mSpeakerMuted);
+ if (linphone_core_dtmf_received_has_listener(getCCore()))
+ audio_stream_play_received_dtmfs(mStream, false);
+ if (mRecordActive)
+ startRecording();
+}
+
+void MS2AudioStream::setupRingbackPlayer () {
+ int pauseTime = 3000;
+ audio_stream_play(mStream, getCCore()->sound_conf.ringback_tone);
+ ms_filter_call_method(mStream->soundread, MS_FILE_PLAYER_LOOP, &pauseTime);
+}
+
+void MS2AudioStream::telephoneEventReceived (int event) {
+ static char dtmfTab[16] = { '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '*', '#', 'A', 'B', 'C', 'D' };
+ if ((event < 0) || (event > 15)) {
+ lWarning() << "Bad dtmf value " << event;
+ return;
+ }
+ getMediaSessionPrivate().dtmfReceived(dtmfTab[event]);
+}
+
+void MS2AudioStream::handleEvent(const OrtpEvent *ev){
+ OrtpEventType evt = ortp_event_get_type(ev);
+ OrtpEventData *evd = ortp_event_get_data(const_cast<OrtpEvent*>(ev));
+ switch (evt){
+ case ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED:
+ if (isMain()) getGroup().zrtpStarted(this);
+ break;
+ case ORTP_EVENT_ZRTP_SAS_READY:
+ getGroup().authTokenReady(evd->info.zrtp_info.sas, !!evd->info.zrtp_info.verified);
+ break;
+ case ORTP_EVENT_TELEPHONE_EVENT:
+ telephoneEventReceived(evd->info.telephone_event);
+ break;
+ }
+}
+
+void MS2AudioStream::enableMic(bool value){
+ mMicMuted = !value;
+
+ if (mMicMuted)
+ audio_stream_set_mic_gain(mStream, 0);
+ else
+ audio_stream_set_mic_gain_db(mStream, getCCore()->sound_conf.soft_mic_lev);
+}
+
+bool MS2AudioStream::micEnabled()const{
+ return !mMicMuted;
+}
+
+void MS2AudioStream::enableSpeaker(bool value){
+ mSpeakerMuted = !value;
+ forceSpeakerMuted(mSpeakerMuted);
+}
+
+bool MS2AudioStream::speakerEnabled()const{
+ return !mSpeakerMuted;
+}
+
+void MS2AudioStream::sendDtmf(int dtmf){
+ audio_stream_send_dtmf(mStream, (char)dtmf);
+}
+
+void MS2AudioStream::startRecording(){
+ if (getMediaSessionPrivate().getParams()->getRecordFilePath().empty()) {
+ lError() << "MS2AudioStream::startRecording(): no output file specified. Use MediaSessionParams::setRecordFilePath()";
+ return;
+ }
+ if (getMediaSessionPrivate().getParams()->getPrivate()->getInConference()){
+ lWarning() << "MS2AudioStream::startRecording(): not supported in conference.";
+ return;
+ }
+ if (media_stream_get_state(&mStream->ms) == MSStreamStarted) audio_stream_mixed_record_start(mStream);
+ mRecordActive = true;
+}
+
+void MS2AudioStream::stopRecording(){
+ if (mRecordActive)
+ audio_stream_mixed_record_stop(mStream);
+ mRecordActive = false;
+}
+
+float MS2AudioStream::getPlayVolume(){
+ if (mStream->volrecv) {
+ float vol = 0;
+ ms_filter_call_method(mStream->volrecv, MS_VOLUME_GET, &vol);
+ return vol;
+ }
+ return LINPHONE_VOLUME_DB_LOWEST;
+}
+
+float MS2AudioStream::getRecordVolume(){
+ if (mStream->volsend && !mMicMuted) {
+ float vol = 0;
+ ms_filter_call_method(mStream->volsend, MS_VOLUME_GET, &vol);
+ return vol;
+ }
+ return LINPHONE_VOLUME_DB_LOWEST;
+}
+
+float MS2AudioStream::getMicGain(){
+ return audio_stream_get_sound_card_input_gain(mStream);
+}
+
+void MS2AudioStream::setMicGain(float gain){
+ audio_stream_set_sound_card_input_gain(mStream, gain);
+}
+
+float MS2AudioStream::getSpeakerGain(){
+ return audio_stream_get_sound_card_output_gain(mStream);
+}
+
+void MS2AudioStream::setSpeakerGain(float gain){
+ audio_stream_set_sound_card_output_gain(mStream, gain);
+}
+
+VideoStream *MS2AudioStream::getPeerVideoStream(){
+#ifdef VIDEO_ENABLED
+ MS2VideoStream *vs = getGroup().lookupMainStreamInterface<MS2VideoStream>(SalVideo);
+ return vs ? (VideoStream*)vs->getMediaStream() : nullptr;
+#else
+ return nullptr;
+#endif
+}
+
+void MS2AudioStream::enableEchoCancellation(bool value){
+ if (mStream->ec) {
+ bool bypassMode = !value;
+ ms_filter_call_method(mStream->ec, MS_ECHO_CANCELLER_SET_BYPASS_MODE, &bypassMode);
+ }
+
+}
+
+bool MS2AudioStream::echoCancellationEnabled()const{
+ if (!mStream->ec)
+ return !!linphone_core_echo_cancellation_enabled(getCCore());
+
+ bool_t val;
+ ms_filter_call_method(mStream->ec, MS_ECHO_CANCELLER_GET_BYPASS_MODE, &val);
+ return !val;
+}
+
+void MS2AudioStream::finish(){
+ if (mStream){
+ audio_stream_stop(mStream);
+ mStream = nullptr;
+ }
+ MS2Stream::finish();
+}
+
+MS2AudioStream::~MS2AudioStream(){
+ if (mStream)
+ audio_stream_stop(mStream);
+}
+
+
+LINPHONE_END_NAMESPACE
diff --git a/src/conference/session/call-session-listener.h b/src/conference/session/call-session-listener.h
index a1233e800a067ac601ec1110bf9283a78594a703..947d14c934e097a05dab5a74b88f2ce645b639c1 100644
--- a/src/conference/session/call-session-listener.h
+++ b/src/conference/session/call-session-listener.h
@@ -53,7 +53,7 @@ public:
virtual void onIncomingCallSessionStarted (const std::shared_ptr<CallSession> &session) {}
virtual void onIncomingCallSessionTimeoutCheck (const std::shared_ptr<CallSession> &session, int elapsed, bool oneSecondElapsed) {}
virtual void onInfoReceived (const std::shared_ptr<CallSession> &session, const LinphoneInfoMessage *im) {}
- virtual void onNoMediaTimeoutCheck (const std::shared_ptr<CallSession> &session, bool oneSecondElapsed) {}
+ virtual void onLossOfMediaDetected (const std::shared_ptr<CallSession> &session) {}
virtual void onTmmbrReceived (const std::shared_ptr<CallSession> &session, int streamIndex, int tmmbr) {}
virtual void onSnapshotTaken(const std::shared_ptr<CallSession> &session, const char *file_path) {}
diff --git a/src/conference/session/call-session-p.h b/src/conference/session/call-session-p.h
index 4e7b1704bb5c7ce2c1544ebda7cdbb0b424d6a26..a052c77bcf3e46f36ba90f7b5a9731f7365a2baf 100644
--- a/src/conference/session/call-session-p.h
+++ b/src/conference/session/call-session-p.h
@@ -67,6 +67,9 @@ public:
virtual void updating (bool isUpdate);
void setCallSessionListener (CallSessionListener *listener) { this->listener = listener; }
+ CallSessionListener *getCallSessionListener()const{
+ return listener;
+ }
protected:
void init ();
diff --git a/src/conference/session/call-session.cpp b/src/conference/session/call-session.cpp
index e68a04f26cc4dad33f3766101a9ab0c932c69239..b952e6bc36212ac1868625df1bcc4949e57fa098 100644
--- a/src/conference/session/call-session.cpp
+++ b/src/conference/session/call-session.cpp
@@ -989,7 +989,7 @@ void CallSession::configure (LinphoneCallDir direction, LinphoneProxyConfig *cfg
d->startPing();
} else if (direction == LinphoneCallIncoming) {
d->setParams(new CallSessionParams());
- d->params->initDefault(getCore());
+ d->params->initDefault(getCore(), LinphoneCallIncoming);
}
}
diff --git a/src/conference/session/media-description-renderer.cpp b/src/conference/session/media-description-renderer.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..ceb6ff77ebbc954d9e4600d3cafe35c1d044af56
--- /dev/null
+++ b/src/conference/session/media-description-renderer.cpp
@@ -0,0 +1,87 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+
+#include "media-description-renderer.h"
+
+LINPHONE_BEGIN_NAMESPACE
+
+void OfferAnswerContext::scopeStreamToIndex(size_t index) const{
+ streamIndex = index;
+ localStreamDescription = localMediaDescription ? &localMediaDescription->streams[index] : nullptr;
+ remoteStreamDescription = remoteMediaDescription ? &remoteMediaDescription->streams[index] : nullptr;
+ resultStreamDescription = resultMediaDescription ? &resultMediaDescription->streams[index] : nullptr;
+}
+
+void OfferAnswerContext::dupFrom(const OfferAnswerContext &ctx){
+ OfferAnswerContext oldCtx = *this; // Transfers *this to a temporary object.
+ localMediaDescription = ctx.localMediaDescription ? sal_media_description_ref(ctx.localMediaDescription) : nullptr;
+ remoteMediaDescription = ctx.remoteMediaDescription ? sal_media_description_ref(const_cast<SalMediaDescription*>(ctx.remoteMediaDescription)) : nullptr;
+ resultMediaDescription = ctx.resultMediaDescription ? sal_media_description_ref(const_cast<SalMediaDescription*>(ctx.resultMediaDescription)) : nullptr;
+ localIsOfferer = ctx.localIsOfferer;
+ mOwnsMediaDescriptions = true;
+ // if the temporary oldCtx owns media descriptions, they will be unrefed by the destructor here.
+}
+
+void OfferAnswerContext::copyFrom(const OfferAnswerContext &ctx){
+ OfferAnswerContext oldCtx = *this; // Transfers *this to a temporary object.
+ localMediaDescription = ctx.localMediaDescription;
+ remoteMediaDescription = ctx.remoteMediaDescription;
+ resultMediaDescription = ctx.resultMediaDescription;
+ localIsOfferer = ctx.localIsOfferer;
+ // if the temporary oldCtx owns media descriptions, they will be unrefed by the destructor here.
+}
+
+void OfferAnswerContext::scopeStreamToIndexWithDiff(size_t index, const OfferAnswerContext &previousCtx) const{
+ scopeStreamToIndex(index);
+ previousCtx.scopeStreamToIndex(index);
+
+ if (previousCtx.localMediaDescription){
+ localStreamDescriptionChanges = sal_media_description_global_equals(previousCtx.localMediaDescription, localMediaDescription)
+ | sal_stream_description_equals(previousCtx.localStreamDescription, localStreamDescription);
+ }else localStreamDescriptionChanges = 0;
+ if (previousCtx.resultMediaDescription && resultMediaDescription){
+ resultStreamDescriptionChanges = sal_media_description_global_equals(previousCtx.resultMediaDescription, resultMediaDescription)
+ | sal_stream_description_equals(previousCtx.resultStreamDescription, resultStreamDescription);
+ }else resultStreamDescriptionChanges = 0;
+}
+
+void OfferAnswerContext::clear(){
+ if (mOwnsMediaDescriptions){
+ if (localMediaDescription) sal_media_description_unref(localMediaDescription);
+ if (remoteMediaDescription) sal_media_description_unref(const_cast<SalMediaDescription*>(remoteMediaDescription));
+ if (resultMediaDescription) sal_media_description_unref(const_cast<SalMediaDescription*>(resultMediaDescription));
+ }
+ localMediaDescription = nullptr;
+ remoteMediaDescription = nullptr;
+ resultMediaDescription = nullptr;
+ localStreamDescription = nullptr;
+ remoteStreamDescription = nullptr;
+ resultStreamDescription = nullptr;
+ localStreamDescriptionChanges = 0;
+ resultStreamDescriptionChanges = 0;
+ mOwnsMediaDescriptions = false;
+}
+
+OfferAnswerContext::~OfferAnswerContext(){
+ clear();
+}
+
+LINPHONE_END_NAMESPACE
+
diff --git a/src/conference/session/media-description-renderer.h b/src/conference/session/media-description-renderer.h
new file mode 100644
index 0000000000000000000000000000000000000000..178d2187c36fd9e045a4be326dddd2a6a95313d7
--- /dev/null
+++ b/src/conference/session/media-description-renderer.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef media_description_renderer_h
+#define media_description_renderer_h
+
+#include "call-session.h"
+#include "c-wrapper/internal/c-sal.h"
+
+LINPHONE_BEGIN_NAMESPACE
+
+
+/**
+ * Represents all offer/answer context.
+ * When passed to a Stream object scopeStreamToIndex() must be called to specify the considered stream index, which
+ * initialize the localStreamDescription, remoteStreamDescription, and resultStreamDescription.
+ */
+class OfferAnswerContext{
+public:
+ OfferAnswerContext() = default;
+ SalMediaDescription *localMediaDescription = nullptr;
+ const SalMediaDescription *remoteMediaDescription = nullptr;
+ const SalMediaDescription *resultMediaDescription = nullptr;
+ bool localIsOfferer = false;
+
+ mutable int localStreamDescriptionChanges = 0;
+ mutable int resultStreamDescriptionChanges = 0;
+ mutable SalStreamDescription *localStreamDescription = nullptr;
+ mutable const SalStreamDescription *remoteStreamDescription = nullptr;
+ mutable const SalStreamDescription *resultStreamDescription = nullptr;
+ mutable size_t streamIndex = 0;
+
+ void scopeStreamToIndex(size_t index)const;
+ void scopeStreamToIndexWithDiff(size_t index, const OfferAnswerContext &previousCtx)const;
+ /* Copy descriptions from 'ctx', taking ownership of descriptions. */
+ void dupFrom(const OfferAnswerContext &ctx);
+ /* Copy descriptions from 'ctx', NOT taking ownership of descriptions. */
+ void copyFrom(const OfferAnswerContext &ctx);
+ void clear();
+ ~OfferAnswerContext();
+private:
+ OfferAnswerContext(const OfferAnswerContext &other) = default;
+ OfferAnswerContext & operator=(const OfferAnswerContext &other) = default;
+ bool mOwnsMediaDescriptions = false;
+};
+
+/*
+ * Interface for any kind of engine that is responsible to render the streams described by
+ * a SalMediaDescription within the context of an offer-answer.
+ */
+class MediaDescriptionRenderer{
+public:
+ /*
+ * Request the engine to fill additional information (that it usually controls) into the local media description.
+ */
+ virtual void fillLocalMediaDescription(OfferAnswerContext & ctx) = 0;
+ /*
+ * Prepare to run.
+ */
+ virtual bool prepare() = 0;
+ /*
+ * Prepare stage is finishing.
+ */
+ virtual void finishPrepare() = 0;
+ /*
+ * Render the streams according to offer answer context.
+ */
+ virtual void render(const OfferAnswerContext & ctx, CallSession::State targetState) = 0;
+ /*
+ * Called to notify that the session is confirmed (corresponding to SIP ACK).
+ */
+ virtual void sessionConfirmed(const OfferAnswerContext &ctx) = 0;
+ /*
+ * Stop rendering streams.
+ */
+ virtual void stop() = 0;
+ /*
+ * Release engine's resource, pending object destruction.
+ */
+ virtual void finish() = 0;
+ virtual ~MediaDescriptionRenderer() = default;
+};
+
+LINPHONE_END_NAMESPACE
+
+#endif
+
diff --git a/src/conference/session/media-session-p.h b/src/conference/session/media-session-p.h
index 752c793d2939787130c8a659c44ab783b97eb2cf..321e366deade9e36be2ff6e31abfc982e16711e3 100644
--- a/src/conference/session/media-session-p.h
+++ b/src/conference/session/media-session-p.h
@@ -24,10 +24,11 @@
#include <functional>
#include "call-session-p.h"
+#include "ms2-streams.h"
#include "media-session.h"
#include "port-config.h"
-#include "nat/ice-agent.h"
+#include "nat/ice-service.h"
#include "nat/stun-client.h"
#include "linphone/call_stats.h"
@@ -36,7 +37,9 @@
LINPHONE_BEGIN_NAMESPACE
-class MediaSessionPrivate : public CallSessionPrivate {
+
+class MediaSessionPrivate : public CallSessionPrivate, private IceServiceListener {
+ friend class StreamsGroup;
public:
static int resumeAfterFailedTransfer (void *userData, unsigned int);
static bool_t startPendingRefer (void *userData);
@@ -58,20 +61,13 @@ public:
void updated (bool isUpdate);
void updating (bool isUpdate) override;
- void enableSymmetricRtp (bool value);
- void oglRender () const;
+ void oglRender ();
void sendVfu ();
- void clearIceCheckList (IceCheckList *cl);
- void deactivateIce ();
- void prepareStreamsForIceGathering (bool hasVideo);
- void stopStreamsForIceGathering ();
-
int getAf () const { return af; }
bool getSpeakerMuted () const;
void setSpeakerMuted (bool muted);
- void forceSpeakerMuted (bool muted);
bool getMicrophoneMuted () const;
void setMicrophoneMuted (bool muted);
@@ -83,28 +79,19 @@ public:
void setParams (MediaSessionParams *msp);
void setRemoteParams (MediaSessionParams *msp);
- IceSession *getIceSession () const { return iceAgent ? iceAgent->getIceSession() : nullptr; }
-
+ IceService &getIceService() const { return streamsGroup->getIceService(); }
SalMediaDescription *getLocalDesc () const { return localDesc; }
unsigned int getAudioStartCount () const;
unsigned int getVideoStartCount () const;
unsigned int getTextStartCount () const;
- MediaStream *getMediaStream (LinphoneStreamType type) const;
LinphoneNatPolicy *getNatPolicy () const { return natPolicy; }
- int getRtcpPort (LinphoneStreamType type) const;
- int getRtpPort (LinphoneStreamType type) const;
LinphoneCallStats *getStats (LinphoneStreamType type) const;
- int getStreamIndex (LinphoneStreamType type) const;
- int getStreamIndex (MediaStream *ms) const;
+
SalCallOp * getOp () const { return op; }
- MSWebCam *getVideoDevice () const;
- void initializeStreams ();
void stopStreams ();
- void stopStream (SalStreamDescription *streamDesc);
- void restartStream (SalStreamDescription *streamDesc, int streamIndex, int sdChanged, CallSession::State targetState);
// Methods used by testers
void addLocalDescChangedFlag (int flag) { localDescChanged |= flag; }
@@ -120,54 +107,52 @@ public:
// Call listener
void snapshotTakenCb(void *userdata, struct _MSFilter *f, unsigned int id, void *arg);
-
+ StreamsGroup & getStreamsGroup()const {
+ return *streamsGroup.get();
+ }
+ std::shared_ptr<Participant> getMe () const;
+ void setDtlsFingerprint(const std::string &fingerPrint);
+ const std::string & getDtlsFingerprint()const;
+ bool isEncryptionMandatory () const;
+ MSWebCam *getVideoDevice()const;
+ void performMutualAuthentication();
+ const std::string &getMediaLocalIp()const{ return mediaLocalIp; }
+ void lossOfMediaDetected();
+ /* test function */
+ IceSession *getIceSession()const;
private:
- static OrtpJitterBufferAlgorithm jitterBufferNameToAlgo (const std::string &name);
-
-#ifdef VIDEO_ENABLED
- static void videoStreamEventCb (void *userData, const MSFilter *f, const unsigned int eventId, const void *args);
-#endif // ifdef VIDEO_ENABLED
+ /* IceServiceListener methods:*/
+ virtual void onGatheringFinished(IceService &service) override;
+ virtual void onIceCompleted(IceService &service) override;
+ virtual void onLosingPairsCompleted(IceService &service) override;
+ virtual void onIceRestartNeeded(IceService & service) override;
+
#ifdef TEST_EXT_RENDERER
static void extRendererCb (void *userData, const MSPicture *local, const MSPicture *remote);
#endif // ifdef TEST_EXT_RENDERER
- static void realTimeTextCharacterReceived (void *userData, MSFilter *f, unsigned int id, void *arg);
static int sendDtmf (void *data, unsigned int revents);
- static float aggregateQualityRatings (float audioRating, float videoRating);
- std::shared_ptr<Participant> getMe () const;
void setState (CallSession::State newState, const std::string &message) override;
- void computeStreamsIndexes (const SalMediaDescription *md);
- void fixCallParams (SalMediaDescription *rmd);
+ void assignStreamsIndexesIncoming(const SalMediaDescription *md);
+ void assignStreamsIndexes();
+ int getFirstStreamWithType(const SalMediaDescription *md, SalStreamType type);
+ void fixCallParams (SalMediaDescription *rmd, bool fromOffer);
void initializeParamsAccordingToIncomingCallParams () override;
void setCompatibleIncomingCallParams (SalMediaDescription *md);
void updateBiggestDesc (SalMediaDescription *md);
void updateRemoteSessionIdAndVer ();
- void initStats (LinphoneCallStats *stats, LinphoneStreamType type);
- void notifyStatsUpdated (int streamIndex);
-
- OrtpEvQueue *getEventQueue (int streamIndex) const;
- MediaStream *getMediaStream (int streamIndex) const;
-
- void fillMulticastMediaAddresses ();
- int selectFixedPort (int streamIndex, std::pair<int, int> portRange);
- int selectRandomPort (int streamIndex, std::pair<int, int> portRange);
- void setPortConfig (int streamIndex, std::pair<int, int> portRange);
- void setPortConfigFromRtpSession (int streamIndex, RtpSession *session);
- void setRandomPortConfig (int streamIndex);
void discoverMtu (const Address &remoteAddr);
- std::string getBindIpForStream (int streamIndex);
void getLocalIp (const Address &remoteAddr);
- std::string getPublicIpForStream (int streamIndex);
void runStunTestsIfNeeded ();
void selectIncomingIpVersion ();
void selectOutgoingIpVersion ();
void forceStreamsDirAccordingToState (SalMediaDescription *md);
bool generateB64CryptoKey (size_t keyLength, char *keyOut, size_t keyOutSize);
- void makeLocalMediaDescription ();
+ void makeLocalMediaDescription (bool localIsOfferer);
int setupEncryptionKey (SalSrtpCryptoAlgo *crypto, MSCryptoSuite suite, unsigned int tag);
void setupDtlsKeys (SalMediaDescription *md);
void setupEncryptionKeys (SalMediaDescription *md);
@@ -177,98 +162,31 @@ private:
void setupImEncryptionEngineParameters (SalMediaDescription *md);
void transferAlreadyAssignedPayloadTypes (SalMediaDescription *oldMd, SalMediaDescription *md);
void updateLocalMediaDescriptionFromIce ();
-
- SalMulticastRole getMulticastRole (SalStreamType type);
- void joinMulticastGroup (int streamIndex, MediaStream *ms);
-
- void configureRtpSession(RtpSession *session, LinphoneStreamType type);
- void setDtlsFingerprint (MSMediaStreamSessions *sessions, const SalStreamDescription *sd, const SalStreamDescription *remote);
- void setDtlsFingerprintOnAllStreams ();
- void setDtlsFingerprintOnAudioStream ();
- void setDtlsFingerprintOnVideoStream ();
- void setDtlsFingerprintOnTextStream ();
- void setupDtlsParams (MediaStream *ms);
- void setZrtpCryptoTypesParameters (MSZrtpParams *params);
- void startDtls (MSMediaStreamSessions *sessions, const SalStreamDescription *sd, const SalStreamDescription *remote);
- //To give a chance for auxilary secret to be used, primary channel (I.E audio) should be started either on 200ok if ZRTP is signaled by a zrtp-hash or when ACK is received in case calling side does not have zrtp-hash.
- void startZrtpPrimaryChannel (const SalStreamDescription *remote);
void startDtlsOnAllStreams ();
- void startDtlsOnAudioStream ();
- void startDtlsOnTextStream ();
- void startDtlsOnVideoStream ();
- void updateStreamCryptoParameters (SalStreamDescription *oldStream, SalStreamDescription *newStream);
- void updateStreamsCryptoParameters (SalMediaDescription *oldMd, SalMediaDescription *newMd);
- bool updateCryptoParameters (const SalStreamDescription *localStreamDesc, SalStreamDescription *oldStream, SalStreamDescription *newStream, MediaStream *ms);
-
- int getIdealAudioBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc);
- int getVideoBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc);
- RtpProfile *makeProfile (const SalMediaDescription *md, const SalStreamDescription *desc, int *usedPt);
- void unsetRtpProfile (int streamIndex);
- void updateAllocatedAudioBandwidth (const PayloadType *pt, int maxbw);
-
- void applyJitterBufferParams (RtpSession *session, LinphoneStreamType type);
- void clearEarlyMediaDestination (MediaStream *ms);
- void clearEarlyMediaDestinations ();
- void configureAdaptiveRateControl (MediaStream *ms, const OrtpPayloadType *pt, bool videoWillBeUsed);
- void configureRtpSessionForRtcpFb (const SalStreamDescription *stream);
- void configureRtpSessionForRtcpXr (SalStreamType type);
- RtpSession *createAudioRtpIoSession ();
- RtpSession *createVideoRtpIoSession ();
+
void freeResources ();
- void handleIceEvents (OrtpEvent *ev);
- void handleStreamEvents (int streamIndex);
- void initializeAudioStream ();
- void initializeTextStream ();
- void initializeVideoStream ();
void prepareEarlyMediaForking ();
- void postConfigureAudioStreams (bool muted);
- void setSymmetricRtp (bool value);
- void setStreamSymmetricRtp(bool value, int streamIndex);
- void setupRingbackPlayer ();
- void startAudioStream (CallSession::State targetState);
- void startStreams (CallSession::State targetState);
- void startStream (SalStreamDescription *streamDesc, int streamIndex, CallSession::State targetState);
- void startTextStream ();
- void startVideoStream (CallSession::State targetState);
- void stopAudioStream ();
- void stopTextStream ();
- void stopVideoStream ();
void tryEarlyMediaForking (SalMediaDescription *md);
void updateStreamFrozenPayloads (SalStreamDescription *resultDesc, SalStreamDescription *localStreamDesc);
void updateFrozenPayloads (SalMediaDescription *result);
- void updateAudioStream (SalMediaDescription *newMd, CallSession::State targetState);
void updateStreams (SalMediaDescription *newMd, CallSession::State targetState);
- void updateTextStream (SalMediaDescription *newMd, CallSession::State targetState);
- void updateVideoStream (SalMediaDescription *newMd, CallSession::State targetState);
- void updateStreamDestination (SalMediaDescription *newMd, SalStreamDescription *newDesc);
- void updateStreamsDestinations (SalMediaDescription *oldMd, SalMediaDescription *newMd);
bool allStreamsAvpfEnabled () const;
bool allStreamsEncrypted () const;
bool atLeastOneStreamStarted () const;
- void audioStreamAuthTokenReady (const std::string &authToken, bool verified);
- void audioStreamEncryptionChanged (bool encrypted);
uint16_t getAvpfRrInterval () const;
unsigned int getNbActiveStreams () const;
- bool isEncryptionMandatory () const;
- int mediaParametersChanged (SalMediaDescription *oldMd, SalMediaDescription *newMd);
void addSecurityEventInChatrooms (const IdentityAddress &faultyDevice, ConferenceSecurityEvent::SecurityEventType securityEventType);
void propagateEncryptionChanged ();
- void fillLogStats (MediaStream *st);
- void updateLocalStats (LinphoneCallStats *stats, MediaStream *stream) const;
- void updateRtpStats (LinphoneCallStats *stats, int streamIndex);
-
void executeBackgroundTasks (bool oneSecondElapsed);
- void reportBandwidth ();
- void reportBandwidthForStream (MediaStream *ms, LinphoneStreamType type);
void abort (const std::string &errorMsg) override;
void handleIncomingReceivedStateInIncomingNotification () override;
- bool isReadyForInvite () const override;
LinphoneStatus pause ();
int restartInvite () override;
void setTerminated () override;
+ void startAccept();
LinphoneStatus startAcceptUpdate (CallSession::State nextState, const std::string &stateInfo) override;
LinphoneStatus startUpdate (const std::string &subject = "") override;
void terminate () override;
@@ -280,6 +198,7 @@ private:
void refreshSockets ();
void reinviteToRecoverFromConnectionLoss () override;
void repairByInviteWithReplaces () override;
+ void addStreamToBundle(SalMediaDescription *md, SalStreamDescription *sd, const char *mid);
#ifdef VIDEO_ENABLED
void videoStreamEventCb (const MSFilter *f, const unsigned int eventId, const void *args);
@@ -288,38 +207,22 @@ private:
int sendDtmf ();
void stunAuthRequestedCb (const char *realm, const char *nonce, const char **username, const char **password, const char **ha1);
+ Stream *getStream(LinphoneStreamType type)const;
private:
static const std::string ecStateStore;
static const int ecStateMaxLen;
+ static constexpr const int rtpExtHeaderMidNumber = 1;
std::weak_ptr<Participant> me;
-
- AudioStream *audioStream = nullptr;
- OrtpEvQueue *audioStreamEvQueue = nullptr;
- LinphoneCallStats *audioStats = nullptr;
- RtpProfile *audioProfile = nullptr;
- RtpProfile *rtpIoAudioProfile = nullptr;
- int mainAudioStreamIndex = LINPHONE_CALL_STATS_AUDIO;
-
- VideoStream *videoStream = nullptr;
- OrtpEvQueue *videoStreamEvQueue = nullptr;
- LinphoneCallStats *videoStats = nullptr;
- RtpProfile *rtpIoVideoProfile = nullptr;
- RtpProfile *videoProfile = nullptr;
- int mainVideoStreamIndex = LINPHONE_CALL_STATS_VIDEO;
- void *videoWindowId = nullptr;
- bool cameraEnabled = true;
-
- TextStream *textStream = nullptr;
- OrtpEvQueue *textStreamEvQueue = nullptr;
- LinphoneCallStats *textStats = nullptr;
- RtpProfile *textProfile = nullptr;
- int mainTextStreamIndex = LINPHONE_CALL_STATS_TEXT;
+
+ std::unique_ptr<StreamsGroup> streamsGroup;
+ int mainAudioStreamIndex = -1;
+ int mainVideoStreamIndex = -1;
+ int mainTextStreamIndex = -1;
LinphoneNatPolicy *natPolicy = nullptr;
std::unique_ptr<StunClient> stunClient;
- std::unique_ptr<IceAgent> iceAgent;
std::vector<std::function<void()>> postProcessHooks;
@@ -330,49 +233,26 @@ private:
belle_sip_source_t *dtmfTimer = nullptr;
std::string mediaLocalIp;
- PortConfig mediaPorts[SAL_MEDIA_DESCRIPTION_MAX_STREAMS];
-
- // The rtp, srtp, zrtp contexts for each stream.
- MSMediaStreamSessions sessions[SAL_MEDIA_DESCRIPTION_MAX_STREAMS];
SalMediaDescription *localDesc = nullptr;
int localDescChanged = 0;
SalMediaDescription *biggestDesc = nullptr;
SalMediaDescription *resultDesc = nullptr;
bool expectMediaInAck = false;
+ int freeStreamIndex = 0;
unsigned int remoteSessionId = 0;
unsigned int remoteSessionVer = 0;
- std::string authToken;
- bool authTokenVerified = false;
std::string dtlsCertificateFingerprint;
- bool forceStreamsReconstruction = false;
-
- unsigned int audioStartCount = 0;
- unsigned int videoStartCount = 0;
- unsigned int textStartCount = 0;
-
// Upload bandwidth setting at the time the call is started. Used to detect if it changes during a call.
int upBandwidth = 0;
- // Upload bandwidth used by audio.
- int audioBandwidth = 0;
-
- bool speakerMuted = false;
- bool microphoneMuted = false;
-
- bool audioMuted = false;
- bool videoMuted = false;
+ bool forceStreamsReconstruction = false;
bool automaticallyPaused = false;
bool pausedByApp = false;
- bool recordActive = false;
bool incomingIceReinvitePending = false;
-
- MSSndCard *currentCaptureCard = nullptr;
- MSSndCard *currentPlayCard = nullptr;
-
- std::string onHoldFile;
+ bool callAcceptanceDefered = false;
L_DECLARE_PUBLIC(MediaSession);
};
diff --git a/src/conference/session/media-session.cpp b/src/conference/session/media-session.cpp
index dc37985a03627c19b336f763d933378013d19ed9..dd92cb70b87a03412e41e6c873c91fbee7c7d706 100644
--- a/src/conference/session/media-session.cpp
+++ b/src/conference/session/media-session.cpp
@@ -17,8 +17,8 @@
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
-#include <iomanip>
-#include <math.h>
+//#include <iomanip>
+//#include <math.h>
#include "address/address-p.h"
#include "call/call-p.h"
@@ -42,8 +42,7 @@
#include <mediastreamer2/msequalizer.h>
#include <mediastreamer2/mseventqueue.h>
#include <mediastreamer2/msfileplayer.h>
-#include <mediastreamer2/msjpegwriter.h>
-#include <mediastreamer2/msogl.h>
+
#include <mediastreamer2/msrtt4103.h>
#include <mediastreamer2/msvolume.h>
#include <ortp/b64.h>
@@ -71,6 +70,15 @@ const int MediaSessionPrivate::ecStateMaxLen = 1048576; /* 1Mo */
// =============================================================================
+
+void MediaSessionPrivate::setDtlsFingerprint(const std::string &fingerPrint){
+ dtlsCertificateFingerprint = fingerPrint;
+}
+
+const std::string & MediaSessionPrivate::getDtlsFingerprint()const{
+ return dtlsCertificateFingerprint;
+}
+
void MediaSessionPrivate::stunAuthRequestedCb (void *userData, const char *realm, const char *nonce, const char **username, const char **password, const char **ha1) {
MediaSessionPrivate *msp = reinterpret_cast<MediaSessionPrivate *>(userData);
msp->stunAuthRequestedCb(realm, nonce, username, password, ha1);
@@ -83,6 +91,34 @@ void MediaSessionPrivate::accepted () {
CallSessionPrivate::accepted();
LinphoneTaskList tl;
linphone_task_list_init(&tl);
+
+ switch (state){
+ case CallSession::State::OutgoingProgress:
+ case CallSession::State::OutgoingRinging:
+ case CallSession::State::OutgoingEarlyMedia:
+ case CallSession::State::Connected:
+ if (q->getCore()->getCCore()->sip_conf.sdp_200_ack){
+ lInfo() << "Initializing local media description according to remote offer in 200Ok";
+ // We were waiting for an incoming offer. Now prepare the local media description according to remote offer.
+ initializeParamsAccordingToIncomingCallParams();
+ makeLocalMediaDescription(op->getRemoteMediaDescription() ? false : true);
+ /*
+ * If ICE is enabled, we'll have to do the prepare() step, however since defering the sending of the ACK is complicated and
+ * confusing from a signaling standpoint, ICE we will skip the STUN gathering by not giving enough time
+ * for the gathering step. Only local candidates will be answered in the ACK.
+ */
+ if (getStreamsGroup().prepare()){
+ lWarning() << "Some gathering is needed for ICE, however since a defered sending of ACK is not supported"
+ " the ICE gathering will only contain local candidates.";
+ }
+ getStreamsGroup().finishPrepare();
+ updateLocalMediaDescriptionFromIce();
+ }
+ break;
+ default:
+ break;
+ }
+
/* Reset the internal call update flag, so it doesn't risk to be copied and used in further re-INVITEs */
getParams()->getPrivate()->setInternalCallUpdate(false);
SalMediaDescription *rmd = op->getRemoteMediaDescription();
@@ -95,10 +131,6 @@ void MediaSessionPrivate::accepted () {
md = nullptr;
if (md) {
/* There is a valid SDP in the response, either offer or answer, and we're able to start/update the streams */
- if (rmd) {
- /* Handle remote ICE attributes if any. */
- iceAgent->updateFromRemoteMediaDescription(localDesc, rmd, !op->isOfferer());
- }
CallSession::State nextState = CallSession::State::Idle;
string nextStateMsg;
switch (state) {
@@ -141,9 +173,9 @@ void MediaSessionPrivate::accepted () {
lError() << "BUG: nextState is not set in accepted(), current state is " << Utils::toString(state);
else {
updateRemoteSessionIdAndVer();
- iceAgent->updateIceStateInCallStats();
+ //getIceAgent().updateIceStateInCallStats();
updateStreams(md, nextState);
- fixCallParams(rmd);
+ fixCallParams(rmd, false);
setState(nextState, nextStateMsg);
}
} else { /* Invalid or no SDP */
@@ -179,7 +211,6 @@ void MediaSessionPrivate::accepted () {
}
void MediaSessionPrivate::ackReceived (LinphoneHeaders *headers) {
- L_Q();
CallSessionPrivate::ackReceived(headers);
if (expectMediaInAck) {
switch (state) {
@@ -192,15 +223,7 @@ void MediaSessionPrivate::ackReceived (LinphoneHeaders *headers) {
}
accepted();
}
- if (linphone_core_media_encryption_supported(q->getCore()->getCCore(), LinphoneMediaEncryptionZRTP)) {
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- const SalStreamDescription *remoteStream = remote?sal_media_description_find_best_stream(remote, SalAudio):NULL;
- //Start zrtp if remote has not offered it but local is configured for zrtp and not offerer
- if (remoteStream && getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP && !op->isOfferer() && remoteStream->haveZrtpHash == 0) {
- lInfo() << "Starting zrtp late";
- startZrtpPrimaryChannel(remoteStream);
- }
- }
+ getStreamsGroup().sessionConfirmed(getStreamsGroup().getCurrentOfferAnswerContext());
}
void MediaSessionPrivate::dtmfReceived (char dtmf) {
@@ -222,7 +245,7 @@ bool MediaSessionPrivate::failure () {
if ((state == CallSession::State::OutgoingInit) || (state == CallSession::State::OutgoingProgress)
|| (state == CallSession::State::OutgoingRinging) /* Push notification case */ || (state == CallSession::State::OutgoingEarlyMedia)) {
for (int i = 0; i < localDesc->nb_streams; i++) {
- if (!sal_stream_description_active(&localDesc->streams[i]))
+ if (!sal_stream_description_enabled(&localDesc->streams[i]))
continue;
if (getParams()->getMediaEncryption() == LinphoneMediaEncryptionSRTP) {
if (getParams()->avpfEnabled()) {
@@ -263,13 +286,11 @@ bool MediaSessionPrivate::failure () {
&MediaSessionPrivate::resumeAfterFailedTransfer, referer.get(),
"Automatic CallSession resuming after failed transfer");
}
-
q->getCore()->getPrivate()->getToneManager()->stop(q->getSharedFromThis());
if (ei->reason != SalReasonNone)
q->getCore()->getPrivate()->getToneManager()->startErrorTone(q->getSharedFromThis(), linphone_reason_from_sal(ei->reason));
- stopStreams();
return false;
}
@@ -298,26 +319,22 @@ void MediaSessionPrivate::remoteRinging () {
/* Initialize the remote call params by invoking linphone_call_get_remote_params(). This is useful as the SDP may not be present in the 200Ok */
q->getRemoteParams();
/* Accept early media */
- if ((audioStream && audio_stream_started(audioStream))
-#ifdef VIDEO_ENABLED
- || (videoStream && video_stream_started(videoStream))
-#endif
- ) {
- /* Streams already started */
- tryEarlyMediaForking(md);
-#ifdef VIDEO_ENABLED
- if (videoStream)
- video_stream_send_vfu(videoStream); /* Request for iframe */
-#endif
+
+ if (getStreamsGroup().isStarted()){
+ OfferAnswerContext ctx;
+ ctx.localMediaDescription = localDesc;
+ ctx.resultMediaDescription = md;
+ ctx.remoteMediaDescription = rmd;
+ getStreamsGroup().tryEarlyMediaForking(ctx);
return;
}
setState(CallSession::State::OutgoingEarlyMedia, "Early media");
q->getCore()->getPrivate()->getToneManager()->stop(q->getSharedFromThis());
lInfo() << "Doing early media...";
- iceAgent->updateFromRemoteMediaDescription(localDesc, rmd, !op->isOfferer());
updateStreams(md, state);
- if ((q->getCurrentParams()->getAudioDirection() == LinphoneMediaDirectionInactive) && audioStream) {
+
+ if ((q->getCurrentParams()->getAudioDirection() == LinphoneMediaDirectionInactive)) {
q->getCore()->getPrivate()->getToneManager()->startRingbackTone(q->getSharedFromThis());
}
} else {
@@ -404,19 +421,19 @@ void MediaSessionPrivate::updated (bool isUpdate) {
-void MediaSessionPrivate::updating (bool isUpdate) {
+void MediaSessionPrivate::updating(bool isUpdate) {
L_Q();
SalMediaDescription *rmd = op->getRemoteMediaDescription();
- fixCallParams(rmd);
+ fixCallParams(rmd, true);
if (state != CallSession::State::Paused) {
/* Refresh the local description, but in paused state, we don't change anything. */
if (!rmd && lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sip", "sdp_200_ack_follow_video_policy", 0)) {
lInfo() << "Applying default policy for offering SDP on CallSession [" << q << "]";
setParams(new MediaSessionParams());
- params->initDefault(q->getCore());
+ // Yes we init parameters as if we were in the case of an outgoing call, because it is a resume with no SDP.
+ params->initDefault(q->getCore(), LinphoneCallOutgoing);
}
- makeLocalMediaDescription();
- op->setLocalMediaDescription(localDesc);
+ makeLocalMediaDescription(false);
}
if (rmd) {
SalErrorInfo sei;
@@ -451,122 +468,43 @@ void MediaSessionPrivate::updating (bool isUpdate) {
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::enableSymmetricRtp (bool value) {
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (sessions[i].rtp_session)
- rtp_session_set_symmetric_rtp(sessions[i].rtp_session, value);
- }
-}
-void MediaSessionPrivate::oglRender () const {
+void MediaSessionPrivate::oglRender () {
#ifdef VIDEO_ENABLED
- if (videoStream && videoStream->output && (ms_filter_get_id(videoStream->output) == MS_OGL_ID))
- ms_filter_call_method(videoStream->output, MS_OGL_RENDER, nullptr);
+ if (mainVideoStreamIndex != -1){
+ MS2VideoStream * vs = dynamic_cast<MS2VideoStream*>(getStreamsGroup().getStream(mainVideoStreamIndex));
+ if (vs) vs->oglRender();
+ }
#endif
}
void MediaSessionPrivate::sendVfu () {
-#ifdef VIDEO_ENABLED
- if (videoStream)
- video_stream_send_vfu(videoStream);
-#endif
-}
-
-// -----------------------------------------------------------------------------
-
-void MediaSessionPrivate::clearIceCheckList (IceCheckList *cl) {
- if (audioStream && audioStream->ms.ice_check_list == cl)
- audioStream->ms.ice_check_list = nullptr;
- if (videoStream && videoStream->ms.ice_check_list == cl)
- videoStream->ms.ice_check_list = nullptr;
- if (textStream && textStream->ms.ice_check_list == cl)
- textStream->ms.ice_check_list = nullptr;
-}
-
-void MediaSessionPrivate::deactivateIce () {
- if (audioStream)
- audioStream->ms.ice_check_list = nullptr;
- if (videoStream)
- videoStream->ms.ice_check_list = nullptr;
- if (textStream)
- textStream->ms.ice_check_list = nullptr;
- _linphone_call_stats_set_ice_state(audioStats, LinphoneIceStateNotActivated);
- _linphone_call_stats_set_ice_state(videoStats, LinphoneIceStateNotActivated);
- _linphone_call_stats_set_ice_state(textStats, LinphoneIceStateNotActivated);
- stopStreamsForIceGathering();
-}
-
-void MediaSessionPrivate::prepareStreamsForIceGathering (bool hasVideo) {
- if (audioStream->ms.state == MSStreamInitialized)
- audio_stream_prepare_sound(audioStream, nullptr, nullptr);
-#ifdef VIDEO_ENABLED
- if (hasVideo && videoStream && (videoStream->ms.state == MSStreamInitialized))
- video_stream_prepare_video(videoStream);
-#endif
- if (getParams()->realtimeTextEnabled() && (textStream->ms.state == MSStreamInitialized))
- text_stream_prepare_text(textStream);
+ getStreamsGroup().forEach<VideoControlInterface>([](VideoControlInterface *i){ i->sendVfu(); });
}
-void MediaSessionPrivate::stopStreamsForIceGathering () {
- if (audioStream && (audioStream->ms.state == MSStreamPreparing))
- audio_stream_unprepare_sound(audioStream);
-#ifdef VIDEO_ENABLED
- if (videoStream && (videoStream->ms.state == MSStreamPreparing))
- video_stream_unprepare_video(videoStream);
-#endif
- if (textStream && (textStream->ms.state == MSStreamPreparing))
- text_stream_unprepare_text(textStream);
-}
// -----------------------------------------------------------------------------
bool MediaSessionPrivate::getSpeakerMuted () const {
- return speakerMuted;
+ AudioControlInterface *i = getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ return i ? !i->speakerEnabled() : false;
}
void MediaSessionPrivate::setSpeakerMuted (bool muted) {
- if (speakerMuted == muted)
- return;
- speakerMuted = muted;
-
- if (state == CallSession::State::StreamsRunning)
- forceSpeakerMuted(speakerMuted);
-}
-
-void MediaSessionPrivate::forceSpeakerMuted (bool muted) {
- L_Q();
-
- if (!audioStream)
- return;
-
- if (muted)
- audio_stream_set_spk_gain(audioStream, 0);
- else
- audio_stream_set_spk_gain_db(audioStream, q->getCore()->getCCore()->sound_conf.soft_play_lev);
+ AudioControlInterface *i = getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (i) i->enableSpeaker(!muted);
}
// -----------------------------------------------------------------------------
bool MediaSessionPrivate::getMicrophoneMuted () const {
- return microphoneMuted;
+ AudioControlInterface *i = getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ return i ? !i->micEnabled() : false;
}
void MediaSessionPrivate::setMicrophoneMuted (bool muted) {
- L_Q();
-
- if (microphoneMuted == muted)
- return;
- microphoneMuted = muted;
-
- if (!audioStream)
- return;
-
- if (state == CallSession::State::StreamsRunning) {
- if (microphoneMuted)
- audio_stream_set_mic_gain(audioStream, 0);
- else
- audio_stream_set_mic_gain_db(audioStream, q->getCore()->getCCore()->sound_conf.soft_mic_lev);
- }
+ AudioControlInterface *i = getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (i) i->enableMic(muted);
}
// -----------------------------------------------------------------------------
@@ -589,203 +527,36 @@ void MediaSessionPrivate::setRemoteParams (MediaSessionParams *msp) {
remoteParams = msp;
}
-MediaStream *MediaSessionPrivate::getMediaStream (LinphoneStreamType type) const {
- return getMediaStream(int(type));
-}
-
-int MediaSessionPrivate::getRtcpPort (LinphoneStreamType type) const {
- return mediaPorts[getStreamIndex(getMediaStream(type))].rtcpPort;
-}
-
-int MediaSessionPrivate::getRtpPort (LinphoneStreamType type) const {
- return mediaPorts[getStreamIndex(getMediaStream(type))].rtpPort;
-}
-
-LinphoneCallStats * MediaSessionPrivate::getStats (LinphoneStreamType type) const {
+Stream *MediaSessionPrivate::getStream(LinphoneStreamType type)const{
switch (type) {
case LinphoneStreamTypeAudio:
- return audioStats;
+ return getStreamsGroup().lookupMainStream(SalAudio);
case LinphoneStreamTypeVideo:
- return videoStats;
+ return getStreamsGroup().lookupMainStream(SalVideo);
case LinphoneStreamTypeText:
- return textStats;
+ return getStreamsGroup().lookupMainStream(SalText);
case LinphoneStreamTypeUnknown:
- default:
- return nullptr;
+ break;
}
+ return nullptr;
}
-int MediaSessionPrivate::getStreamIndex (LinphoneStreamType type) const {
- return getStreamIndex(getMediaStream(type));
-}
-
-int MediaSessionPrivate::getStreamIndex (MediaStream *ms) const {
- if (ms == &audioStream->ms)
- return mainAudioStreamIndex;
- else if (ms == &videoStream->ms)
- return mainVideoStreamIndex;
- else if (ms == &textStream->ms)
- return mainTextStreamIndex;
- return -1;
-}
-
-MSWebCam * MediaSessionPrivate::getVideoDevice () const {
- L_Q();
- bool paused = (state == CallSession::State::Pausing) || (state == CallSession::State::Paused);
- if (paused || videoMuted || !cameraEnabled)
-#ifdef VIDEO_ENABLED
- return ms_web_cam_manager_get_cam(ms_factory_get_web_cam_manager(q->getCore()->getCCore()->factory),
- "StaticImage: Static picture");
-#else
- return nullptr;
-#endif
- else
- return q->getCore()->getCCore()->video_conf.device;
+LinphoneCallStats * MediaSessionPrivate::getStats(LinphoneStreamType type) const {
+ Stream *s = getStream(type);
+ if (s) return s->getStats();
+ lError() << "There is no stats for main stream of type " << linphone_stream_type_to_string(type) << " because this stream doesn't exist.";
+ return nullptr;
}
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::initializeStreams () {
- initializeAudioStream();
- initializeVideoStream();
- initializeTextStream();
-}
-
-void MediaSessionPrivate::stopStream (SalStreamDescription *streamDesc) {
- L_Q();
-
- if (streamDesc->type == SalAudio && audioStream) {
- if (videoStream)
- audio_stream_unlink_video(audioStream, videoStream);
- stopAudioStream();
-
- if (q->getCore()->getCCore()->msevq)
- ms_event_queue_skip(q->getCore()->getCCore()->msevq);
-
- if (audioProfile) {
- rtp_profile_destroy(audioProfile);
- audioProfile = nullptr;
- unsetRtpProfile(mainAudioStreamIndex);
- }
-
- if (rtpIoAudioProfile) {
- rtp_profile_destroy(rtpIoAudioProfile);
- rtpIoAudioProfile = nullptr;
- }
-
- q->getCore()->soundcardHintCheck();
- } else if (streamDesc->type == SalVideo && videoStream) {
- if (audioStream)
- audio_stream_unlink_video(audioStream, videoStream);
- stopVideoStream();
-
- if (q->getCore()->getCCore()->msevq)
- ms_event_queue_skip(q->getCore()->getCCore()->msevq);
-
- if (videoProfile) {
- rtp_profile_destroy(videoProfile);
- videoProfile = nullptr;
- unsetRtpProfile(mainVideoStreamIndex);
- }
-
- if (rtpIoVideoProfile) {
- rtp_profile_destroy(rtpIoVideoProfile);
- rtpIoVideoProfile = nullptr;
- }
- } else if (streamDesc->type == SalText && textStream) {
- stopTextStream();
-
- if (q->getCore()->getCCore()->msevq)
- ms_event_queue_skip(q->getCore()->getCCore()->msevq);
-
- if (textProfile) {
- rtp_profile_destroy(textProfile);
- textProfile = nullptr;
- unsetRtpProfile(mainTextStreamIndex);
- }
- }
-}
void MediaSessionPrivate::stopStreams () {
L_Q();
- if (audioStream || videoStream || textStream) {
- if (audioStream && videoStream)
- audio_stream_unlink_video(audioStream, videoStream);
- stopAudioStream();
- stopVideoStream();
- stopTextStream();
- if (q->getCore()->getCCore()->msevq)
- ms_event_queue_skip(q->getCore()->getCCore()->msevq);
- }
-
- if (audioProfile) {
- rtp_profile_destroy(audioProfile);
- audioProfile = nullptr;
- unsetRtpProfile(mainAudioStreamIndex);
- }
- if (videoProfile) {
- rtp_profile_destroy(videoProfile);
- videoProfile = nullptr;
- unsetRtpProfile(mainVideoStreamIndex);
- }
- if (textProfile) {
- rtp_profile_destroy(textProfile);
- textProfile = nullptr;
- unsetRtpProfile(mainTextStreamIndex);
- }
- if (rtpIoAudioProfile) {
- rtp_profile_destroy(rtpIoAudioProfile);
- rtpIoAudioProfile = nullptr;
- }
- if (rtpIoVideoProfile) {
- rtp_profile_destroy(rtpIoVideoProfile);
- rtpIoVideoProfile = nullptr;
- }
-
+ if (getStreamsGroup().isStarted()) getStreamsGroup().stop();
q->getCore()->soundcardHintCheck();
}
-void MediaSessionPrivate::restartStream (SalStreamDescription *streamDesc, int streamIndex, int sdChanged, CallSession::State targetState) {
- L_Q();
- string streamTypeName = sal_stream_description_get_type_as_string(streamDesc);
-
- stopStream(streamDesc);
-
- if (streamDesc->type == SalAudio) {
- if (sdChanged & SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED) {
- lInfo() << "Media ip type has changed, destroying sessions context on CallSession [" << q << "] for " << streamTypeName << " stream";
- ms_media_stream_sessions_uninit(&sessions[mainAudioStreamIndex]);
- }
-
- initializeAudioStream();
- } else if (streamDesc->type == SalVideo) {
- if (sdChanged & SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED) {
- lInfo() << "Media ip type has changed, destroying sessions context on CallSession [" << q << "] for " << streamTypeName << " stream";
- ms_media_stream_sessions_uninit(&sessions[mainVideoStreamIndex]);
- }
-
- initializeVideoStream();
- } else if (streamDesc->type == SalText) {
- if (sdChanged & SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED) {
- lInfo() << "Media ip type has changed, destroying sessions context on CallSession [" << q << "] for " << streamTypeName << " stream";
- ms_media_stream_sessions_uninit(&sessions[mainTextStreamIndex]);
- }
-
- initializeTextStream();
- }
-
- if (getParams()->earlyMediaSendingEnabled() && (state == CallSession::State::OutgoingEarlyMedia)) {
- if (streamDesc->type == SalAudio && audioStream)
- rtp_session_set_symmetric_rtp(audioStream->ms.sessions.rtp_session, false);
- else if (streamDesc->type == SalVideo && videoStream)
- rtp_session_set_symmetric_rtp(videoStream->ms.sessions.rtp_session, false);
- }
-
- startStream(streamDesc, streamIndex, targetState);
-
- updateStreamFrozenPayloads(streamDesc, &localDesc->streams[streamIndex]);
-}
-
// -----------------------------------------------------------------------------
void MediaSessionPrivate::onNetworkReachable (bool sipNetworkReachable, bool mediaNetworkReachable) {
@@ -802,19 +573,6 @@ void MediaSessionPrivate::onNetworkReachable (bool sipNetworkReachable, bool med
// -----------------------------------------------------------------------------
-OrtpJitterBufferAlgorithm MediaSessionPrivate::jitterBufferNameToAlgo (const string &name) {
- if (name == "basic") return OrtpJitterBufferBasic;
- if (name == "rls") return OrtpJitterBufferRecursiveLeastSquare;
- lError() << "Invalid jitter buffer algorithm: " << name;
- return OrtpJitterBufferRecursiveLeastSquare;
-}
-
-#ifdef VIDEO_ENABLED
-void MediaSessionPrivate::videoStreamEventCb (void *userData, const MSFilter *f, const unsigned int eventId, const void *args) {
- MediaSessionPrivate *msp = reinterpret_cast<MediaSessionPrivate *>(userData);
- msp->videoStreamEventCb(f, eventId, args);
-}
-#endif
#ifdef TEST_EXT_RENDERER
void MediaSessionPrivate::extRendererCb (void *userData, const MSPicture *local, const MSPicture *remote) {
@@ -823,10 +581,6 @@ void MediaSessionPrivate::extRendererCb (void *userData, const MSPicture *local,
}
#endif
-void MediaSessionPrivate::realTimeTextCharacterReceived (void *userData, MSFilter *f, unsigned int id, void *arg) {
- MediaSessionPrivate *msp = reinterpret_cast<MediaSessionPrivate *>(userData);
- msp->realTimeTextCharacterReceived(f, id, arg);
-}
int MediaSessionPrivate::sendDtmf (void *data, unsigned int revents) {
MediaSession *session = reinterpret_cast<MediaSession *>(data);
@@ -835,18 +589,7 @@ int MediaSessionPrivate::sendDtmf (void *data, unsigned int revents) {
// -----------------------------------------------------------------------------
-float MediaSessionPrivate::aggregateQualityRatings (float audioRating, float videoRating) {
- float result;
- if ((audioRating < 0) && (videoRating < 0))
- result = -1;
- else if (audioRating < 0)
- result = videoRating * 5.0f;
- else if (videoRating < 0)
- result = audioRating * 5.0f;
- else
- result = audioRating * videoRating * 5.0f;
- return result;
-}
+
// -----------------------------------------------------------------------------
@@ -875,7 +618,7 @@ void MediaSessionPrivate::setState (CallSession::State newState, const string &m
// Handle specifically the case of an incoming ICE-concluded reINVITE
lInfo() << "Checking for ICE reINVITE";
rmd = op->getRemoteMediaDescription();
- if (iceAgent && rmd && iceAgent->checkIceReinviteNeedsDeferedResponse(rmd)) {
+ if (rmd && getIceService().reinviteNeedsDeferedResponse(rmd)) {
deferUpdate = true;
deferUpdateInternal = true;
incomingIceReinvitePending = true;
@@ -889,106 +632,47 @@ void MediaSessionPrivate::setState (CallSession::State newState, const string &m
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::computeStreamsIndexes (const SalMediaDescription *md) {
- bool audioFound = false;
- bool videoFound = false;
- bool textFound = false;
- for (int i = 0; i < md->nb_streams; i++) {
- if (md->streams[i].type == SalAudio) {
- if (audioFound)
- lInfo() << "audio stream index found: " << i << ", but main audio stream already set to " << mainAudioStreamIndex;
- else {
- mainAudioStreamIndex = i;
- audioFound = true;
- lInfo() << "audio stream index found: " << i << ", updating main audio stream index";
- }
- /* Check that the default value of a another stream doesn't match the new one */
- if (i == mainVideoStreamIndex) {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
- if (sal_stream_description_active(&md->streams[j]))
- continue;
- if ((j != mainVideoStreamIndex) && (j != mainTextStreamIndex)) {
- lInfo() << i << " was used for video stream ; now using " << j;
- mainVideoStreamIndex = j;
- break;
- }
- }
- }
- if (i == mainTextStreamIndex) {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
- if (sal_stream_description_active(&md->streams[j]))
- continue;
- if ((j != mainVideoStreamIndex) && (j != mainTextStreamIndex)) {
- lInfo() << i << " was used for text stream ; now using " << j;
- mainTextStreamIndex = j;
- break;
- }
- }
- }
- } else if (md->streams[i].type == SalVideo) {
- if (videoFound)
- lInfo() << "video stream index found: " << i << ", but main video stream already set to " << mainVideoStreamIndex;
- else {
- mainVideoStreamIndex = i;
- videoFound = true;
- lInfo() << "video stream index found: " << i << ", updating main video stream index";
- }
- /* Check that the default value of a another stream doesn't match the new one */
- if (i == mainAudioStreamIndex) {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
- if (sal_stream_description_active(&md->streams[j]))
- continue;
- if ((j != mainAudioStreamIndex) && (j != mainTextStreamIndex)) {
- lInfo() << i << " was used for audio stream ; now using " << j;
- mainAudioStreamIndex = j;
- break;
- }
- }
- }
- if (i == mainTextStreamIndex) {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
- if (sal_stream_description_active(&md->streams[j]))
- continue;
- if ((j != mainAudioStreamIndex) && (j != mainTextStreamIndex)) {
- lInfo() << i << " was used for text stream ; now using " << j;
- mainTextStreamIndex = j;
- break;
- }
- }
- }
- } else if (md->streams[i].type == SalText) {
- if (textFound)
- lInfo() << "text stream index found: " << i << ", but main text stream already set to " << mainTextStreamIndex;
- else {
- mainTextStreamIndex = i;
- textFound = true;
- lInfo() << "text stream index found: " << i << ", updating main text stream index";
- }
- /* Check that the default value of a another stream doesn't match the new one */
- if (i == mainAudioStreamIndex) {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
- if (sal_stream_description_active(&md->streams[j]))
- continue;
- if ((j != mainVideoStreamIndex) && (j != mainAudioStreamIndex)) {
- lInfo() << i << " was used for audio stream ; now using " << j;
- mainAudioStreamIndex = j;
- break;
- }
- }
- }
- if (i == mainVideoStreamIndex) {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
- if (sal_stream_description_active(&md->streams[j]))
- continue;
- if ((j != mainVideoStreamIndex) && (j != mainAudioStreamIndex)) {
- lInfo() << i << " was used for video stream ; now using " << j;
- mainVideoStreamIndex = j;
- break;
- }
- }
- }
- }
+
+int MediaSessionPrivate::getFirstStreamWithType(const SalMediaDescription *md, SalStreamType type){
+ int i;
+ for (i = 0; i < md->nb_streams; ++i) {
+ if (md->streams[i].type == type) return i;
+ }
+ return -1;
+}
+
+void MediaSessionPrivate::assignStreamsIndexes(){
+ if (biggestDesc && freeStreamIndex < biggestDesc->nb_streams) freeStreamIndex = biggestDesc->nb_streams;
+
+ /*Initialize stream indexes from potential incoming offer.*/
+ SalMediaDescription *rmd = op ? op->getRemoteMediaDescription() : nullptr;
+ if (rmd) assignStreamsIndexesIncoming(rmd);
+
+ /*Assign indexes for our streams, if no incoming offer was received, or if new streams are requested.*/
+ if (getParams()->audioEnabled() && mainAudioStreamIndex == -1){
+ mainAudioStreamIndex = freeStreamIndex++;
+ }
+ if (getParams()->videoEnabled() && mainVideoStreamIndex == -1){
+ mainVideoStreamIndex = freeStreamIndex++;
+ }
+ if (getParams()->realtimeTextEnabled() && mainTextStreamIndex == -1){
+ mainTextStreamIndex = freeStreamIndex++;
+ }
+ lInfo() << "Stream indexes selected (-1 = unassigned): mainAudioStreamIndex=" << mainAudioStreamIndex <<
+ ", mainVideoStreamIndex=" << mainVideoStreamIndex << ", mainTextStreamIndex=" << mainTextStreamIndex;
+}
+
+void MediaSessionPrivate::assignStreamsIndexesIncoming(const SalMediaDescription *md) {
+ if (mainAudioStreamIndex == -1){
+ mainAudioStreamIndex = getFirstStreamWithType(md, SalAudio);
+ }
+ if (mainVideoStreamIndex == -1){
+ mainVideoStreamIndex = getFirstStreamWithType(md, SalVideo);
}
+ if (mainTextStreamIndex == -1){
+ mainTextStreamIndex = getFirstStreamWithType(md, SalText);
+ }
+ if (freeStreamIndex < md->nb_streams) freeStreamIndex = md->nb_streams;
}
/*
@@ -997,59 +681,57 @@ void MediaSessionPrivate::computeStreamsIndexes (const SalMediaDescription *md)
* - the video enablement parameter according to what is offered and our local policy.
* Fixing the params to proper values avoid request video by accident during internal call updates, pauses and resumes
*/
-void MediaSessionPrivate::fixCallParams (SalMediaDescription *rmd) {
+void MediaSessionPrivate::fixCallParams (SalMediaDescription *rmd, bool fromOffer) {
L_Q();
- if (rmd) {
- computeStreamsIndexes(rmd);
- updateBiggestDesc(rmd);
- /* Why disabling implicit_rtcp_fb ? It is a local policy choice actually. It doesn't disturb to propose it again and again
- * even if the other end apparently doesn't support it.
- * The following line of code is causing trouble, while for example making an audio call, then adding video.
- * Due to the 200Ok response of the audio-only offer where no rtcp-fb attribute is present, implicit_rtcp_fb is set to
- * false, which is then preventing it to be eventually used when video is later added to the call.
- * I did the choice of commenting it out.
- */
- /*params.getPrivate()->enableImplicitRtcpFb(params.getPrivate()->implicitRtcpFbEnabled() & sal_media_description_has_implicit_avpf(rmd));*/
- }
+ if (!rmd) return;
+
+ updateBiggestDesc(rmd);
+ /* Why disabling implicit_rtcp_fb ? It is a local policy choice actually. It doesn't disturb to propose it again and again
+ * even if the other end apparently doesn't support it.
+ * The following line of code is causing trouble, while for example making an audio call, then adding video.
+ * Due to the 200Ok response of the audio-only offer where no rtcp-fb attribute is present, implicit_rtcp_fb is set to
+ * false, which is then preventing it to be eventually used when video is later added to the call.
+ * I did the choice of commenting it out.
+ */
+ /*params.getPrivate()->enableImplicitRtcpFb(params.getPrivate()->implicitRtcpFbEnabled() & sal_media_description_has_implicit_avpf(rmd));*/
const MediaSessionParams *rcp = q->getRemoteParams();
if (rcp) {
- if (getParams()->audioEnabled() && !rcp->audioEnabled()) {
- lInfo() << "CallSession [" << q << "]: disabling audio in our call params because the remote doesn't want it";
- getParams()->enableAudio(false);
- }
- if (getParams()->videoEnabled() && !rcp->videoEnabled()) {
- lInfo() << "CallSession [" << q << "]: disabling video in our call params because the remote doesn't want it";
- getParams()->enableVideo(false);
+ if (!fromOffer){
+ /*
+ * This is to avoid to re-propose again some streams that have just been declined.
+ */
+ if (getParams()->audioEnabled() && !rcp->audioEnabled()) {
+ lInfo() << "CallSession [" << q << "]: disabling audio in our call params because the remote doesn't want it";
+ getParams()->enableAudio(false);
+ }
+ if (getParams()->videoEnabled() && !rcp->videoEnabled()) {
+ lInfo() << "CallSession [" << q << "]: disabling video in our call params because the remote doesn't want it";
+ getParams()->enableVideo(false);
+ }
+ if (getParams()->realtimeTextEnabled() && !rcp->realtimeTextEnabled()) {
+ lInfo() << "CallSession [" << q << "]: disabling RTT in our call params because the remote doesn't want it";
+ getParams()->enableRealtimeText(false);
+ }
}
+ // Real Time Text is always by default accepted when proposed.
+ if (!getParams()->realtimeTextEnabled() && rcp->realtimeTextEnabled())
+ getParams()->enableRealtimeText(true);
+
if (rcp->videoEnabled() && q->getCore()->getCCore()->video_policy.automatically_accept && linphone_core_video_enabled(q->getCore()->getCCore()) && !getParams()->videoEnabled()) {
lInfo() << "CallSession [" << q << "]: re-enabling video in our call params because the remote wants it and the policy allows to automatically accept";
getParams()->enableVideo(true);
}
- if (rcp->realtimeTextEnabled() && !getParams()->realtimeTextEnabled())
- getParams()->enableRealtimeText(true);
+
}
}
void MediaSessionPrivate::initializeParamsAccordingToIncomingCallParams () {
- L_Q();
CallSessionPrivate::initializeParamsAccordingToIncomingCallParams();
- getCurrentParams()->getPrivate()->setUpdateCallWhenIceCompleted(getParams()->getPrivate()->getUpdateCallWhenIceCompleted());
- getParams()->enableVideo(linphone_core_video_enabled(q->getCore()->getCCore()) && q->getCore()->getCCore()->video_policy.automatically_accept);
SalMediaDescription *md = op->getRemoteMediaDescription();
if (md) {
+ assignStreamsIndexesIncoming(md);
/* It is licit to receive an INVITE without SDP, in this case WE choose the media parameters according to policy */
setCompatibleIncomingCallParams(md);
- /* Set multicast role & address if any */
- if (!op->isOfferer()) {
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (md->streams[i].dir == SalStreamInactive)
- continue;
- if ((md->streams[i].rtp_addr[0] != '\0') && ms_is_multicast(md->streams[i].rtp_addr)) {
- md->streams[i].multicast_role = SalMulticastReceiver;
- mediaPorts[i].multicastIp = md->streams[i].rtp_addr;
- }
- }
- }
}
}
@@ -1079,18 +761,35 @@ void MediaSessionPrivate::setCompatibleIncomingCallParams (SalMediaDescription *
if (!mandatory || (mandatory && linphone_core_get_media_encryption(lc) == LinphoneMediaEncryptionNone))
getParams()->setMediaEncryption(LinphoneMediaEncryptionNone);
}
+ if (mainAudioStreamIndex != -1){
+ SalStreamDescription *sd = &md->streams[mainAudioStreamIndex];
+ const char *rtpAddr = (sd->rtp_addr[0] != '\0') ? sd->rtp_addr : md->addr;
+ if (ms_is_multicast(rtpAddr)){
+ lInfo() << "Incoming offer has audio multicast, enabling it in local params.";
+ getParams()->enableAudioMulticast(true);
+ }else getParams()->enableAudioMulticast(false);
+ }
+ if (mainVideoStreamIndex != -1){
+ SalStreamDescription *sd = &md->streams[mainVideoStreamIndex];
+ const char *rtpAddr = (sd->rtp_addr[0] != '\0') ? sd->rtp_addr : md->addr;
+ if (ms_is_multicast(rtpAddr)){
+ lInfo() << "Incoming offer has video multicast, enabling it in local params.";
+ getParams()->enableVideoMulticast(true);
+ }else getParams()->enableVideoMulticast(false);
+ }
+
/* In case of nat64, even ipv4 addresses are reachable from v6. Should be enhanced to manage stream by stream connectivity (I.E v6 or v4) */
/*if (!sal_media_description_has_ipv6(md)){
lInfo() << "The remote SDP doesn't seem to offer any IPv6 connectivity, so disabling IPv6 for this call";
af = AF_INET;
}*/
- fixCallParams(md);
+ fixCallParams(md, true);
}
void MediaSessionPrivate::updateBiggestDesc (SalMediaDescription *md) {
if (!biggestDesc || (md->nb_streams > biggestDesc->nb_streams)) {
/* We have been offered and now are ready to proceed, or we added a new stream,
- * store the media description to remember the mapping of calls */
+ * store the media description to remember the mapping of streams within this call. */
if (biggestDesc) {
sal_media_description_unref(biggestDesc);
biggestDesc = nullptr;
@@ -1109,214 +808,42 @@ void MediaSessionPrivate::updateRemoteSessionIdAndVer () {
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::initStats (LinphoneCallStats *stats, LinphoneStreamType type) {
- _linphone_call_stats_set_type(stats, type);
- _linphone_call_stats_set_received_rtcp(stats, nullptr);
- _linphone_call_stats_set_sent_rtcp(stats, nullptr);
- _linphone_call_stats_set_ice_state(stats, LinphoneIceStateNotActivated);
-}
-void MediaSessionPrivate::notifyStatsUpdated (int streamIndex) {
- L_Q();
- LinphoneCallStats *stats = nullptr;
- if (streamIndex == mainAudioStreamIndex)
- stats = audioStats;
- else if (streamIndex == mainVideoStreamIndex)
- stats = videoStats;
- else if (streamIndex == mainTextStreamIndex)
- stats = textStats;
- else
- return;
- if (_linphone_call_stats_get_updated(stats)) {
- switch (_linphone_call_stats_get_updated(stats)) {
- case LINPHONE_CALL_STATS_RECEIVED_RTCP_UPDATE:
- case LINPHONE_CALL_STATS_SENT_RTCP_UPDATE:
- if (listener) {
- listener->onRtcpUpdateForReporting(q->getSharedFromThis(),
- (streamIndex == mainAudioStreamIndex)
- ? SalAudio
- : (streamIndex == mainVideoStreamIndex) ? SalVideo : SalText
- );
- }
- break;
- default:
- break;
- }
- if (listener)
- listener->onStatsUpdated(q->getSharedFromThis(), stats);
- _linphone_call_stats_set_updated(stats, 0);
- }
-}
// -----------------------------------------------------------------------------
-OrtpEvQueue * MediaSessionPrivate::getEventQueue (int streamIndex) const {
- if (streamIndex == mainAudioStreamIndex)
- return audioStreamEvQueue;
- if (streamIndex == mainVideoStreamIndex)
- return videoStreamEvQueue;
- if (streamIndex == mainTextStreamIndex)
- return textStreamEvQueue;
- lError() << "getEventQueue(): no stream index " << streamIndex;
- return nullptr;
-}
unsigned int MediaSessionPrivate::getAudioStartCount () const {
- return audioStartCount;
+ Stream *s = getStreamsGroup().lookupMainStream(SalAudio);
+ return s ? (unsigned int)s->getStartCount() : 0;
}
unsigned int MediaSessionPrivate::getVideoStartCount () const {
- return videoStartCount;
+ Stream *s = getStreamsGroup().lookupMainStream(SalVideo);
+ return s ? (unsigned int)s->getStartCount() : 0;
}
unsigned int MediaSessionPrivate::getTextStartCount () const {
- return textStartCount;
-}
-
-MediaStream *MediaSessionPrivate::getMediaStream (int streamIndex) const {
- if (streamIndex == mainAudioStreamIndex)
- return audioStream ? &audioStream->ms : nullptr;
- if (streamIndex == mainVideoStreamIndex)
- return videoStream ? &videoStream->ms : nullptr;
- if (streamIndex == mainTextStreamIndex)
- return textStream ? &textStream->ms : nullptr;
- lError() << "getMediaStream(): no stream index " << streamIndex;
- return nullptr;
+ Stream *s = getStreamsGroup().lookupMainStream(SalText);
+ return s ? (unsigned int)s->getStartCount() : 0;
}
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::fillMulticastMediaAddresses () {
- L_Q();
- if (getParams()->audioMulticastEnabled())
- mediaPorts[mainAudioStreamIndex].multicastIp = linphone_core_get_audio_multicast_addr(q->getCore()->getCCore());
- else
- mediaPorts[mainAudioStreamIndex].multicastIp.clear();
- if (getParams()->videoMulticastEnabled())
- mediaPorts[mainVideoStreamIndex].multicastIp = linphone_core_get_video_multicast_addr(q->getCore()->getCCore());
- else
- mediaPorts[mainVideoStreamIndex].multicastIp.clear();
-}
+// -----------------------------------------------------------------------------
-int MediaSessionPrivate::selectFixedPort (int streamIndex, pair<int, int> portRange) {
+void MediaSessionPrivate::discoverMtu (const Address &remoteAddr) {
L_Q();
- for (int triedPort = portRange.first; triedPort < (portRange.first + 100); triedPort += 2) {
- bool alreadyUsed = false;
- for (const bctbx_list_t *elem = linphone_core_get_calls(q->getCore()->getCCore()); elem != nullptr; elem = bctbx_list_next(elem)) {
- LinphoneCall *lcall = reinterpret_cast<LinphoneCall *>(bctbx_list_get_data(elem));
- shared_ptr<MediaSession> session = static_pointer_cast<MediaSession>(L_GET_CPP_PTR_FROM_C_OBJECT(lcall)->getPrivate()->getActiveSession());
- int existingPort = session->getPrivate()->mediaPorts[streamIndex].rtpPort;
- if (existingPort == triedPort) {
- alreadyUsed = true;
- break;
- }
+ if (q->getCore()->getCCore()->net_conf.mtu == 0) {
+ /* Attempt to discover mtu */
+ int mtu = ms_discover_mtu(remoteAddr.getDomain().c_str());
+ if (mtu > 0) {
+ ms_factory_set_mtu(q->getCore()->getCCore()->factory, mtu);
+ lInfo() << "Discovered mtu is " << mtu << ", RTP payload max size is " << ms_factory_get_payload_max_size(q->getCore()->getCCore()->factory);
}
- if (!alreadyUsed)
- return triedPort;
}
-
- lError() << "Could not find any free port !";
- return -1;
}
-int MediaSessionPrivate::selectRandomPort (int streamIndex, pair<int, int> portRange) {
- L_Q();
- unsigned int rangeSize = static_cast<unsigned int>(portRange.second - portRange.first);
-
- for (int nbTries = 0; nbTries < 100; nbTries++) {
- bool alreadyUsed = false;
- unsigned int randomInRangeSize = (bctbx_random() % rangeSize) & (unsigned int)~0x1; /* Select an even number */
- int triedPort = ((int)randomInRangeSize) + portRange.first;
- /*If portRange.first is even, the triedPort will be even too. The one who configures a port range that starts with an odd number will
- * get odd RTP port numbers.*/
-
- for (const bctbx_list_t *elem = linphone_core_get_calls(q->getCore()->getCCore()); elem != nullptr; elem = bctbx_list_next(elem)) {
- LinphoneCall *lcall = reinterpret_cast<LinphoneCall *>(bctbx_list_get_data(elem));
- shared_ptr<MediaSession> session = static_pointer_cast<MediaSession>(L_GET_CPP_PTR_FROM_C_OBJECT(lcall)->getPrivate()->getActiveSession());
- int existingPort = session->getPrivate()->mediaPorts[streamIndex].rtpPort;
- if (existingPort == triedPort) {
- alreadyUsed = true;
- break;
- }
- }
- lInfo() << "Port " << triedPort << " randomly taken from range [ " << portRange.first << " , " << portRange.second << "]";
- if (!alreadyUsed)
- return triedPort;
- }
-
- lError() << "Could not find any free port!";
- return -1;
-}
-
-void MediaSessionPrivate::setPortConfig(int streamIndex, pair<int, int> portRange) {
- if ((portRange.first <= 0) && (portRange.second <= 0)) {
- setRandomPortConfig(streamIndex);
- } else {
- if (portRange.first == portRange.second) {
- /* Fixed port */
- int port = selectFixedPort(streamIndex, portRange);
- if (port == -1) {
- setRandomPortConfig(streamIndex);
- return;
- }
- mediaPorts[streamIndex].rtpPort = port;
- } else {
- /* Select random port in the specified range */
- mediaPorts[streamIndex].rtpPort = selectRandomPort(streamIndex, portRange);
- }
- mediaPorts[streamIndex].rtcpPort = mediaPorts[streamIndex].rtpPort + 1;
- }
-}
-
-void MediaSessionPrivate::setPortConfigFromRtpSession (int streamIndex, RtpSession *session) {
- mediaPorts[streamIndex].rtpPort = rtp_session_get_local_port(session);
- mediaPorts[streamIndex].rtcpPort = rtp_session_get_local_rtcp_port(session);
-}
-
-void MediaSessionPrivate::setRandomPortConfig (int streamIndex) {
- mediaPorts[streamIndex].rtpPort = -1;
- mediaPorts[streamIndex].rtcpPort = -1;
-}
-
-// -----------------------------------------------------------------------------
-
-void MediaSessionPrivate::discoverMtu (const Address &remoteAddr) {
- L_Q();
- if (q->getCore()->getCCore()->net_conf.mtu == 0) {
- /* Attempt to discover mtu */
- int mtu = ms_discover_mtu(remoteAddr.getDomain().c_str());
- if (mtu > 0) {
- ms_factory_set_mtu(q->getCore()->getCCore()->factory, mtu);
- lInfo() << "Discovered mtu is " << mtu << ", RTP payload max size is " << ms_factory_get_payload_max_size(q->getCore()->getCCore()->factory);
- }
- }
-}
-
-string MediaSessionPrivate::getBindIpForStream (int streamIndex) {
- L_Q();
- string bindIp = lp_config_get_string(linphone_core_get_config(q->getCore()->getCCore()), "rtp", "bind_address", "");
- PortConfig *pc = &mediaPorts[streamIndex];
- if (!pc->multicastIp.empty()){
- if (direction == LinphoneCallOutgoing) {
- /* As multicast sender, we must decide a local interface to use to send multicast, and bind to it */
- char multicastBindIp[LINPHONE_IPADDR_SIZE];
- memset(multicastBindIp, 0, sizeof(multicastBindIp));
- linphone_core_get_local_ip_for((pc->multicastIp.find_first_of(':') == string::npos) ? AF_INET : AF_INET6, nullptr, multicastBindIp);
- bindIp = pc->multicastBindIp = multicastBindIp;
- } else {
- /* Otherwise we shall use an address family of the same family of the multicast address, because
- * dual stack socket and multicast don't work well on Mac OS (linux is OK, as usual). */
- bindIp = (pc->multicastIp.find_first_of(':') == string::npos) ? "0.0.0.0" : "::0";
- }
- }else if (bindIp.empty()){
- /*If ipv6 is not enabled, for listening to 0.0.0.0. The default behavior of mediastreamer when no IP is passed is to try ::0, and in
- * case of failure try 0.0.0.0 . But we don't want this if IPv6 is explicitely disabled.*/
- if (!linphone_core_ipv6_enabled(q->getCore()->getCCore())){
- bindIp = "0.0.0.0";
- }
- }
- return bindIp;
-}
/**
* Fill the local ip that routes to the internet according to the destination, or guess it by other special means.
@@ -1377,17 +904,14 @@ void MediaSessionPrivate::getLocalIp (const Address &remoteAddr) {
}
}
-string MediaSessionPrivate::getPublicIpForStream (int streamIndex) {
- if (!mediaPorts[streamIndex].multicastIp.empty())
- return mediaPorts[streamIndex].multicastIp;
- return mediaLocalIp;
-}
-
void MediaSessionPrivate::runStunTestsIfNeeded () {
L_Q();
if (linphone_nat_policy_stun_enabled(natPolicy) && !(linphone_nat_policy_ice_enabled(natPolicy) || linphone_nat_policy_turn_enabled(natPolicy))) {
stunClient = makeUnique<StunClient>(q->getCore());
- int ret = stunClient->run(mediaPorts[mainAudioStreamIndex].rtpPort, mediaPorts[mainVideoStreamIndex].rtpPort, mediaPorts[mainTextStreamIndex].rtpPort);
+ int audioPort = mainAudioStreamIndex ? getStreamsGroup().getStream(mainAudioStreamIndex)->getPortConfig().rtpPort : 0;
+ int videoPort = mainVideoStreamIndex ? getStreamsGroup().getStream(mainVideoStreamIndex)->getPortConfig().rtpPort : 0;
+ int textPort = mainTextStreamIndex ? getStreamsGroup().getStream(mainTextStreamIndex)->getPortConfig().rtpPort : 0;
+ int ret = stunClient->run(audioPort, videoPort, textPort);
if (ret >= 0)
pingTime = ret;
}
@@ -1459,7 +983,7 @@ void MediaSessionPrivate::selectOutgoingIpVersion () {
void MediaSessionPrivate::forceStreamsDirAccordingToState (SalMediaDescription *md) {
L_Q();
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
+ for (int i = 0; i < md->nb_streams; i++) {
SalStreamDescription *sd = &md->streams[i];
switch (state) {
case CallSession::State::Pausing:
@@ -1510,25 +1034,27 @@ bool MediaSessionPrivate::generateB64CryptoKey (size_t keyLength, char *keyOut,
return true;
}
-void MediaSessionPrivate::makeLocalMediaDescription () {
+void MediaSessionPrivate::addStreamToBundle(SalMediaDescription *md, SalStreamDescription *sd, const char *mid){
+ SalStreamBundle *bundle;
+ if (md->bundles == nullptr){
+ bundle = sal_media_description_add_new_bundle(md);
+ }else{
+ bundle = (SalStreamBundle*) md->bundles->data;
+ }
+ sal_stream_bundle_add_stream(bundle, sd, mid);
+ sd->mid_rtp_ext_header_id = rtpExtHeaderMidNumber;
+ /* rtcp-mux must be enabled when bundle mode is proposed.*/
+ sd->rtcp_mux = TRUE;
+}
+
+void MediaSessionPrivate::makeLocalMediaDescription(bool localIsOfferer) {
L_Q();
- int maxIndex = 0;
bool rtcpMux = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "rtp", "rtcp_mux", 0);
SalMediaDescription *md = sal_media_description_new();
SalMediaDescription *oldMd = localDesc;
-
- /* Multicast is only set in case of outgoing call */
- if (direction == LinphoneCallOutgoing) {
- if (getParams()->audioMulticastEnabled()) {
- md->streams[mainAudioStreamIndex].ttl = linphone_core_get_audio_multicast_ttl(q->getCore()->getCCore());
- md->streams[mainAudioStreamIndex].multicast_role = SalMulticastSender;
- }
- if (getParams()->videoMulticastEnabled()) {
- md->streams[mainVideoStreamIndex].ttl = linphone_core_get_video_multicast_ttl(q->getCore()->getCCore());
- md->streams[mainVideoStreamIndex].multicast_role = SalMulticastSender;
- }
- }
-
+
+ assignStreamsIndexes();
+
getParams()->getPrivate()->adaptToNetwork(q->getCore()->getCCore(), pingTime);
string subject = q->getParams()->getSessionName();
@@ -1539,6 +1065,9 @@ void MediaSessionPrivate::makeLocalMediaDescription () {
md->session_id = (oldMd ? oldMd->session_id : (bctbx_random() & 0xfff));
md->session_ver = (oldMd ? (oldMd->session_ver + 1) : (bctbx_random() & 0xfff));
md->nb_streams = (biggestDesc ? biggestDesc->nb_streams : 1);
+
+ md->accept_bundles = getParams()->rtpBundleEnabled() ||
+ linphone_config_get_bool(linphone_core_get_config(q->getCore()->getCCore()), "rtp", "accept_bundle", TRUE);
/* Re-check local ip address each time we make a new offer, because it may change in case of network reconnection */
{
@@ -1570,109 +1099,122 @@ void MediaSessionPrivate::makeLocalMediaDescription () {
SalCustomSdpAttribute *customSdpAttributes = getParams()->getPrivate()->getCustomSdpAttributes();
if (customSdpAttributes)
md->custom_sdp_attributes = sal_custom_sdp_attribute_clone(customSdpAttributes);
+
PayloadTypeHandler pth(q->getCore());
- bctbx_list_t *l = pth.makeCodecsList(SalAudio, getParams()->getAudioBandwidthLimit(), -1,
- oldMd ? oldMd->streams[mainAudioStreamIndex].already_assigned_payloads : nullptr);
- if (l && getParams()->audioEnabled()) {
- strncpy(md->streams[mainAudioStreamIndex].rtp_addr, getPublicIpForStream(mainAudioStreamIndex).c_str(), sizeof(md->streams[mainAudioStreamIndex].rtp_addr));
- strncpy(md->streams[mainAudioStreamIndex].rtcp_addr, getPublicIpForStream(mainAudioStreamIndex).c_str(), sizeof(md->streams[mainAudioStreamIndex].rtcp_addr));
- strncpy(md->streams[mainAudioStreamIndex].name, "Audio", sizeof(md->streams[mainAudioStreamIndex].name) - 1);
- md->streams[mainAudioStreamIndex].rtp_port = mediaPorts[mainAudioStreamIndex].rtpPort;
- md->streams[mainAudioStreamIndex].rtcp_port = mediaPorts[mainAudioStreamIndex].rtcpPort;
+ bctbx_list_t *l = NULL;
+ if (mainAudioStreamIndex != -1){
+ l = nullptr;
md->streams[mainAudioStreamIndex].proto = getParams()->getMediaProto();
md->streams[mainAudioStreamIndex].dir = getParams()->getPrivate()->getSalAudioDirection();
md->streams[mainAudioStreamIndex].type = SalAudio;
- md->streams[mainAudioStreamIndex].rtcp_mux = rtcpMux;
- int downPtime = getParams()->getPrivate()->getDownPtime();
- if (downPtime)
- md->streams[mainAudioStreamIndex].ptime = downPtime;
- else
- md->streams[mainAudioStreamIndex].ptime = linphone_core_get_download_ptime(q->getCore()->getCCore());
- md->streams[mainAudioStreamIndex].max_rate = pth.getMaxCodecSampleRate(l);
- md->streams[mainAudioStreamIndex].payloads = l;
- if (audioStream && audioStream->ms.sessions.rtp_session) {
- md->streams[mainAudioStreamIndex].rtp_ssrc = rtp_session_get_send_ssrc(audioStream->ms.sessions.rtp_session);
+ if (getParams()->audioEnabled() && (l = pth.makeCodecsList(SalAudio, getParams()->getAudioBandwidthLimit(), -1,
+ oldMd ? oldMd->streams[mainAudioStreamIndex].already_assigned_payloads : nullptr))) {
+ strncpy(md->streams[mainAudioStreamIndex].name, "Audio", sizeof(md->streams[mainAudioStreamIndex].name) - 1);
+ md->streams[mainAudioStreamIndex].rtcp_mux = rtcpMux;
+ md->streams[mainAudioStreamIndex].rtp_port = SAL_STREAM_DESCRIPTION_PORT_TO_BE_DETERMINED;
+ int downPtime = getParams()->getPrivate()->getDownPtime();
+ if (downPtime)
+ md->streams[mainAudioStreamIndex].ptime = downPtime;
+ else
+ md->streams[mainAudioStreamIndex].ptime = linphone_core_get_download_ptime(q->getCore()->getCCore());
+ md->streams[mainAudioStreamIndex].max_rate = pth.getMaxCodecSampleRate(l);
+ md->streams[mainAudioStreamIndex].payloads = l;
strncpy(md->streams[mainAudioStreamIndex].rtcp_cname, getMe()->getAddress().asString().c_str(), sizeof(md->streams[mainAudioStreamIndex].rtcp_cname));
- }
- else
- lWarning() << "Cannot get audio local ssrc for CallSession [" << q << "]";
- if (mainAudioStreamIndex > maxIndex)
- maxIndex = mainAudioStreamIndex;
- } else {
- lInfo() << "Don't put audio stream on local offer for CallSession [" << q << "]";
- md->streams[mainAudioStreamIndex].dir = SalStreamInactive;
- if(l)
- l = bctbx_list_free_with_data(l, (bctbx_list_free_func)payload_type_destroy);
- }
- SalCustomSdpAttribute *sdpMediaAttributes = getParams()->getPrivate()->getCustomSdpMediaAttributes(LinphoneStreamTypeAudio);
- if (sdpMediaAttributes)
- md->streams[mainAudioStreamIndex].custom_sdp_attributes = sal_custom_sdp_attribute_clone(sdpMediaAttributes);
-
- md->streams[mainVideoStreamIndex].proto = getParams()->getMediaProto();
- md->streams[mainVideoStreamIndex].dir = getParams()->getPrivate()->getSalVideoDirection();
- md->streams[mainVideoStreamIndex].type = SalVideo;
- md->streams[mainVideoStreamIndex].rtcp_mux = rtcpMux;
- strncpy(md->streams[mainVideoStreamIndex].name, "Video", sizeof(md->streams[mainVideoStreamIndex].name) - 1);
-
- l = pth.makeCodecsList(SalVideo, 0, -1,
- oldMd ? oldMd->streams[mainVideoStreamIndex].already_assigned_payloads : nullptr);
- if (l && getParams()->videoEnabled()){
- strncpy(md->streams[mainVideoStreamIndex].rtp_addr, getPublicIpForStream(mainVideoStreamIndex).c_str(), sizeof(md->streams[mainVideoStreamIndex].rtp_addr));
- strncpy(md->streams[mainVideoStreamIndex].rtcp_addr, getPublicIpForStream(mainVideoStreamIndex).c_str(), sizeof(md->streams[mainVideoStreamIndex].rtcp_addr));
- md->streams[mainVideoStreamIndex].rtp_port = mediaPorts[mainVideoStreamIndex].rtpPort;
- md->streams[mainVideoStreamIndex].rtcp_port = mediaPorts[mainVideoStreamIndex].rtcpPort;
- md->streams[mainVideoStreamIndex].payloads = l;
- if (videoStream && videoStream->ms.sessions.rtp_session) {
- md->streams[mainVideoStreamIndex].rtp_ssrc = rtp_session_get_send_ssrc(videoStream->ms.sessions.rtp_session);
+ if (getParams()->rtpBundleEnabled()) addStreamToBundle(md, &md->streams[mainAudioStreamIndex], "as");
+
+ if (getParams()->audioMulticastEnabled()) {
+ md->streams[mainAudioStreamIndex].ttl = linphone_core_get_audio_multicast_ttl(q->getCore()->getCCore());
+ md->streams[mainAudioStreamIndex].multicast_role = (direction == LinphoneCallOutgoing) ? SalMulticastSender : SalMulticastReceiver;
+ }
+
+ } else {
+ lInfo() << "Don't put audio stream on local offer for CallSession [" << q << "]";
+ md->streams[mainAudioStreamIndex].dir = SalStreamInactive;
+ if(l)
+ l = bctbx_list_free_with_data(l, (bctbx_list_free_func)payload_type_destroy);
+ }
+ customSdpAttributes = getParams()->getPrivate()->getCustomSdpMediaAttributes(LinphoneStreamTypeAudio);
+ if (customSdpAttributes)
+ md->streams[mainAudioStreamIndex].custom_sdp_attributes = sal_custom_sdp_attribute_clone(customSdpAttributes);
+ }
+ if (mainVideoStreamIndex != -1){
+ l = nullptr;
+ md->streams[mainVideoStreamIndex].proto = getParams()->getMediaProto();
+ md->streams[mainVideoStreamIndex].dir = getParams()->getPrivate()->getSalVideoDirection();
+ md->streams[mainVideoStreamIndex].type = SalVideo;
+
+ if (getParams()->videoEnabled() && (l = pth.makeCodecsList(SalVideo, 0, -1,
+ oldMd ? oldMd->streams[mainVideoStreamIndex].already_assigned_payloads : nullptr)) ){
+ md->streams[mainVideoStreamIndex].rtcp_mux = rtcpMux;
+ md->streams[mainVideoStreamIndex].rtp_port = SAL_STREAM_DESCRIPTION_PORT_TO_BE_DETERMINED;
+ strncpy(md->streams[mainVideoStreamIndex].name, "Video", sizeof(md->streams[mainVideoStreamIndex].name) - 1);
+ md->streams[mainVideoStreamIndex].payloads = l;
strncpy(md->streams[mainVideoStreamIndex].rtcp_cname, getMe()->getAddress().asString().c_str(), sizeof(md->streams[mainVideoStreamIndex].rtcp_cname));
- } else
- lWarning() << "Cannot get video local ssrc for CallSession [" << q << "]";
- if (mainVideoStreamIndex > maxIndex)
- maxIndex = mainVideoStreamIndex;
- } else {
- lInfo() << "Don't put video stream on local offer for CallSession [" << q << "]";
- md->streams[mainVideoStreamIndex].dir = SalStreamInactive;
- if(l)
- l = bctbx_list_free_with_data(l, (bctbx_list_free_func)payload_type_destroy);
- }
- sdpMediaAttributes = getParams()->getPrivate()->getCustomSdpMediaAttributes(LinphoneStreamTypeVideo);
- if (sdpMediaAttributes)
- md->streams[mainVideoStreamIndex].custom_sdp_attributes = sal_custom_sdp_attribute_clone(sdpMediaAttributes);
-
- md->streams[mainTextStreamIndex].proto = getParams()->getMediaProto();
- md->streams[mainTextStreamIndex].dir = SalStreamSendRecv;
- md->streams[mainTextStreamIndex].type = SalText;
- md->streams[mainTextStreamIndex].rtcp_mux = rtcpMux;
- strncpy(md->streams[mainTextStreamIndex].name, "Text", sizeof(md->streams[mainTextStreamIndex].name) - 1);
- if (getParams()->realtimeTextEnabled()) {
- strncpy(md->streams[mainTextStreamIndex].rtp_addr, getPublicIpForStream(mainTextStreamIndex).c_str(), sizeof(md->streams[mainTextStreamIndex].rtp_addr));
- strncpy(md->streams[mainTextStreamIndex].rtcp_addr, getPublicIpForStream(mainTextStreamIndex).c_str(), sizeof(md->streams[mainTextStreamIndex].rtcp_addr));
-
- md->streams[mainTextStreamIndex].rtp_port = mediaPorts[mainTextStreamIndex].rtpPort;
- md->streams[mainTextStreamIndex].rtcp_port = mediaPorts[mainTextStreamIndex].rtcpPort;
-
- l = pth.makeCodecsList(SalText, 0, -1,
- oldMd ? oldMd->streams[mainTextStreamIndex].already_assigned_payloads : nullptr);
- md->streams[mainTextStreamIndex].payloads = l;
- if (textStream && textStream->ms.sessions.rtp_session) {
- md->streams[mainTextStreamIndex].rtp_ssrc = rtp_session_get_send_ssrc(textStream->ms.sessions.rtp_session);
+ if (getParams()->rtpBundleEnabled()) addStreamToBundle(md, &md->streams[mainVideoStreamIndex], "vs");
+
+ if (getParams()->videoMulticastEnabled()) {
+ md->streams[mainVideoStreamIndex].ttl = linphone_core_get_video_multicast_ttl(q->getCore()->getCCore());
+ md->streams[mainVideoStreamIndex].multicast_role = (direction == LinphoneCallOutgoing) ? SalMulticastSender : SalMulticastReceiver;
+ }
+ } else {
+ lInfo() << "Don't put video stream on local offer for CallSession [" << q << "]";
+ md->streams[mainVideoStreamIndex].dir = SalStreamInactive;
+ if(l)
+ l = bctbx_list_free_with_data(l, (bctbx_list_free_func)payload_type_destroy);
+ }
+ customSdpAttributes = getParams()->getPrivate()->getCustomSdpMediaAttributes(LinphoneStreamTypeVideo);
+ if (customSdpAttributes)
+ md->streams[mainVideoStreamIndex].custom_sdp_attributes = sal_custom_sdp_attribute_clone(customSdpAttributes);
+ }
+
+ if (mainTextStreamIndex != -1){
+ l = nullptr;
+ md->streams[mainTextStreamIndex].proto = getParams()->getMediaProto();
+ md->streams[mainTextStreamIndex].dir = SalStreamSendRecv;
+ md->streams[mainTextStreamIndex].type = SalText;
+ if (getParams()->realtimeTextEnabled() && (l = pth.makeCodecsList(SalText, 0, -1,
+ oldMd ? oldMd->streams[mainTextStreamIndex].already_assigned_payloads : nullptr)) ) {
+ md->streams[mainTextStreamIndex].rtcp_mux = rtcpMux;
+ md->streams[mainTextStreamIndex].rtp_port = getParams()->realtimeTextEnabled() ? SAL_STREAM_DESCRIPTION_PORT_TO_BE_DETERMINED : 0;
+ strncpy(md->streams[mainTextStreamIndex].name, "Text", sizeof(md->streams[mainTextStreamIndex].name) - 1);
+ md->streams[mainTextStreamIndex].payloads = l;
strncpy(md->streams[mainTextStreamIndex].rtcp_cname, getMe()->getAddress().asString().c_str(), sizeof(md->streams[mainTextStreamIndex].rtcp_cname));
- } else
- lWarning() << "Cannot get text local ssrc for CallSession [" << q << "]";
- if (mainTextStreamIndex > maxIndex)
- maxIndex = mainTextStreamIndex;
- } else {
- lInfo() << "Don't put text stream on local offer for CallSession [" << q << "]";
- md->streams[mainTextStreamIndex].dir = SalStreamInactive;
+ if (getParams()->rtpBundleEnabled()) addStreamToBundle(md, &md->streams[mainTextStreamIndex], "ts");
+ } else {
+ lInfo() << "Don't put text stream on local offer for CallSession [" << q << "]";
+ md->streams[mainTextStreamIndex].dir = SalStreamInactive;
+ if(l)
+ l = bctbx_list_free_with_data(l, (bctbx_list_free_func)payload_type_destroy);
+ }
+ customSdpAttributes = getParams()->getPrivate()->getCustomSdpMediaAttributes(LinphoneStreamTypeText);
+ if (customSdpAttributes)
+ md->streams[mainTextStreamIndex].custom_sdp_attributes = sal_custom_sdp_attribute_clone(customSdpAttributes);
}
- sdpMediaAttributes = getParams()->getPrivate()->getCustomSdpMediaAttributes(LinphoneStreamTypeText);
- if (sdpMediaAttributes)
- md->streams[mainTextStreamIndex].custom_sdp_attributes = sal_custom_sdp_attribute_clone(sdpMediaAttributes);
- md->nb_streams = MAX(md->nb_streams, maxIndex + 1);
+ md->nb_streams = freeStreamIndex;
+ setupEncryptionKeys(md);
+ setupImEncryptionEngineParameters(md);
+ setupRtcpFb(md);
+ setupRtcpXr(md);
+ if (stunClient)
+ stunClient->updateMediaDescription(md);
+ localDesc = md;
+
+ OfferAnswerContext ctx;
+ ctx.localMediaDescription = localDesc;
+ ctx.remoteMediaDescription = localIsOfferer ? nullptr : ( op ? op->getRemoteMediaDescription() : nullptr);
+ ctx.localIsOfferer = localIsOfferer;
+ /* Now instanciate the streams according to the media description. */
+ getStreamsGroup().createStreams(ctx);
+ if (mainAudioStreamIndex != -1) getStreamsGroup().setStreamMain((size_t)mainAudioStreamIndex);
+ if (mainVideoStreamIndex != -1) getStreamsGroup().setStreamMain((size_t)mainVideoStreamIndex);
+ if (mainTextStreamIndex != -1) getStreamsGroup().setStreamMain((size_t)mainTextStreamIndex);
+ /* Get the transport addresses filled in to the media description. */
+ getStreamsGroup().fillLocalMediaDescription(ctx);
+
/* Deactivate unused streams */
for (int i = md->nb_streams; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
if (md->streams[i].rtp_port == 0) {
@@ -1683,15 +1225,7 @@ void MediaSessionPrivate::makeLocalMediaDescription () {
}
}
}
- setupEncryptionKeys(md);
- setupDtlsKeys(md);
- setupZrtpHash(md);
- setupImEncryptionEngineParameters(md);
- setupRtcpFb(md);
- setupRtcpXr(md);
- if (stunClient)
- stunClient->updateMediaDescription(md);
- localDesc = md;
+
updateLocalMediaDescriptionFromIce();
if (oldMd) {
transferAlreadyAssignedPayloadTypes(oldMd, md);
@@ -1707,24 +1241,7 @@ void MediaSessionPrivate::makeLocalMediaDescription () {
}
}
forceStreamsDirAccordingToState(md);
-}
-
-void MediaSessionPrivate::setupDtlsKeys (SalMediaDescription *md) {
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i]))
- continue;
- /* If media encryption is set to DTLS check presence of fingerprint in the call which shall have been set at stream init
- * but it may have failed when retrieving certificate resulting in no fingerprint present and then DTLS not usable */
- if (sal_stream_description_has_dtls(&md->streams[i])) {
- /* Get the self fingerprint from call (it's computed at stream init) */
- strncpy(md->streams[i].dtls_fingerprint, dtlsCertificateFingerprint.c_str(), sizeof(md->streams[i].dtls_fingerprint));
- /* If we are offering, SDP will have actpass setup attribute when role is unset, if we are responding the result mediadescription will be set to SalDtlsRoleIsClient */
- md->streams[i].dtls_role = SalDtlsRoleUnset;
- } else {
- md->streams[i].dtls_fingerprint[0] = '\0';
- md->streams[i].dtls_role = SalDtlsRoleInvalid;
- }
- }
+ if (op) op->setLocalMediaDescription(localDesc);
}
int MediaSessionPrivate::setupEncryptionKey (SalSrtpCryptoAlgo *crypto, MSCryptoSuite suite, unsigned int tag) {
@@ -1756,9 +1273,7 @@ int MediaSessionPrivate::setupEncryptionKey (SalSrtpCryptoAlgo *crypto, MSCrypto
void MediaSessionPrivate::setupRtcpFb (SalMediaDescription *md) {
L_Q();
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i]))
- continue;
+ for (int i = 0; i < md->nb_streams; i++) {
md->streams[i].rtcp_fb.generic_nack_enabled = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "rtp", "rtcp_fb_generic_nack_enabled", 0);
md->streams[i].rtcp_fb.tmmbr_enabled = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "rtp", "rtcp_fb_tmmbr_enabled", 1);
md->streams[i].implicit_rtcp_fb = getParams()->getPrivate()->implicitRtcpFbEnabled();
@@ -1796,29 +1311,11 @@ void MediaSessionPrivate::setupRtcpXr (SalMediaDescription *md) {
md->rtcp_xr.stat_summary_flags = OrtpRtcpXrStatSummaryLoss | OrtpRtcpXrStatSummaryDup | OrtpRtcpXrStatSummaryJitt | OrtpRtcpXrStatSummaryTTL;
md->rtcp_xr.voip_metrics_enabled = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "rtp", "rtcp_xr_voip_metrics_enabled", 1);
}
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i]))
- continue;
+ for (int i = 0; i < md->nb_streams; i++) {
memcpy(&md->streams[i].rtcp_xr, &md->rtcp_xr, sizeof(md->streams[i].rtcp_xr));
}
}
-void MediaSessionPrivate::setupZrtpHash (SalMediaDescription *md) {
- L_Q();
- if (linphone_core_media_encryption_supported(q->getCore()->getCCore(), LinphoneMediaEncryptionZRTP)) {
- /* Set the hello hash for all streams */
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i]))
- continue;
- if (sessions[i].zrtp_context) {
- ms_zrtp_getHelloHash(sessions[i].zrtp_context, md->streams[i].zrtphash, 128);
- /* Turn on the flag to use it if ZRTP is set */
- md->streams[i].haveZrtpHash = (getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP);
- } else
- md->streams[i].haveZrtpHash = 0;
- }
- }
-}
void MediaSessionPrivate::setupImEncryptionEngineParameters (SalMediaDescription *md) {
L_Q();
@@ -1839,11 +1336,9 @@ void MediaSessionPrivate::setupEncryptionKeys (SalMediaDescription *md) {
L_Q();
SalMediaDescription *oldMd = localDesc;
bool keepSrtpKeys = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sip", "keep_srtp_keys", 1);
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i]))
- continue;
+ for (int i = 0; i < md->nb_streams; i++) {
if (sal_stream_description_has_srtp(&md->streams[i])) {
- if (keepSrtpKeys && oldMd && sal_stream_description_active(&oldMd->streams[i]) && sal_stream_description_has_srtp(&oldMd->streams[i])) {
+ if (keepSrtpKeys && oldMd && sal_stream_description_enabled(&oldMd->streams[i]) && sal_stream_description_has_srtp(&oldMd->streams[i])) {
lInfo() << "Keeping same crypto keys";
for (int j = 0; j < SAL_CRYPTO_ALGO_MAX; j++) {
memcpy(&md->streams[i].crypto[j], &oldMd->streams[i].crypto[j], sizeof(SalSrtpCryptoAlgo));
@@ -1866,2141 +1361,216 @@ void MediaSessionPrivate::transferAlreadyAssignedPayloadTypes (SalMediaDescripti
}
void MediaSessionPrivate::updateLocalMediaDescriptionFromIce () {
- iceAgent->updateLocalMediaDescriptionFromIce(localDesc);
- iceAgent->updateIceStateInCallStats();
+ OfferAnswerContext ctx;
+ ctx.localMediaDescription = localDesc;
+ ctx.remoteMediaDescription = op ? op->getRemoteMediaDescription() : nullptr;
+ getStreamsGroup().fillLocalMediaDescription(ctx);
+ if (op) op->setLocalMediaDescription(localDesc);
}
-// -----------------------------------------------------------------------------
-SalMulticastRole MediaSessionPrivate::getMulticastRole (SalStreamType type) {
+void MediaSessionPrivate::performMutualAuthentication(){
L_Q();
- SalMulticastRole multicastRole = SalMulticastInactive;
- if (op) {
- SalStreamDescription *streamDesc = nullptr;
- SalMediaDescription *remoteDesc = op->getRemoteMediaDescription();
- if (!localDesc && !remoteDesc && (direction == LinphoneCallOutgoing)) {
- /* Well using call dir */
- if (((type == SalAudio) && getParams()->audioMulticastEnabled())
- || ((type == SalVideo) && getParams()->videoMulticastEnabled()))
- multicastRole = SalMulticastSender;
- } else if (localDesc && (!remoteDesc || op->isOfferer())) {
- streamDesc = sal_media_description_find_best_stream(localDesc, type);
- } else if (!op->isOfferer() && remoteDesc) {
- streamDesc = sal_media_description_find_best_stream(remoteDesc, type);
- }
-
- if (streamDesc)
- multicastRole = streamDesc->multicast_role;
+
+ // Perform mutual authentication if instant messaging encryption is enabled
+ auto encryptionEngine = q->getCore()->getEncryptionEngine();
+ // Is call direction really relevant ? might be linked to offerer/answerer rather than call direction ?
+ Stream *stream = mainAudioStreamIndex != -1 ? getStreamsGroup().getStream(mainAudioStreamIndex) : nullptr;
+ MS2AudioStream *ms2a = dynamic_cast<MS2AudioStream*>(stream);
+ if (encryptionEngine && ms2a && ms2a->getZrtpContext()) {
+ encryptionEngine->mutualAuthentication(
+ ms2a->getZrtpContext(),
+ op->getLocalMediaDescription(),
+ op->getRemoteMediaDescription(),
+ q->getDirection()
+ );
}
- lInfo() << "CallSession [" << q << "], stream type [" << sal_stream_type_to_string(type) << "], multicast role is ["
- << sal_multicast_role_to_string(multicastRole) << "]";
- return multicastRole;
}
-void MediaSessionPrivate::joinMulticastGroup (int streamIndex, MediaStream *ms) {
- L_Q();
- if (!mediaPorts[streamIndex].multicastIp.empty())
- media_stream_join_multicast_group(ms, mediaPorts[streamIndex].multicastIp.c_str());
- else
- lError() << "Cannot join multicast group if multicast ip is not set for call [" << q << "]";
-}
-
-// -----------------------------------------------------------------------------
-void MediaSessionPrivate::setDtlsFingerprint (MSMediaStreamSessions *sessions, const SalStreamDescription *sd, const SalStreamDescription *remote) {
- if (sal_stream_description_has_dtls(sd)) {
- if (sd->dtls_role == SalDtlsRoleInvalid)
- lWarning() << "Unable to start DTLS engine on stream session [" << sessions << "], Dtls role in resulting media description is invalid";
- else { /* If DTLS is available at both end points */
- /* Give the peer certificate fingerprint to dtls context */
- ms_dtls_srtp_set_peer_fingerprint(sessions->dtls_context, remote->dtls_fingerprint);
- }
+void MediaSessionPrivate::startDtlsOnAllStreams () {
+ OfferAnswerContext params;
+ params.localMediaDescription = localDesc;
+ params.remoteMediaDescription = op->getRemoteMediaDescription();
+ params.resultMediaDescription = resultDesc;
+ if (params.remoteMediaDescription && params.resultMediaDescription){
+ getStreamsGroup().startDtls(params);
}
}
-void MediaSessionPrivate::setDtlsFingerprintOnAudioStream () {
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- SalMediaDescription *result = op->getFinalMediaDescription();
-
- if (!remote || !result) {
- /* This can happen in some tricky cases (early-media without SDP in the 200). In that case, simply skip DTLS code */
- return;
- }
- if (audioStream && (media_stream_get_state(&audioStream->ms) == MSStreamStarted))
- setDtlsFingerprint(&audioStream->ms.sessions, sal_media_description_find_best_stream(result, SalAudio), sal_media_description_find_best_stream(remote, SalAudio));
+/*
+ * Frees the media resources of the call.
+ * This has to be done at the earliest, unlike signaling resources that sometimes need to be kept a bit more longer.
+ * It is called by setTerminated() (for termination of calls signaled to the application), or directly by the destructor of the session
+ * if it was never notified to the application.
+ */
+void MediaSessionPrivate::freeResources () {
+ getStreamsGroup().finish();
}
-void MediaSessionPrivate::setDtlsFingerprintOnVideoStream () {
-#if VIDEO_ENABLED
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- SalMediaDescription *result = op->getFinalMediaDescription();
-
- if (!remote || !result) {
- /* This can happen in some tricky cases (early-media without SDP in the 200). In that case, simply skip DTLS code */
- return;
+/*
+ * IceServiceListener implementation
+ */
+void MediaSessionPrivate::onGatheringFinished(IceService &service){
+ L_Q();
+ updateLocalMediaDescriptionFromIce();
+ switch (state) {
+ case CallSession::State::IncomingReceived:
+ case CallSession::State::IncomingEarlyMedia:
+ if (callAcceptanceDefered) startAccept();
+ break;
+ case CallSession::State::Updating:
+ startUpdate();
+ break;
+ case CallSession::State::UpdatedByRemote:
+ startAcceptUpdate(prevState, Utils::toString(prevState));
+ break;
+ case CallSession::State::OutgoingInit:
+ q->startInvite(nullptr, "");
+ break;
+ case CallSession::State::Idle:
+ deferIncomingNotification = false;
+ startIncomingNotification();
+ break;
+ default:
+ break;
}
-
- if (videoStream && (media_stream_get_state(&videoStream->ms) == MSStreamStarted))
- setDtlsFingerprint(&videoStream->ms.sessions, sal_media_description_find_best_stream(result, SalVideo), sal_media_description_find_best_stream(remote, SalVideo));
-#endif
}
-void MediaSessionPrivate::setDtlsFingerprintOnTextStream () {
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- SalMediaDescription *result = op->getFinalMediaDescription();
-
- if (!remote || !result) {
- /* This can happen in some tricky cases (early-media without SDP in the 200). In that case, simply skip DTLS code */
- return;
+void MediaSessionPrivate::onIceCompleted(IceService &service){
+ L_Q();
+ /* The ICE session has succeeded, so perform a call update */
+ if (!getStreamsGroup().getIceService().hasCompletedCheckList()) return;
+ if (getStreamsGroup().getIceService().isControlling() && getParams()->getPrivate()->getUpdateCallWhenIceCompleted()) {
+ if (state == CallSession::State::StreamsRunning){
+ MediaSessionParams newParams(*getParams());
+ newParams.getPrivate()->setInternalCallUpdate(true);
+ q->update(&newParams);
+ }else{
+ lWarning() << "Cannot send reINVITE for ICE during state " << state;
+ }
}
-
- if (textStream && (media_stream_get_state(&textStream->ms) == MSStreamStarted))
- setDtlsFingerprint(&textStream->ms.sessions, sal_media_description_find_best_stream(result, SalText), sal_media_description_find_best_stream(remote, SalText));
-}
-
-void MediaSessionPrivate::setDtlsFingerprintOnAllStreams () {
- setDtlsFingerprintOnAudioStream();
- setDtlsFingerprintOnVideoStream();
- setDtlsFingerprintOnTextStream();
+ startDtlsOnAllStreams();
}
-void MediaSessionPrivate::setupDtlsParams (MediaStream *ms) {
- L_Q();
- if (getParams()->getMediaEncryption() == LinphoneMediaEncryptionDTLS) {
- MSDtlsSrtpParams dtlsParams = { 0 };
-
- /* TODO : search for a certificate with CNAME=sip uri(retrieved from variable me) or default : linphone-dtls-default-identity */
- /* This will parse the directory to find a matching fingerprint or generate it if not found */
- /* returned string must be freed */
- char *certificate = nullptr;
- char *key = nullptr;
- char *fingerprint = nullptr;
-
- sal_certificates_chain_parse_directory(&certificate, &key, &fingerprint,
- linphone_core_get_user_certificates_path(q->getCore()->getCCore()), "linphone-dtls-default-identity", SAL_CERTIFICATE_RAW_FORMAT_PEM, true, true);
- if (fingerprint) {
- dtlsCertificateFingerprint = fingerprint;
- ms_free(fingerprint);
- }
- if (key && certificate) {
- dtlsParams.pem_certificate = certificate;
- dtlsParams.pem_pkey = key;
- dtlsParams.role = MSDtlsSrtpRoleUnset; /* Default is unset, then check if we have a result SalMediaDescription */
- media_stream_enable_dtls(ms, &dtlsParams);
- ms_free(certificate);
- ms_free(key);
- } else {
- lError() << "Unable to retrieve or generate DTLS certificate and key - DTLS disabled";
- /* TODO : check if encryption forced, if yes, stop call */
+void MediaSessionPrivate::onLosingPairsCompleted(IceService &service){
+ if (state == CallSession::State::UpdatedByRemote) {
+ if (incomingIceReinvitePending){
+ lInfo() << "Finished adding losing pairs, ICE re-INVITE can be answered.";
+ startAcceptUpdate(prevState, Utils::toString(prevState));
+ incomingIceReinvitePending = false;
}
}
}
-void MediaSessionPrivate::setZrtpCryptoTypesParameters (MSZrtpParams *params) {
+void MediaSessionPrivate::onIceRestartNeeded(IceService & service){
L_Q();
- if (!params)
- return;
-
- const MSCryptoSuite *srtpSuites = linphone_core_get_srtp_crypto_suites(q->getCore()->getCCore());
- if (srtpSuites) {
- for(int i = 0; (srtpSuites[i] != MS_CRYPTO_SUITE_INVALID) && (i < SAL_CRYPTO_ALGO_MAX) && (i < MS_MAX_ZRTP_CRYPTO_TYPES); i++) {
- switch (srtpSuites[i]) {
- case MS_AES_128_SHA1_32:
- params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES1;
- params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS32;
- break;
- case MS_AES_128_NO_AUTH:
- params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES1;
- break;
- case MS_NO_CIPHER_SHA1_80:
- params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS80;
- break;
- case MS_AES_128_SHA1_80:
- params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES1;
- params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS80;
- break;
- case MS_AES_CM_256_SHA1_80:
- lWarning() << "Deprecated crypto suite MS_AES_CM_256_SHA1_80, use MS_AES_256_SHA1_80 instead";
- BCTBX_NO_BREAK;
- case MS_AES_256_SHA1_80:
- params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES3;
- params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS80;
- break;
- case MS_AES_256_SHA1_32:
- params->ciphers[params->ciphersCount++] = MS_ZRTP_CIPHER_AES3;
- params->authTags[params->authTagsCount++] = MS_ZRTP_AUTHTAG_HS32;
- break;
- case MS_CRYPTO_SUITE_INVALID:
- break;
- }
- }
- }
+ getStreamsGroup().getIceService().restartSession(IR_Controlling);
+ MediaSessionParams newParams(*getParams());
+ q->update(&newParams);
+}
- /* linphone_core_get_srtp_crypto_suites is used to determine sensible defaults; here each can be overridden */
- MsZrtpCryptoTypesCount ciphersCount = linphone_core_get_zrtp_cipher_suites(q->getCore()->getCCore(), params->ciphers); /* if not present in config file, params->ciphers is not modified */
- if (ciphersCount != 0) /* Use zrtp_cipher_suites config only when present, keep config from srtp_crypto_suite otherwise */
- params->ciphersCount = ciphersCount;
- params->hashesCount = linphone_core_get_zrtp_hash_suites(q->getCore()->getCCore(), params->hashes);
- MsZrtpCryptoTypesCount authTagsCount = linphone_core_get_zrtp_auth_suites(q->getCore()->getCCore(), params->authTags); /* If not present in config file, params->authTags is not modified */
- if (authTagsCount != 0)
- params->authTagsCount = authTagsCount; /* Use zrtp_auth_suites config only when present, keep config from srtp_crypto_suite otherwise */
- params->sasTypesCount = linphone_core_get_zrtp_sas_suites(q->getCore()->getCCore(), params->sasTypes);
- params->keyAgreementsCount = linphone_core_get_zrtp_key_agreement_suites(q->getCore()->getCCore(), params->keyAgreements);
-
- bool haveRemoteZrtpHash = false;
- if (op && op->getRemoteMediaDescription()) {
- const SalStreamDescription *remoteStream = sal_media_description_find_best_stream(op->getRemoteMediaDescription(), SalAudio);
- if (remoteStream) {
- haveRemoteZrtpHash = remoteStream->haveZrtpHash;
- }
- }
-
- params->autoStart = (getParams()->getMediaEncryption() != LinphoneMediaEncryptionZRTP) && (haveRemoteZrtpHash == false) ;
+void MediaSessionPrivate::tryEarlyMediaForking (SalMediaDescription *md) {
+ OfferAnswerContext ctx;
+ ctx.localMediaDescription = localDesc;
+ ctx.remoteMediaDescription = md;
+ ctx.resultMediaDescription = resultDesc;
+ lInfo() << "Early media response received from another branch, checking if media can be forked to this new destination";
+ getStreamsGroup().tryEarlyMediaForking(ctx);
}
-void MediaSessionPrivate::startDtls (MSMediaStreamSessions *sessions, const SalStreamDescription *sd, const SalStreamDescription *remote) {
+void MediaSessionPrivate::updateStreamFrozenPayloads (SalStreamDescription *resultDesc, SalStreamDescription *localStreamDesc) {
L_Q();
-
- if (sal_stream_description_has_dtls(sd)) {
- if (sd->dtls_role == SalDtlsRoleInvalid)
- lWarning() << "Unable to start DTLS engine on stream session [" << sessions << "], Dtls role in resulting media description is invalid";
- else {
- /* Workaround for buggy openssl versions that send DTLS packets bigger than the MTU. We need to increase the recv buf size of the RtpSession.*/
- int recv_buf_size = lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()),"rtp", "dtls_recv_buf_size", 5000);
- rtp_session_set_recv_buf_size(sessions->rtp_session, recv_buf_size);
-
- /* If DTLS is available at both end points */
- /* Give the peer certificate fingerprint to dtls context */
- ms_dtls_srtp_set_peer_fingerprint(sessions->dtls_context, remote->dtls_fingerprint);
- ms_dtls_srtp_set_role(sessions->dtls_context, (sd->dtls_role == SalDtlsRoleIsClient) ? MSDtlsSrtpRoleIsClient : MSDtlsSrtpRoleIsServer); /* Set the role to client */
- ms_dtls_srtp_start(sessions->dtls_context); /* Then start the engine, it will send the DTLS client Hello */
+ for (bctbx_list_t *elem = resultDesc->payloads; elem != nullptr; elem = bctbx_list_next(elem)) {
+ OrtpPayloadType *pt = reinterpret_cast<OrtpPayloadType *>(bctbx_list_get_data(elem));
+ if (PayloadTypeHandler::isPayloadTypeNumberAvailable(localStreamDesc->already_assigned_payloads, payload_type_get_number(pt), nullptr)) {
+ /* New codec, needs to be added to the list */
+ localStreamDesc->already_assigned_payloads = bctbx_list_append(localStreamDesc->already_assigned_payloads, payload_type_clone(pt));
+ lInfo() << "CallSession[" << q << "] : payload type " << payload_type_get_number(pt) << " " << pt->mime_type << "/" << pt->clock_rate
+ << " fmtp=" << L_C_TO_STRING(pt->recv_fmtp) << " added to frozen list";
}
}
}
-void MediaSessionPrivate::startDtlsOnAudioStream () {
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- SalMediaDescription *result = op->getFinalMediaDescription();
-
- if (!remote || !result) {
- /* This can happen in some tricky cases (early-media without SDP in the 200). In that case, simply skip DTLS code */
- return;
+void MediaSessionPrivate::updateFrozenPayloads (SalMediaDescription *result) {
+ for (int i = 0; i < result->nb_streams; i++) {
+ updateStreamFrozenPayloads(&result->streams[i], &localDesc->streams[i]);
}
-
- if (audioStream && (media_stream_get_state(&audioStream->ms) == MSStreamStarted))
- startDtls(&audioStream->ms.sessions, sal_media_description_find_best_stream(result, SalAudio), sal_media_description_find_best_stream(remote, SalAudio));
}
-void MediaSessionPrivate::startDtlsOnVideoStream () {
-#ifdef VIDEO_ENABLED
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- SalMediaDescription *result = op->getFinalMediaDescription();
+void MediaSessionPrivate::updateStreams (SalMediaDescription *newMd, CallSession::State targetState) {
+ L_Q();
- if (!remote || !result) {
- /* This can happen in some tricky cases (early-media without SDP in the 200). In that case, simply skip DTLS code */
- return;
+ if (state == CallSession::State::Connected || state == CallSession::State::Resuming ||
+ (state == CallSession::State::IncomingEarlyMedia && !linphone_core_get_ring_during_incoming_early_media(q->getCore()->getCCore()))) {
+ q->getCore()->getPrivate()->getToneManager()->goToCall(q->getSharedFromThis());
}
- if (videoStream && (media_stream_get_state(&videoStream->ms) == MSStreamStarted))
- startDtls(&videoStream->ms.sessions, sal_media_description_find_best_stream(result, SalVideo), sal_media_description_find_best_stream(remote, SalVideo));
-#endif
-}
-
-void MediaSessionPrivate::startDtlsOnTextStream () {
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- SalMediaDescription *result = op->getFinalMediaDescription();
-
- if (!remote || !result) {
- /* This can happen in some tricky cases (early-media without SDP in the 200). In that case, simply skip DTLS code */
+ if (!newMd) {
+ lError() << "updateStreams() called with null media description";
return;
}
+
+ updateBiggestDesc(localDesc);
+ sal_media_description_ref(newMd);
+ SalMediaDescription *oldMd = resultDesc;
+ resultDesc = newMd;
+
+ OfferAnswerContext ctx;
+ ctx.localMediaDescription = localDesc;
+ ctx.remoteMediaDescription = op->getRemoteMediaDescription();
+ ctx.resultMediaDescription = resultDesc;
+ getStreamsGroup().render(ctx, targetState);
- if (textStream && (media_stream_get_state(&textStream->ms) == MSStreamStarted))
- startDtls(&textStream->ms.sessions, sal_media_description_find_best_stream(result, SalText), sal_media_description_find_best_stream(remote, SalText));
-}
-
-
-//might be the same interface as startDtls if audio_stream_start_zrtp is replaced by audio_streamsessions_start_zrtp
-void MediaSessionPrivate::startZrtpPrimaryChannel(const SalStreamDescription *remote) {
- if (remote->type != SalAudio) {
- lError() << "Cannot start primary zrtp channel for stream type ["
- << sal_stream_type_to_string(remote->type) << "]";
- return;
- }
- audio_stream_start_zrtp(audioStream);
- if (remote->haveZrtpHash == 1) {
- int retval = ms_zrtp_setPeerHelloHash(audioStream->ms.sessions.zrtp_context, (uint8_t *)remote->zrtphash, strlen((const char *)(remote->zrtphash)));
- if (retval != 0)
- lError() << "ZRTP hash mismatch 0x" << hex << retval;
+ if ((state == CallSession::State::Pausing) && pausedByApp && (q->getCore()->getCallCount() == 1)) {
+ q->getCore()->getPrivate()->getToneManager()->startNamedTone(q->getSharedFromThis(), LinphoneToneCallOnHold);
}
- return;
-}
-void MediaSessionPrivate::startDtlsOnAllStreams () {
- startDtlsOnAudioStream();
- startDtlsOnVideoStream();
- startDtlsOnTextStream();
+ updateFrozenPayloads(newMd);
+ upBandwidth = linphone_core_get_upload_bandwidth(q->getCore()->getCCore());
+
+ if (oldMd)
+ sal_media_description_unref(oldMd);
}
-void MediaSessionPrivate::updateStreamCryptoParameters (SalStreamDescription *oldStream, SalStreamDescription *newStream) {
- if (!oldStream || !newStream || oldStream->type != newStream->type)
- return;
+// -----------------------------------------------------------------------------
- const SalStreamDescription *localStreamDesc = sal_media_description_find_secure_stream_of_type(localDesc, newStream->type);
- if (newStream->type == SalAudio) {
- if (audioStream && localStreamDesc) {
- updateCryptoParameters(localStreamDesc, oldStream, newStream, &audioStream->ms);
- startDtlsOnAudioStream();
- }
- }
-#ifdef VIDEO_ENABLED
- else if (newStream->type == SalVideo) {
- if (videoStream && localStreamDesc) {
- updateCryptoParameters(localStreamDesc, oldStream, newStream, &videoStream->ms);
- startDtlsOnVideoStream();
- }
- }
-#endif
- else if (newStream->type == SalText) {
- if (textStream && localStreamDesc) {
- updateCryptoParameters(localStreamDesc, oldStream, newStream, &textStream->ms);
- startDtlsOnTextStream();
- }
- }
+bool MediaSessionPrivate::allStreamsAvpfEnabled () const {
+ return getStreamsGroup().avpfEnabled();
}
-void MediaSessionPrivate::updateStreamsCryptoParameters (SalMediaDescription *oldMd, SalMediaDescription *newMd) {
- const SalStreamDescription *localStreamDesc = sal_media_description_find_secure_stream_of_type(localDesc, SalAudio);
- SalStreamDescription *oldStream = sal_media_description_find_secure_stream_of_type(oldMd, SalAudio);
- SalStreamDescription *newStream = sal_media_description_find_secure_stream_of_type(newMd, SalAudio);
- if (audioStream && localStreamDesc && oldStream && newStream)
- updateCryptoParameters(localStreamDesc, oldStream, newStream, &audioStream->ms);
-#ifdef VIDEO_ENABLED
- localStreamDesc = sal_media_description_find_secure_stream_of_type(localDesc, SalVideo);
- oldStream = sal_media_description_find_secure_stream_of_type(oldMd, SalVideo);
- newStream = sal_media_description_find_secure_stream_of_type(newMd, SalVideo);
- if (videoStream && localStreamDesc && oldStream && newStream)
- updateCryptoParameters(localStreamDesc, oldStream, newStream, &videoStream->ms);
-#endif
- localStreamDesc = sal_media_description_find_secure_stream_of_type(localDesc, SalText);
- oldStream = sal_media_description_find_secure_stream_of_type(oldMd, SalText);
- newStream = sal_media_description_find_secure_stream_of_type(newMd, SalText);
- if (textStream && localStreamDesc && oldStream && newStream)
- updateCryptoParameters(localStreamDesc, oldStream, newStream, &textStream->ms);
- startDtlsOnAllStreams();
+bool MediaSessionPrivate::allStreamsEncrypted () const {
+ return getStreamsGroup().allStreamsEncrypted();
}
-bool MediaSessionPrivate::updateCryptoParameters (const SalStreamDescription *localStreamDesc, SalStreamDescription *oldStream, SalStreamDescription *newStream, MediaStream *ms) {
- int cryptoIdx = Sal::findCryptoIndexFromTag(localStreamDesc->crypto, static_cast<unsigned char>(newStream->crypto_local_tag));
- if (cryptoIdx >= 0) {
- if (localDescChanged & SAL_MEDIA_DESCRIPTION_CRYPTO_KEYS_CHANGED)
- ms_media_stream_sessions_set_srtp_send_key_b64(&ms->sessions, newStream->crypto[0].algo, localStreamDesc->crypto[cryptoIdx].master_key);
- if (strcmp(oldStream->crypto[0].master_key, newStream->crypto[0].master_key) != 0)
- ms_media_stream_sessions_set_srtp_recv_key_b64(&ms->sessions, newStream->crypto[0].algo, newStream->crypto[0].master_key);
- return true;
- } else
- lWarning() << "Failed to find local crypto algo with tag: " << newStream->crypto_local_tag;
- return false;
+bool MediaSessionPrivate::atLeastOneStreamStarted () const {
+ return getStreamsGroup().isStarted();
}
-// -----------------------------------------------------------------------------
+uint16_t MediaSessionPrivate::getAvpfRrInterval () const {
+ return (uint16_t)getStreamsGroup().getAvpfRrInterval();
+}
-int MediaSessionPrivate::getIdealAudioBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc) {
- L_Q();
- int remoteBandwidth = 0;
- if (desc->bandwidth > 0)
- remoteBandwidth = desc->bandwidth;
- else if (md->bandwidth > 0) {
- /* Case where b=AS is given globally, not per stream */
- remoteBandwidth = md->bandwidth;
- }
- int uploadBandwidth = 0;
- bool forced = false;
- if (getParams()->getPrivate()->getUpBandwidth() > 0) {
- forced = true;
- uploadBandwidth = getParams()->getPrivate()->getUpBandwidth();
- } else
- uploadBandwidth = linphone_core_get_upload_bandwidth(q->getCore()->getCCore());
- uploadBandwidth = PayloadTypeHandler::getMinBandwidth(uploadBandwidth, remoteBandwidth);
- if (!linphone_core_media_description_contains_video_stream(md) || forced)
- return uploadBandwidth;
- if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 512))
- uploadBandwidth = 100;
- else if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 256))
- uploadBandwidth = 64;
- else if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 128))
- uploadBandwidth = 40;
- else if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 0))
- uploadBandwidth = 24;
- return uploadBandwidth;
-}
-
-int MediaSessionPrivate::getVideoBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc) {
- L_Q();
- int remoteBandwidth = 0;
- if (desc->bandwidth > 0)
- remoteBandwidth = desc->bandwidth;
- else if (md->bandwidth > 0) {
- /* Case where b=AS is given globally, not per stream */
- remoteBandwidth = PayloadTypeHandler::getRemainingBandwidthForVideo(md->bandwidth, audioBandwidth);
- }
- return PayloadTypeHandler::getMinBandwidth(PayloadTypeHandler::getRemainingBandwidthForVideo(linphone_core_get_upload_bandwidth(q->getCore()->getCCore()), audioBandwidth), remoteBandwidth);
+unsigned int MediaSessionPrivate::getNbActiveStreams () const {
+ return (unsigned int)getStreamsGroup().getActiveStreamsCount();
}
-RtpProfile * MediaSessionPrivate::makeProfile (const SalMediaDescription *md, const SalStreamDescription *desc, int *usedPt) {
+bool MediaSessionPrivate::isEncryptionMandatory () const {
L_Q();
- *usedPt = -1;
- int bandwidth = 0;
- if (desc->type == SalAudio)
- bandwidth = getIdealAudioBandwidth(md, desc);
- else if (desc->type == SalVideo)
- bandwidth = getVideoBandwidth(md, desc);
-
- bool first = true;
- RtpProfile *profile = rtp_profile_new("Call profile");
- for (const bctbx_list_t *elem = desc->payloads; elem != nullptr; elem = bctbx_list_next(elem)) {
- OrtpPayloadType *pt = reinterpret_cast<OrtpPayloadType *>(bctbx_list_get_data(elem));
- /* Make a copy of the payload type, so that we left the ones from the SalStreamDescription unchanged.
- * If the SalStreamDescription is freed, this will have no impact on the running streams. */
- pt = payload_type_clone(pt);
- int upPtime = 0;
- if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
- /* First codec in list is the selected one */
- if (desc->type == SalAudio) {
- updateAllocatedAudioBandwidth(pt, bandwidth);
- bandwidth = audioBandwidth;
- upPtime = getParams()->getPrivate()->getUpPtime();
- if (!upPtime)
- upPtime = linphone_core_get_upload_ptime(q->getCore()->getCCore());
- }
- first = false;
- }
- if (*usedPt == -1) {
- /* Don't select telephone-event as a payload type */
- if (strcasecmp(pt->mime_type, "telephone-event") != 0)
- *usedPt = payload_type_get_number(pt);
- }
- if (pt->flags & PAYLOAD_TYPE_BITRATE_OVERRIDE) {
- lInfo() << "Payload type [" << pt->mime_type << "/" << pt->clock_rate << "] has explicit bitrate [" << (pt->normal_bitrate / 1000) << "] kbit/s";
- pt->normal_bitrate = PayloadTypeHandler::getMinBandwidth(pt->normal_bitrate, bandwidth * 1000);
- } else
- pt->normal_bitrate = bandwidth * 1000;
- if (desc->maxptime > 0) {// follow the same schema for maxptime as for ptime. (I.E add it to fmtp)
- ostringstream os;
- os << "maxptime=" << desc->maxptime;
- payload_type_append_send_fmtp(pt, os.str().c_str());
- }
- if (desc->ptime > 0)
- upPtime = desc->ptime;
- if (upPtime > 0) {
- ostringstream os;
- os << "ptime=" << upPtime;
- payload_type_append_send_fmtp(pt, os.str().c_str());
- }
- int number = payload_type_get_number(pt);
- if (rtp_profile_get_payload(profile, number))
- lWarning() << "A payload type with number " << number << " already exists in profile!";
- else
- rtp_profile_set_payload(profile, number, pt);
+ if (getParams()->getMediaEncryption() == LinphoneMediaEncryptionDTLS) {
+ lInfo() << "Forced encryption mandatory on CallSession [" << q << "] due to SRTP-DTLS";
+ return true;
}
- return profile;
-}
-
-void MediaSessionPrivate::unsetRtpProfile (int streamIndex) {
- if (sessions[streamIndex].rtp_session)
- rtp_session_set_profile(sessions[streamIndex].rtp_session, &av_profile);
-}
-
-void MediaSessionPrivate::updateAllocatedAudioBandwidth (const PayloadType *pt, int maxbw) {
- L_Q();
- audioBandwidth = PayloadTypeHandler::getAudioPayloadTypeBandwidth(pt, maxbw);
- lInfo() << "Audio bandwidth for CallSession [" << q << "] is " << audioBandwidth;
-}
-
-// -----------------------------------------------------------------------------
-
-void MediaSessionPrivate::applyJitterBufferParams (RtpSession *session, LinphoneStreamType type) {
- L_Q();
- LinphoneConfig *config = linphone_core_get_config(q->getCore()->getCCore());
- JBParameters params;
- rtp_session_get_jitter_buffer_params(session, ¶ms);
- params.min_size = lp_config_get_int(config, "rtp", "jitter_buffer_min_size", 40);
- params.max_size = lp_config_get_int(config, "rtp", "jitter_buffer_max_size", 500);
- params.max_packets = params.max_size * 200 / 1000; /* Allow 200 packet per seconds, quite large */
- const char *algo = lp_config_get_string(config, "rtp", "jitter_buffer_algorithm", "rls");
- params.buffer_algorithm = jitterBufferNameToAlgo(algo ? algo : "");
- params.refresh_ms = lp_config_get_int(config, "rtp", "jitter_buffer_refresh_period", 5000);
- params.ramp_refresh_ms = lp_config_get_int(config, "rtp", "jitter_buffer_ramp_refresh_period", 5000);
- params.ramp_step_ms = lp_config_get_int(config, "rtp", "jitter_buffer_ramp_step", 20);
- params.ramp_threshold = lp_config_get_int(config, "rtp", "jitter_buffer_ramp_threshold", 70);
-
- switch (type) {
- case LinphoneStreamTypeAudio:
- case LinphoneStreamTypeText: /* Let's use the same params for text as for audio */
- params.nom_size = linphone_core_get_audio_jittcomp(q->getCore()->getCCore());
- params.adaptive = linphone_core_audio_adaptive_jittcomp_enabled(q->getCore()->getCCore());
- break;
- case LinphoneStreamTypeVideo:
- params.nom_size = linphone_core_get_video_jittcomp(q->getCore()->getCCore());
- params.adaptive = linphone_core_video_adaptive_jittcomp_enabled(q->getCore()->getCCore());
- break;
- case LinphoneStreamTypeUnknown:
- lError() << "applyJitterBufferParams: should not happen";
- break;
- }
- params.enabled = params.nom_size > 0;
- if (params.enabled) {
- if (params.min_size > params.nom_size)
- params.min_size = params.nom_size;
- if (params.max_size < params.nom_size)
- params.max_size = params.nom_size;
- }
- rtp_session_set_jitter_buffer_params(session, ¶ms);
-}
-
-void MediaSessionPrivate::clearEarlyMediaDestination (MediaStream *ms) {
- L_Q();
- RtpSession *session = ms->sessions.rtp_session;
- rtp_session_clear_aux_remote_addr(session);
- /* Restore symmetric rtp if ICE is not used */
- if (!iceAgent->hasSession())
- rtp_session_set_symmetric_rtp(session, linphone_core_symmetric_rtp_enabled(q->getCore()->getCCore()));
-}
-
-void MediaSessionPrivate::clearEarlyMediaDestinations () {
- if (audioStream)
- clearEarlyMediaDestination(&audioStream->ms);
- if (videoStream)
- clearEarlyMediaDestination(&videoStream->ms);
-}
-
-void MediaSessionPrivate::configureAdaptiveRateControl (MediaStream *ms, const OrtpPayloadType *pt, bool videoWillBeUsed) {
- L_Q();
- bool enabled = !!linphone_core_adaptive_rate_control_enabled(q->getCore()->getCCore());
- if (!enabled) {
- media_stream_enable_adaptive_bitrate_control(ms, false);
- return;
- }
- bool isAdvanced = true;
- string algo = linphone_core_get_adaptive_rate_algorithm(q->getCore()->getCCore());
- if (algo == "basic")
- isAdvanced = false;
- else if (algo == "advanced")
- isAdvanced = true;
- if (isAdvanced) {
- /* We can't use media_stream_avpf_enabled() here because the active PayloadType is not set yet in the MediaStream */
- if (!pt || !(pt->flags & PAYLOAD_TYPE_RTCP_FEEDBACK_ENABLED)) {
- lWarning() << "CallSession [" << q << "] - advanced adaptive rate control requested but avpf is not activated in this stream. Reverting to basic rate control instead";
- isAdvanced = false;
- } else
- lInfo() << "CallSession [" << q << "] - setting up advanced rate control";
- }
- if (isAdvanced) {
- ms_bandwidth_controller_add_stream(q->getCore()->getCCore()->bw_controller, ms);
- media_stream_enable_adaptive_bitrate_control(ms, false);
- } else {
- media_stream_set_adaptive_bitrate_algorithm(ms, MSQosAnalyzerAlgorithmSimple);
- if ((ms->type == MSAudio) && videoWillBeUsed) {
- /* If this is an audio stream but video is going to be used, there is no need to perform
- * basic rate control on the audio stream, just the video stream. */
- enabled = false;
- }
- media_stream_enable_adaptive_bitrate_control(ms, enabled);
- }
-}
-
-void MediaSessionPrivate::configureRtpSessionForRtcpFb (const SalStreamDescription *stream) {
- RtpSession *session = nullptr;
- if (stream->type == SalAudio)
- session = audioStream->ms.sessions.rtp_session;
- else if (stream->type == SalVideo)
- session = videoStream->ms.sessions.rtp_session;
- else
- return; /* Do nothing for streams that are not audio or video */
- if (stream->rtcp_fb.generic_nack_enabled)
- rtp_session_enable_avpf_feature(session, ORTP_AVPF_FEATURE_GENERIC_NACK, true);
- else
- rtp_session_enable_avpf_feature(session, ORTP_AVPF_FEATURE_GENERIC_NACK, false);
- if (stream->rtcp_fb.tmmbr_enabled)
- rtp_session_enable_avpf_feature(session, ORTP_AVPF_FEATURE_TMMBR, true);
- else
- rtp_session_enable_avpf_feature(session, ORTP_AVPF_FEATURE_TMMBR, false);
-}
-
-void MediaSessionPrivate::configureRtpSessionForRtcpXr (SalStreamType type) {
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- if (!remote)
- return;
- const SalStreamDescription *localStream = sal_media_description_find_best_stream(localDesc, type);
- if (!localStream)
- return;
- const SalStreamDescription *remoteStream = sal_media_description_find_best_stream(remote, type);
- if (!remoteStream)
- return;
- OrtpRtcpXrConfiguration currentConfig;
- const OrtpRtcpXrConfiguration *remoteConfig = &remoteStream->rtcp_xr;
- if (localStream->dir == SalStreamInactive)
- return;
- else if (localStream->dir == SalStreamRecvOnly) {
- /* Use local config for unilateral parameters and remote config for collaborative parameters */
- memcpy(¤tConfig, &localStream->rtcp_xr, sizeof(currentConfig));
- currentConfig.rcvr_rtt_mode = remoteConfig->rcvr_rtt_mode;
- currentConfig.rcvr_rtt_max_size = remoteConfig->rcvr_rtt_max_size;
- } else
- memcpy(¤tConfig, remoteConfig, sizeof(currentConfig));
- RtpSession *session = nullptr;
- if (type == SalAudio) {
- session = audioStream->ms.sessions.rtp_session;
- } else if (type == SalVideo) {
- session = videoStream->ms.sessions.rtp_session;
- } else if (type == SalText) {
- session = textStream->ms.sessions.rtp_session;
- }
- rtp_session_configure_rtcp_xr(session, ¤tConfig);
-}
-
-RtpSession * MediaSessionPrivate::createAudioRtpIoSession () {
- L_Q();
- LinphoneConfig *config = linphone_core_get_config(q->getCore()->getCCore());
- const char *rtpmap = lp_config_get_string(config, "sound", "rtp_map", "pcmu/8000/1");
- OrtpPayloadType *pt = rtp_profile_get_payload_from_rtpmap(audioProfile, rtpmap);
- if (!pt)
- return nullptr;
- rtpIoAudioProfile = rtp_profile_new("RTP IO audio profile");
- int ptnum = lp_config_get_int(config, "sound", "rtp_ptnum", 0);
- rtp_profile_set_payload(rtpIoAudioProfile, ptnum, payload_type_clone(pt));
- const char *localIp = lp_config_get_string(config, "sound", "rtp_local_addr", "127.0.0.1");
- int localPort = lp_config_get_int(config, "sound", "rtp_local_port", 17076);
- RtpSession *rtpSession = ms_create_duplex_rtp_session(localIp, localPort, -1, ms_factory_get_mtu(q->getCore()->getCCore()->factory));
- rtp_session_set_profile(rtpSession, rtpIoAudioProfile);
- const char *remoteIp = lp_config_get_string(config, "sound", "rtp_remote_addr", "127.0.0.1");
- int remotePort = lp_config_get_int(config, "sound", "rtp_remote_port", 17078);
- rtp_session_set_remote_addr_and_port(rtpSession, remoteIp, remotePort, -1);
- rtp_session_enable_rtcp(rtpSession, false);
- rtp_session_set_payload_type(rtpSession, ptnum);
- int jittcomp = lp_config_get_int(config, "sound", "rtp_jittcomp", 0); /* 0 means no jitter buffer */
- rtp_session_set_jitter_compensation(rtpSession, jittcomp);
- rtp_session_enable_jitter_buffer(rtpSession, (jittcomp > 0));
- bool symmetric = !!lp_config_get_int(config, "sound", "rtp_symmetric", 0);
- rtp_session_set_symmetric_rtp(rtpSession, symmetric);
- return rtpSession;
-}
-
-RtpSession * MediaSessionPrivate::createVideoRtpIoSession () {
-#ifdef VIDEO_ENABLED
- L_Q();
- LinphoneConfig *config = linphone_core_get_config(q->getCore()->getCCore());
- const char *rtpmap = lp_config_get_string(config, "video", "rtp_map", "vp8/90000/1");
- OrtpPayloadType *pt = rtp_profile_get_payload_from_rtpmap(videoProfile, rtpmap);
- if (!pt)
- return nullptr;
- rtpIoVideoProfile = rtp_profile_new("RTP IO video profile");
- int ptnum = lp_config_get_int(config, "video", "rtp_ptnum", 0);
- rtp_profile_set_payload(rtpIoVideoProfile, ptnum, payload_type_clone(pt));
- const char *localIp = lp_config_get_string(config, "video", "rtp_local_addr", "127.0.0.1");
- int localPort = lp_config_get_int(config, "video", "rtp_local_port", 19076);
- RtpSession *rtpSession = ms_create_duplex_rtp_session(localIp, localPort, -1, ms_factory_get_mtu(q->getCore()->getCCore()->factory));
- rtp_session_set_profile(rtpSession, rtpIoVideoProfile);
- const char *remoteIp = lp_config_get_string(config, "video", "rtp_remote_addr", "127.0.0.1");
- int remotePort = lp_config_get_int(config, "video", "rtp_remote_port", 19078);
- rtp_session_set_remote_addr_and_port(rtpSession, remoteIp, remotePort, -1);
- rtp_session_enable_rtcp(rtpSession, false);
- rtp_session_set_payload_type(rtpSession, ptnum);
- rtp_session_set_symmetric_rtp(rtpSession, linphone_config_get_bool(config, "video", "rtp_symmetric", FALSE));
- int jittcomp = lp_config_get_int(config, "video", "rtp_jittcomp", 0); /* 0 means no jitter buffer */
- rtp_session_set_jitter_compensation(rtpSession, jittcomp);
- rtp_session_enable_jitter_buffer(rtpSession, (jittcomp > 0));
- return rtpSession;
-#else
- return nullptr;
-#endif
-}
-
-/*
- * Frees the media resources of the call.
- * This has to be done at the earliest, unlike signaling resources that sometimes need to be kept a bit more longer.
- * It is called by setTerminated() (for termination of calls signaled to the application), or directly by the destructor of the session
- * if it was never notified to the application.
- */
-void MediaSessionPrivate::freeResources () {
- stopStreams();
- iceAgent->deleteSession();
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++)
- ms_media_stream_sessions_uninit(&sessions[i]);
- _linphone_call_stats_uninit(audioStats);
- _linphone_call_stats_uninit(videoStats);
- _linphone_call_stats_uninit(textStats);
-}
-
-void MediaSessionPrivate::handleIceEvents (OrtpEvent *ev) {
- L_Q();
- OrtpEventType evt = ortp_event_get_type(ev);
- OrtpEventData *evd = ortp_event_get_data(ev);
- if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
- if (iceAgent->hasCompletedCheckList()) {
- /* The ICE session has succeeded, so perform a call update */
- if (iceAgent->isControlling() && q->getCurrentParams()->getPrivate()->getUpdateCallWhenIceCompleted()) {
- if (state == CallSession::State::StreamsRunning){
- MediaSessionParams newParams(*getParams());
- newParams.getPrivate()->setInternalCallUpdate(true);
- q->update(&newParams);
- }else{
- lWarning() << "Cannot send reINVITE for ICE during state " << state;
- }
- }else if (!iceAgent->isControlling() && incomingIceReinvitePending){
- q->acceptUpdate(nullptr);
- incomingIceReinvitePending = false;
- }
- startDtlsOnAllStreams();
- }
- iceAgent->updateIceStateInCallStats();
- } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
- if (!evd->info.ice_processing_successful)
- lWarning() << "No STUN answer from [" << linphone_nat_policy_get_stun_server(q->getPrivate()->getNatPolicy()) << "], continuing without STUN";
- iceAgent->gatheringFinished();
- switch (state) {
- case CallSession::State::Updating:
- startUpdate();
- break;
- case CallSession::State::UpdatedByRemote:
- startAcceptUpdate(prevState, Utils::toString(prevState));
- break;
- case CallSession::State::OutgoingInit:
- stopStreamsForIceGathering();
- if (isReadyForInvite())
- q->startInvite(nullptr, "");
- break;
- case CallSession::State::Idle:
- stopStreamsForIceGathering();
- updateLocalMediaDescriptionFromIce();
- op->setLocalMediaDescription(localDesc);
- deferIncomingNotification = false;
- startIncomingNotification();
- break;
- default:
- break;
- }
- } else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
- if (state == CallSession::State::UpdatedByRemote) {
- startAcceptUpdate(prevState, Utils::toString(prevState));
- iceAgent->updateIceStateInCallStats();
- }
- } else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
- iceAgent->restartSession(IR_Controlling);
- q->update(getCurrentParams());
- }
-}
-
-void MediaSessionPrivate::handleStreamEvents (int streamIndex) {
- L_Q();
-
- MediaStream *ms = getMediaStream(streamIndex);
- if (ms) {
- /* Ensure there is no dangling ICE check list */
- if (!iceAgent->hasSession())
- media_stream_set_ice_check_list(ms, nullptr);
- switch(ms->type){
- case MSAudio:
- audio_stream_iterate((AudioStream *)ms);
- break;
- case MSVideo:
-#ifdef VIDEO_ENABLED
- video_stream_iterate((VideoStream *)ms);
-#endif
- break;
- case MSText:
- text_stream_iterate((TextStream *)ms);
- break;
- default:
- lError() << "handleStreamEvents(): unsupported stream type";
- return;
- }
- }
- OrtpEvQueue *evq;
- OrtpEvent *ev;
- /* Yes the event queue has to be taken at each iteration, because ice events may perform operations re-creating the streams */
- while ((evq = getEventQueue(streamIndex)) && (ev = ortp_ev_queue_get(evq))) {
- LinphoneCallStats *stats = nullptr;
- if (streamIndex == mainAudioStreamIndex)
- stats = audioStats;
- else if (streamIndex == mainVideoStreamIndex)
- stats = videoStats;
- else
- stats = textStats;
-
- OrtpEventType evt = ortp_event_get_type(ev);
- OrtpEventData *evd = ortp_event_get_data(ev);
-
- /*This MUST be done before any call to "linphone_call_stats_fill" since it has ownership over evd->packet*/
- if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
- do {
- if (evd->packet && rtcp_is_RTPFB(evd->packet)) {
- if (rtcp_RTPFB_get_type(evd->packet) == RTCP_RTPFB_TMMBR) {
- listener->onTmmbrReceived(q->getSharedFromThis(), streamIndex, (int)rtcp_RTPFB_tmmbr_get_max_bitrate(evd->packet));
- }
- }
- } while (rtcp_next_packet(evd->packet));
- rtcp_rewind(evd->packet);
- }
-
- /* And yes the MediaStream must be taken at each iteration, because it may have changed due to the handling of events
- * in this loop*/
- ms = getMediaStream(streamIndex);
- if (ms)
- linphone_call_stats_fill(stats, ms, ev);
- notifyStatsUpdated(streamIndex);
- if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED) {
- if (streamIndex == mainAudioStreamIndex)
- audioStreamEncryptionChanged(!!evd->info.zrtp_stream_encrypted);
- else if (streamIndex == mainVideoStreamIndex)
- propagateEncryptionChanged();
- } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
- if (streamIndex == mainAudioStreamIndex)
- audioStreamAuthTokenReady(evd->info.zrtp_info.sas, !!evd->info.zrtp_info.verified);
- } else if (evt == ORTP_EVENT_DTLS_ENCRYPTION_CHANGED) {
- if (streamIndex == mainAudioStreamIndex)
- audioStreamEncryptionChanged(!!evd->info.dtls_stream_encrypted);
- else if (streamIndex == mainVideoStreamIndex)
- propagateEncryptionChanged();
- } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
- || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
- if (ms)
- handleIceEvents(ev);
- } else if (evt == ORTP_EVENT_TELEPHONE_EVENT) {
- telephoneEventReceived(evd->info.telephone_event);
- } else if (evt == ORTP_EVENT_NEW_VIDEO_BANDWIDTH_ESTIMATION_AVAILABLE) {
- lInfo() << "Video bandwidth estimation is " << (int)(evd->info.video_bandwidth_available / 1000.) << " kbit/s";
- /* If this event happens then it should be a video stream */
- if (streamIndex == mainVideoStreamIndex)
- linphone_call_stats_set_estimated_download_bandwidth(stats, (float)(evd->info.video_bandwidth_available*1e-3));
- }
- ortp_event_destroy(ev);
- }
-}
-
-void MediaSessionPrivate::configureRtpSession(RtpSession *session, LinphoneStreamType streamType){
- L_Q();
-
- rtp_session_enable_network_simulation(session, &q->getCore()->getCCore()->net_conf.netsim_params);
- applyJitterBufferParams(session, streamType);
- string userAgent = linphone_core_get_user_agent(q->getCore()->getCCore());
- rtp_session_set_source_description(session, getMe()->getAddress().asString().c_str(), NULL, NULL, NULL, NULL, userAgent.c_str(), NULL);
- rtp_session_set_symmetric_rtp(session, linphone_core_symmetric_rtp_enabled(q->getCore()->getCCore()));
-
- if (streamType == LinphoneStreamTypeVideo){
- int videoRecvBufSize = lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "video", "recv_buf_size", 0);
- if (videoRecvBufSize > 0)
- rtp_session_set_recv_buf_size(videoStream->ms.sessions.rtp_session, videoRecvBufSize);
- }
-}
-
-
-void MediaSessionPrivate::initializeAudioStream () {
- L_Q();
-
- if (audioStream)
- return;
- if (!sessions[mainAudioStreamIndex].rtp_session) {
- SalMulticastRole multicastRole = getMulticastRole(SalAudio);
- SalMediaDescription *remoteDesc = nullptr;
- SalStreamDescription *streamDesc = nullptr;
- if (op)
- remoteDesc = op->getRemoteMediaDescription();
- if (remoteDesc)
- streamDesc = sal_media_description_find_best_stream(remoteDesc, SalAudio);
-
- audioStream = audio_stream_new2(q->getCore()->getCCore()->factory, L_STRING_TO_C(getBindIpForStream(mainAudioStreamIndex)),
- (multicastRole == SalMulticastReceiver) ? streamDesc->rtp_port : mediaPorts[mainAudioStreamIndex].rtpPort,
- (multicastRole == SalMulticastReceiver) ? 0 /* Disabled for now */ : mediaPorts[mainAudioStreamIndex].rtcpPort);
- if (multicastRole == SalMulticastReceiver)
- joinMulticastGroup(mainAudioStreamIndex, &audioStream->ms);
-
- configureRtpSession(audioStream->ms.sessions.rtp_session, LinphoneStreamTypeAudio);
- setupDtlsParams(&audioStream->ms);
-
- /* Initialize zrtp even if we didn't explicitely set it, just in case peer offers it */
- if (linphone_core_media_encryption_supported(q->getCore()->getCCore(), LinphoneMediaEncryptionZRTP)) {
- LinphoneAddress *peerAddr = (direction == LinphoneCallIncoming) ? log->from : log->to;
- LinphoneAddress *selfAddr = (direction == LinphoneCallIncoming) ? log->to : log->from;
- char *peerUri = ms_strdup_printf("%s:%s@%s" , linphone_address_get_scheme(peerAddr)
- , linphone_address_get_username(peerAddr)
- , linphone_address_get_domain(peerAddr));
- char *selfUri = ms_strdup_printf("%s:%s@%s" , linphone_address_get_scheme(selfAddr)
- , linphone_address_get_username(selfAddr)
- , linphone_address_get_domain(selfAddr));
-
- MSZrtpParams params;
- zrtpCacheAccess zrtpCacheInfo = linphone_core_get_zrtp_cache_access(q->getCore()->getCCore());
-
- memset(¶ms, 0, sizeof(MSZrtpParams));
- /* media encryption of current params will be set later when zrtp is activated */
- params.zidCacheDB = zrtpCacheInfo.db;
- params.zidCacheDBMutex = zrtpCacheInfo.dbMutex;
- params.peerUri = peerUri;
- params.selfUri = selfUri;
- /* Get key lifespan from config file, default is 0:forever valid */
- params.limeKeyTimeSpan = bctbx_time_string_to_sec(lp_config_get_string(linphone_core_get_config(q->getCore()->getCCore()), "sip", "lime_key_validity", "0"));
- setZrtpCryptoTypesParameters(¶ms);
- audio_stream_enable_zrtp(audioStream, ¶ms);
- if (peerUri)
- ms_free(peerUri);
- if (selfUri)
- ms_free(selfUri);
- }
-
- media_stream_reclaim_sessions(&audioStream->ms, &sessions[mainAudioStreamIndex]);
- } else {
- audioStream = audio_stream_new_with_sessions(q->getCore()->getCCore()->factory, &sessions[mainAudioStreamIndex]);
- }
-
- MSSndCard *playcard = q->getCore()->getCCore()->sound_conf.lsd_card ? q->getCore()->getCCore()->sound_conf.lsd_card : q->getCore()->getCCore()->sound_conf.play_sndcard;
- if (playcard) {
- // Set the stream type immediately, as on iOS AudioUnit is instanciated very early because it is
- // otherwise too slow to start.
- ms_snd_card_set_stream_type(playcard, MS_SND_CARD_STREAM_VOICE);
- }
-
- if (mediaPorts[mainAudioStreamIndex].rtpPort == -1)
- setPortConfigFromRtpSession(mainAudioStreamIndex, audioStream->ms.sessions.rtp_session);
- int dscp = linphone_core_get_audio_dscp(q->getCore()->getCCore());
- if (dscp != -1)
- audio_stream_set_dscp(audioStream, dscp);
- if (linphone_core_echo_limiter_enabled(q->getCore()->getCCore())) {
- string type = lp_config_get_string(linphone_core_get_config(q->getCore()->getCCore()), "sound", "el_type", "mic");
- if (type == "mic")
- audio_stream_enable_echo_limiter(audioStream, ELControlMic);
- else if (type == "full")
- audio_stream_enable_echo_limiter(audioStream, ELControlFull);
- }
-
- // Equalizer location in the graph: 'mic' = in input graph, otherwise in output graph.
- // Any other value than mic will default to output graph for compatibility.
- string location = lp_config_get_string(linphone_core_get_config(q->getCore()->getCCore()), "sound", "eq_location", "hp");
- audioStream->eq_loc = (location == "mic") ? MSEqualizerMic : MSEqualizerHP;
- lInfo() << "Equalizer location: " << location;
-
- audio_stream_enable_gain_control(audioStream, true);
- if (linphone_core_echo_cancellation_enabled(q->getCore()->getCCore())) {
- int len = lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "ec_tail_len", 0);
- int delay = lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "ec_delay", 0);
- int framesize = lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "ec_framesize", 0);
- audio_stream_set_echo_canceller_params(audioStream, len, delay, framesize);
- if (audioStream->ec) {
- char *statestr=reinterpret_cast<char *>(ms_malloc0(ecStateMaxLen));
- if (lp_config_relative_file_exists(linphone_core_get_config(q->getCore()->getCCore()), ecStateStore.c_str())
- && (lp_config_read_relative_file(linphone_core_get_config(q->getCore()->getCCore()), ecStateStore.c_str(), statestr, ecStateMaxLen) == 0)) {
- ms_filter_call_method(audioStream->ec, MS_ECHO_CANCELLER_SET_STATE_STRING, statestr);
- }
- ms_free(statestr);
- }
- }
- audio_stream_enable_automatic_gain_control(audioStream, linphone_core_agc_enabled(q->getCore()->getCCore()));
- bool_t enabled = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "noisegate", 0);
- audio_stream_enable_noise_gate(audioStream, enabled);
- audio_stream_set_features(audioStream, linphone_core_get_audio_features(q->getCore()->getCCore()));
-
- if (q->getCore()->getCCore()->rtptf) {
- RtpTransport *meta_rtp;
- RtpTransport *meta_rtcp;
- rtp_session_get_transports(audioStream->ms.sessions.rtp_session, &meta_rtp, &meta_rtcp);
- if (!meta_rtp_transport_get_endpoint(meta_rtp)) {
- lInfo() << "CallSession [" << q << "] using custom audio RTP transport endpoint";
- meta_rtp_transport_set_endpoint(meta_rtp, q->getCore()->getCCore()->rtptf->audio_rtp_func(q->getCore()->getCCore()->rtptf->audio_rtp_func_data, mediaPorts[mainAudioStreamIndex].rtpPort));
- }
- if (!meta_rtp_transport_get_endpoint(meta_rtcp))
- meta_rtp_transport_set_endpoint(meta_rtcp, q->getCore()->getCCore()->rtptf->audio_rtcp_func(q->getCore()->getCCore()->rtptf->audio_rtcp_func_data, mediaPorts[mainAudioStreamIndex].rtcpPort));
- }
-
- audioStreamEvQueue = ortp_ev_queue_new();
- rtp_session_register_event_queue(audioStream->ms.sessions.rtp_session, audioStreamEvQueue);
- iceAgent->prepareIceForStream(&audioStream->ms, false);
-}
-
-void MediaSessionPrivate::initializeTextStream () {
- L_Q();
- if (textStream)
- return;
- if (!sessions[mainTextStreamIndex].rtp_session) {
- SalMulticastRole multicastRole = getMulticastRole(SalText);
- SalMediaDescription *remoteDesc = nullptr;
- SalStreamDescription *streamDesc = nullptr;
- if (op)
- remoteDesc = op->getRemoteMediaDescription();
- if (remoteDesc)
- streamDesc = sal_media_description_find_best_stream(remoteDesc, SalText);
-
- textStream = text_stream_new2(q->getCore()->getCCore()->factory, L_STRING_TO_C(getBindIpForStream(mainTextStreamIndex)),
- (multicastRole == SalMulticastReceiver) ? streamDesc->rtp_port : mediaPorts[mainTextStreamIndex].rtpPort,
- (multicastRole == SalMulticastReceiver) ? 0 /* Disabled for now */ : mediaPorts[mainTextStreamIndex].rtcpPort);
- if (multicastRole == SalMulticastReceiver)
- joinMulticastGroup(mainTextStreamIndex, &textStream->ms);
-
- configureRtpSession(textStream->ms.sessions.rtp_session, LinphoneStreamTypeText);
- setupDtlsParams(&textStream->ms);
- media_stream_reclaim_sessions(&textStream->ms, &sessions[mainTextStreamIndex]);
- } else
- textStream = text_stream_new_with_sessions(q->getCore()->getCCore()->factory, &sessions[mainTextStreamIndex]);
- if (mediaPorts[mainTextStreamIndex].rtpPort == -1)
- setPortConfigFromRtpSession(mainTextStreamIndex, textStream->ms.sessions.rtp_session);
-
- if (q->getCore()->getCCore()->rtptf) {
- RtpTransport *meta_rtp;
- RtpTransport *meta_rtcp;
- rtp_session_get_transports(textStream->ms.sessions.rtp_session, &meta_rtp, &meta_rtcp);
- if (!meta_rtp_transport_get_endpoint(meta_rtp))
- meta_rtp_transport_set_endpoint(meta_rtp, q->getCore()->getCCore()->rtptf->audio_rtp_func(q->getCore()->getCCore()->rtptf->audio_rtp_func_data, mediaPorts[mainTextStreamIndex].rtpPort));
- if (!meta_rtp_transport_get_endpoint(meta_rtcp))
- meta_rtp_transport_set_endpoint(meta_rtcp, q->getCore()->getCCore()->rtptf->audio_rtcp_func(q->getCore()->getCCore()->rtptf->audio_rtcp_func_data, mediaPorts[mainTextStreamIndex].rtcpPort));
- }
-
- textStreamEvQueue = ortp_ev_queue_new();
- rtp_session_register_event_queue(textStream->ms.sessions.rtp_session, textStreamEvQueue);
- iceAgent->prepareIceForStream(&textStream->ms, false);
-}
-
-void MediaSessionPrivate::initializeVideoStream () {
-#ifdef VIDEO_ENABLED
- L_Q();
- if (videoStream)
- return;
- if (!sessions[mainVideoStreamIndex].rtp_session) {
- SalMulticastRole multicastRole = getMulticastRole(SalVideo);
- SalMediaDescription *remoteDesc = nullptr;
- SalStreamDescription *streamDesc = nullptr;
- if (op)
- remoteDesc = op->getRemoteMediaDescription();
- if (remoteDesc)
- streamDesc = sal_media_description_find_best_stream(remoteDesc, SalVideo);
-
- videoStream = video_stream_new2(q->getCore()->getCCore()->factory, L_STRING_TO_C(getBindIpForStream(mainVideoStreamIndex)),
- (multicastRole == SalMulticastReceiver) ? streamDesc->rtp_port : mediaPorts[mainVideoStreamIndex].rtpPort,
- (multicastRole == SalMulticastReceiver) ? 0 /* Disabled for now */ : mediaPorts[mainVideoStreamIndex].rtcpPort);
- if (multicastRole == SalMulticastReceiver)
- joinMulticastGroup(mainVideoStreamIndex, &videoStream->ms);
-
- configureRtpSession(videoStream->ms.sessions.rtp_session, LinphoneStreamTypeVideo);
- setupDtlsParams(&videoStream->ms);
- /* Initialize zrtp even if we didn't explicitely set it, just in case peer offers it */
- if (linphone_core_media_encryption_supported(q->getCore()->getCCore(), LinphoneMediaEncryptionZRTP))
- video_stream_enable_zrtp(videoStream, audioStream);
-
- media_stream_reclaim_sessions(&videoStream->ms, &sessions[mainVideoStreamIndex]);
- } else
- videoStream = video_stream_new_with_sessions(q->getCore()->getCCore()->factory, &sessions[mainVideoStreamIndex]);
-
- if (mediaPorts[mainVideoStreamIndex].rtpPort == -1)
- setPortConfigFromRtpSession(mainVideoStreamIndex, videoStream->ms.sessions.rtp_session);
- int dscp = linphone_core_get_video_dscp(q->getCore()->getCCore());
- if (dscp!=-1)
- video_stream_set_dscp(videoStream, dscp);
- video_stream_enable_display_filter_auto_rotate(
- videoStream,
- !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "video", "display_filter_auto_rotate", 0)
- );
-
- const char *displayFilter = linphone_core_get_video_display_filter(q->getCore()->getCCore());
- if (displayFilter)
- video_stream_set_display_filter_name(videoStream, displayFilter);
- video_stream_set_event_callback(videoStream, videoStreamEventCb, this);
-
- if (q->getCore()->getCCore()->rtptf) {
- RtpTransport *meta_rtp;
- RtpTransport *meta_rtcp;
- rtp_session_get_transports(videoStream->ms.sessions.rtp_session, &meta_rtp, &meta_rtcp);
- if (!meta_rtp_transport_get_endpoint(meta_rtp)) {
- lInfo() << "CallSession [" << q << "] using custom video RTP transport endpoint";
- meta_rtp_transport_set_endpoint(meta_rtp, q->getCore()->getCCore()->rtptf->video_rtp_func(q->getCore()->getCCore()->rtptf->video_rtp_func_data, mediaPorts[mainVideoStreamIndex].rtpPort));
- }
- if (!meta_rtp_transport_get_endpoint(meta_rtcp))
- meta_rtp_transport_set_endpoint(meta_rtcp, q->getCore()->getCCore()->rtptf->video_rtcp_func(q->getCore()->getCCore()->rtptf->video_rtcp_func_data, mediaPorts[mainVideoStreamIndex].rtcpPort));
- }
- videoStreamEvQueue = ortp_ev_queue_new();
- rtp_session_register_event_queue(videoStream->ms.sessions.rtp_session, videoStreamEvQueue);
- iceAgent->prepareIceForStream(&videoStream->ms, false);
-#ifdef TEST_EXT_RENDERER
- video_stream_set_render_callback(videoStream, extRendererCb, nullptr);
-#endif
-#else
- videoStream = nullptr;
-#endif
-}
-
-void MediaSessionPrivate::prepareEarlyMediaForking () {
- /* We need to disable symmetric rtp otherwise our outgoing streams will be switching permanently between the multiple destinations */
- if (audioStream)
- rtp_session_set_symmetric_rtp(audioStream->ms.sessions.rtp_session, false);
- if (videoStream)
- rtp_session_set_symmetric_rtp(videoStream->ms.sessions.rtp_session, false);
-}
-
-void MediaSessionPrivate::postConfigureAudioStreams (bool muted) {
- L_Q();
- q->getCore()->getPrivate()->postConfigureAudioStream(audioStream, muted);
- forceSpeakerMuted(speakerMuted);
- if (linphone_core_dtmf_received_has_listener(q->getCore()->getCCore()))
- audio_stream_play_received_dtmfs(audioStream, false);
- if (recordActive)
- q->startRecording();
-}
-
-void MediaSessionPrivate::setSymmetricRtp (bool value) {
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- setStreamSymmetricRtp(value, i);
- }
-}
-
-void MediaSessionPrivate::setStreamSymmetricRtp(bool value, int streamIndex) {
- MSMediaStreamSessions *mss = &sessions[streamIndex];
- if (mss->rtp_session)
- rtp_session_set_symmetric_rtp(mss->rtp_session, value);
-}
-
-void MediaSessionPrivate::setupRingbackPlayer () {
- L_Q();
- int pauseTime = 3000;
- audio_stream_play(audioStream, q->getCore()->getCCore()->sound_conf.ringback_tone);
- ms_filter_call_method(audioStream->soundread, MS_FILE_PLAYER_LOOP, &pauseTime);
-}
-
-void MediaSessionPrivate::startAudioStream (CallSession::State targetState) {
- L_Q();
- const SalStreamDescription *stream = sal_media_description_find_best_stream(resultDesc, SalAudio);
- if (stream && (stream->dir != SalStreamInactive) && (stream->rtp_port != 0)) {
- int usedPt = -1;
- onHoldFile = "";
- audioProfile = makeProfile(resultDesc, stream, &usedPt);
- if (usedPt == -1)
- lWarning() << "No audio stream accepted?";
- else {
- const char *rtpAddr = (stream->rtp_addr[0] != '\0') ? stream->rtp_addr : resultDesc->addr;
- bool isMulticast = !!ms_is_multicast(rtpAddr);
- bool ok = true;
- getCurrentParams()->getPrivate()->setUsedAudioCodec(rtp_profile_get_payload(audioProfile, usedPt));
- getCurrentParams()->enableAudio(true);
- MSSndCard *playcard = q->getCore()->getCCore()->sound_conf.lsd_card ? q->getCore()->getCCore()->sound_conf.lsd_card : q->getCore()->getCCore()->sound_conf.play_sndcard;
- if (!playcard)
- lWarning() << "No card defined for playback!";
- MSSndCard *captcard = q->getCore()->getCCore()->sound_conf.capt_sndcard;
- if (!captcard)
- lWarning() << "No card defined for capture!";
- string playfile = L_C_TO_STRING(q->getCore()->getCCore()->play_file);
- string recfile = L_C_TO_STRING(q->getCore()->getCCore()->rec_file);
- /* Don't use file or soundcard capture when placed in recv-only mode */
- if ((stream->rtp_port == 0) || (stream->dir == SalStreamRecvOnly) || ((stream->multicast_role == SalMulticastReceiver) && isMulticast)) {
- captcard = nullptr;
- playfile = "";
- }
- if (targetState == CallSession::State::Paused) {
- // In paused state, we never use soundcard
- playcard = captcard = nullptr;
- recfile = "";
- // And we will eventually play "playfile" if set by the user
- }
- if (listener && listener->isPlayingRingbackTone(q->getSharedFromThis())) {
- captcard = nullptr;
- playfile = ""; /* It is setup later */
- if (lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "send_ringback_without_playback", 0) == 1) {
- playcard = nullptr;
- recfile = "";
- }
- }
- // If playfile are supplied don't use soundcards
- bool useRtpIo = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "rtp_io", false);
- bool useRtpIoEnableLocalOutput = !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "sound", "rtp_io_enable_local_output", false);
- if (q->getCore()->getCCore()->use_files || (useRtpIo && !useRtpIoEnableLocalOutput)) {
- captcard = playcard = nullptr;
- }
- if (getParams()->getPrivate()->getInConference()) {
- // First create the graph without soundcard resources
- captcard = playcard = nullptr;
- }
- if (listener && !listener->areSoundResourcesAvailable(q->getSharedFromThis())) {
- lInfo() << "Sound resources are used by another CallSession, not using soundcard";
- captcard = playcard = nullptr;
- }
-
- if (playcard) {
- ms_snd_card_set_stream_type(playcard, MS_SND_CARD_STREAM_VOICE);
- }
- media_stream_set_max_network_bitrate(&audioStream->ms, linphone_core_get_upload_bandwidth(q->getCore()->getCCore()) * 1000);
- bool useEc = captcard && linphone_core_echo_cancellation_enabled(q->getCore()->getCCore());
- audio_stream_enable_echo_canceller(audioStream, useEc);
- if (playcard && (stream->max_rate > 0))
- ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
- if (captcard && (stream->max_rate > 0))
- ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
- rtp_session_enable_rtcp_mux(audioStream->ms.sessions.rtp_session, stream->rtcp_mux);
- if (!getParams()->getPrivate()->getInConference() && !getParams()->getRecordFilePath().empty()) {
- audio_stream_mixed_record_open(audioStream, getParams()->getRecordFilePath().c_str());
- getCurrentParams()->setRecordFilePath(getParams()->getRecordFilePath());
- }
- // Valid local tags are > 0
- if (sal_stream_description_has_srtp(stream)) {
- const SalStreamDescription *localStreamDesc = sal_media_description_find_stream(localDesc, stream->proto, SalAudio);
- int cryptoIdx = Sal::findCryptoIndexFromTag(localStreamDesc->crypto, static_cast<unsigned char>(stream->crypto_local_tag));
- if (cryptoIdx >= 0) {
- ms_media_stream_sessions_set_srtp_recv_key_b64(&audioStream->ms.sessions, stream->crypto[0].algo, stream->crypto[0].master_key);
- ms_media_stream_sessions_set_srtp_send_key_b64(&audioStream->ms.sessions, stream->crypto[0].algo, localStreamDesc->crypto[cryptoIdx].master_key);
- } else
- lWarning() << "Failed to find local crypto algo with tag: " << stream->crypto_local_tag;
- }
- configureRtpSessionForRtcpFb(stream);
- configureRtpSessionForRtcpXr(SalAudio);
- bool videoWillBeUsed = false;
-#if defined(VIDEO_ENABLED)
- const SalStreamDescription *vstream = sal_media_description_find_best_stream(resultDesc, SalVideo);
- if (vstream && (vstream->dir != SalStreamInactive) && vstream->payloads) {
- /* When video is used, do not make adaptive rate control on audio, it is stupid */
- videoWillBeUsed = true;
- }
-#endif
- configureAdaptiveRateControl(&audioStream->ms, getCurrentParams()->getUsedAudioCodec(), videoWillBeUsed);
- if (isMulticast)
- rtp_session_set_multicast_ttl(audioStream->ms.sessions.rtp_session, stream->ttl);
- MSMediaStreamIO io = MS_MEDIA_STREAM_IO_INITIALIZER;
- if (useRtpIo) {
- if (useRtpIoEnableLocalOutput) {
- io.input.type = MSResourceRtp;
- io.input.session = createAudioRtpIoSession();
- if (playcard) {
- io.output.type = MSResourceSoundcard;
- io.output.soundcard = playcard;
- } else {
- io.output.type = MSResourceFile;
- io.output.file = recfile.empty() ? nullptr : recfile.c_str();
- }
- } else {
- io.input.type = io.output.type = MSResourceRtp;
- io.input.session = io.output.session = createAudioRtpIoSession();
- }
- if (!io.input.session)
- ok = false;
- } else {
- if (playcard) {
- io.output.type = MSResourceSoundcard;
- io.output.soundcard = playcard;
- } else {
- io.output.type = MSResourceFile;
- io.output.file = recfile.empty() ? nullptr : recfile.c_str();
- }
- if (captcard) {
- io.input.type = MSResourceSoundcard;
- io.input.soundcard = captcard;
- } else {
- io.input.type = MSResourceFile;
- onHoldFile = playfile;
- io.input.file = nullptr; /* We prefer to use the remote_play api, that allows to play multimedia files */
- }
- }
- if (ok) {
- currentCaptureCard = ms_media_resource_get_soundcard(&io.input);
- currentPlayCard = ms_media_resource_get_soundcard(&io.output);
-
- int err = audio_stream_start_from_io(audioStream, audioProfile, rtpAddr, stream->rtp_port,
- (stream->rtcp_addr[0] != '\0') ? stream->rtcp_addr : resultDesc->addr,
- (linphone_core_rtcp_enabled(q->getCore()->getCCore()) && !isMulticast) ? (stream->rtcp_port ? stream->rtcp_port : stream->rtp_port + 1) : 0,
- usedPt, &io);
- if (err == 0)
- postConfigureAudioStreams((audioMuted || microphoneMuted) && (listener && !listener->isPlayingRingbackTone(q->getSharedFromThis())));
- }
- ms_media_stream_sessions_set_encryption_mandatory(&audioStream->ms.sessions, isEncryptionMandatory());
- if ((targetState == CallSession::State::Paused) && !captcard && !playfile.empty()) {
- int pauseTime = 500;
- ms_filter_call_method(audioStream->soundread, MS_FILE_PLAYER_LOOP, &pauseTime);
- }
- if (listener && listener->isPlayingRingbackTone(q->getSharedFromThis()))
- setupRingbackPlayer();
- if (getParams()->getPrivate()->getInConference() && listener) {
- // Transform the graph to connect it to the conference filter
- bool mute = (stream->dir == SalStreamRecvOnly);
- listener->onCallSessionConferenceStreamStarting(q->getSharedFromThis(), mute);
- }
- getCurrentParams()->getPrivate()->setInConference(getParams()->getPrivate()->getInConference());
- getCurrentParams()->enableLowBandwidth(getParams()->lowBandwidthEnabled());
- // Start ZRTP engine if needed : set here or remote have a zrtp-hash attribute
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- const SalStreamDescription *remoteStream = sal_media_description_find_best_stream(remote, SalAudio);
- if (linphone_core_media_encryption_supported(q->getCore()->getCCore(), LinphoneMediaEncryptionZRTP)) {
- // Perform mutual authentication if instant messaging encryption is enabled
- auto encryptionEngine = q->getCore()->getEncryptionEngine();
- //Is call direction really relevant ? might be linked to offerer/answerer rather than call direction ?
- LinphoneCallDir direction = this->getPublic()->CallSession::getDirection();
- if (encryptionEngine && audioStream->ms.sessions.zrtp_context) {
- encryptionEngine->mutualAuthentication(
- audioStream->ms.sessions.zrtp_context,
- op->getLocalMediaDescription(),
- op->getRemoteMediaDescription(),
- direction
- );
- }
-
- //Start zrtp if remote has offered it or if local is configured for zrtp and is the offerrer. If not, defered when ACK is received
- if ((getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP && op->isOfferer()) || (remoteStream->haveZrtpHash == 1)) {
- startZrtpPrimaryChannel(remoteStream);
- }
- }
- }
- }
-}
-
-void MediaSessionPrivate::startStreams (CallSession::State targetState) {
- L_Q();
- switch (targetState) {
- case CallSession::State::IncomingEarlyMedia:
- if (listener)
- listener->onRingbackToneRequested(q->getSharedFromThis(), true);
- BCTBX_NO_BREAK;
- case CallSession::State::OutgoingEarlyMedia:
- if (!getParams()->earlyMediaSendingEnabled()) {
- audioMuted = true;
- videoMuted = true;
- }
- break;
- default:
- if (listener)
- listener->onRingbackToneRequested(q->getSharedFromThis(), false);
- audioMuted = false;
- videoMuted = false;
- break;
- }
-
- getCurrentParams()->getPrivate()->setUsedAudioCodec(nullptr);
- getCurrentParams()->getPrivate()->setUsedVideoCodec(nullptr);
- getCurrentParams()->getPrivate()->setUsedRealtimeTextCodec(nullptr);
-
- if (!audioStream && !videoStream) {
- lFatal() << "startStreams() called without prior init!";
- return;
- }
- if (iceAgent->hasSession()) {
- /* If there is an ICE session when we are about to start streams, then ICE will conduct the media path checking and authentication properly.
- * Symmetric RTP must be turned off */
- setSymmetricRtp(false);
- }
-
- if (audioStream) audioStartCount++;
- if (videoStream) videoStartCount++;
- if (textStream) textStartCount++;
-
- lInfo() << "startStreams() CallSession=[" << q << "] local upload_bandwidth=[" << linphone_core_get_upload_bandwidth(q->getCore()->getCCore())
- << "] kbit/s; local download_bandwidth=[" << linphone_core_get_download_bandwidth(q->getCore()->getCCore()) << "] kbit/s";
- getCurrentParams()->enableAudio(false);
- if (audioStream)
- startAudioStream(targetState);
- else
- lWarning() << "startStreams(): no audio stream!";
- getCurrentParams()->enableVideo(false);
- if (videoStream) {
- if (audioStream)
- audio_stream_link_video(audioStream, videoStream);
- startVideoStream(targetState);
- }
- /* The on-hold file is to be played once both audio and video are ready */
- if (!onHoldFile.empty() && !getParams()->getPrivate()->getInConference() && audioStream) {
- MSFilter *player = audio_stream_open_remote_play(audioStream, onHoldFile.c_str());
- if (player) {
- int pauseTime = 500;
- ms_filter_call_method(player, MS_PLAYER_SET_LOOP, &pauseTime);
- ms_filter_call_method_noarg(player, MS_PLAYER_START);
- }
- }
- if (getParams()->realtimeTextEnabled())
- startTextStream();
-
- setDtlsFingerprintOnAllStreams();
- if (!iceAgent->hasCompleted()) {
- iceAgent->startConnectivityChecks();
- } else {
- /* Should not start dtls until ice is completed */
- startDtlsOnAllStreams();
- }
-}
-
-void MediaSessionPrivate::startStream (SalStreamDescription *streamDesc, int streamIndex, CallSession::State targetState) {
- L_Q();
- string streamTypeName = sal_stream_description_get_type_as_string(streamDesc);
-
- if (streamDesc->type == SalAudio) {
- if (audioStream && audioStream->ms.state != MSStreamInitialized)
- audio_stream_unprepare_sound(audioStream);
-
- switch (targetState) {
- case CallSession::State::IncomingEarlyMedia:
- if (listener)
- listener->onRingbackToneRequested(q->getSharedFromThis(), true);
- BCTBX_NO_BREAK;
- case CallSession::State::OutgoingEarlyMedia:
- if (!getParams()->earlyMediaSendingEnabled())
- audioMuted = true;
- break;
- default:
- if (listener)
- listener->onRingbackToneRequested(q->getSharedFromThis(), false);
- audioMuted = false;
- break;
- }
-
- getCurrentParams()->getPrivate()->setUsedAudioCodec(nullptr);
-
- if (!audioStream) {
- lFatal() << "startStream() for audio stream called without prior init!";
- return;
- }
- } else if (streamDesc->type == SalVideo) {
-#ifdef VIDEO_ENABLED
- if (videoStream && videoStream->ms.state != MSStreamInitialized)
- video_stream_unprepare_video(videoStream);
-
- switch (targetState) {
- case CallSession::State::OutgoingEarlyMedia:
- if (!getParams()->earlyMediaSendingEnabled())
- videoMuted = true;
- break;
- default:
- videoMuted = false;
- break;
- }
-
- getCurrentParams()->getPrivate()->setUsedVideoCodec(nullptr);
-
- if (!videoStream) {
- lFatal() << "startStream() for video stream called without prior init!";
- return;
- }
-#endif
- } else if (streamDesc->type == SalText) {
- if (textStream && textStream->ms.state != MSStreamInitialized)
- text_stream_unprepare_text(textStream);
-
- getCurrentParams()->getPrivate()->setUsedRealtimeTextCodec(nullptr);
- }
-
- if (iceAgent->hasSession()) {
- /* If there is an ICE session when we are about to start streams, then ICE will conduct the media path checking and authentication properly.
- * Symmetric RTP must be turned off */
- setStreamSymmetricRtp(false, streamIndex);
- }
-
- lInfo() << "startStream() for " << streamTypeName << " stream CallSession=[" << q << "] local upload_bandwidth=[" << linphone_core_get_upload_bandwidth(q->getCore()->getCCore())
- << "] kbit/s; local download_bandwidth=[" << linphone_core_get_download_bandwidth(q->getCore()->getCCore()) << "] kbit/s";
-
- if (streamDesc->type == SalAudio) {
- audioStartCount++;
-
- getCurrentParams()->enableAudio(false);
- if (audioStream)
- startAudioStream(targetState);
- else
- lWarning() << "startStreams(): no audio stream!";
-
- postProcessHooks.push_back([this] {
- /* The on-hold file is to be played once both audio and video are ready */
- if (!onHoldFile.empty() && !getParams()->getPrivate()->getInConference() && audioStream) {
- MSFilter *player = audio_stream_open_remote_play(audioStream, onHoldFile.c_str());
- if (player) {
- int pauseTime = 500;
- ms_filter_call_method(player, MS_PLAYER_SET_LOOP, &pauseTime);
- ms_filter_call_method_noarg(player, MS_PLAYER_START);
- }
- }
- });
-
- setDtlsFingerprintOnAudioStream();
- if (iceAgent->hasCompleted())
- startDtlsOnAudioStream();
- } else if (streamDesc->type == SalVideo) {
- videoStartCount++;
-
- getCurrentParams()->enableVideo(false);
- if (videoStream) {
- if (audioStream)
- audio_stream_link_video(audioStream, videoStream);
- startVideoStream(targetState);
- }
-
- setDtlsFingerprintOnVideoStream();
- if (iceAgent->hasCompleted())
- startDtlsOnVideoStream();
- } else if (streamDesc->type == SalText) {
- textStartCount++;
-
- if (getParams()->realtimeTextEnabled())
- startTextStream();
-
- setDtlsFingerprintOnTextStream();
- if (iceAgent->hasCompleted())
- startDtlsOnTextStream();
- }
-}
-
-void MediaSessionPrivate::startTextStream () {
- L_Q();
- const SalStreamDescription *tstream = sal_media_description_find_best_stream(resultDesc, SalText);
- if (tstream && (tstream->dir != SalStreamInactive) && (tstream->rtp_port != 0)) {
- const char *rtpAddr = tstream->rtp_addr[0] != '\0' ? tstream->rtp_addr : resultDesc->addr;
- const char *rtcpAddr = tstream->rtcp_addr[0] != '\0' ? tstream->rtcp_addr : resultDesc->addr;
- const SalStreamDescription *localStreamDesc = sal_media_description_find_stream(localDesc, tstream->proto, SalText);
- int usedPt = -1;
- textProfile = makeProfile(resultDesc, tstream, &usedPt);
- if (usedPt == -1)
- lWarning() << "No text stream accepted";
- else {
- getCurrentParams()->getPrivate()->setUsedRealtimeTextCodec(rtp_profile_get_payload(textProfile, usedPt));
- getCurrentParams()->enableRealtimeText(true);
- unsigned int interval = getParams()->realtimeTextKeepaliveInterval();
- getCurrentParams()->setRealtimeTextKeepaliveInterval(interval);
- if (sal_stream_description_has_srtp(tstream)) {
- int cryptoIdx = Sal::findCryptoIndexFromTag(localStreamDesc->crypto, static_cast<unsigned char>(tstream->crypto_local_tag));
- if (cryptoIdx >= 0) {
- ms_media_stream_sessions_set_srtp_recv_key_b64(&textStream->ms.sessions, tstream->crypto[0].algo, tstream->crypto[0].master_key);
- ms_media_stream_sessions_set_srtp_send_key_b64(&textStream->ms.sessions, tstream->crypto[0].algo, localStreamDesc->crypto[cryptoIdx].master_key);
- }
- }
- configureRtpSessionForRtcpFb(tstream);
- configureRtpSessionForRtcpXr(SalText);
- rtp_session_enable_rtcp_mux(textStream->ms.sessions.rtp_session, tstream->rtcp_mux);
- bool isMulticast = !!ms_is_multicast(rtpAddr);
- if (isMulticast)
- rtp_session_set_multicast_ttl(textStream->ms.sessions.rtp_session, tstream->ttl);
- text_stream_start(textStream, textProfile, rtpAddr, tstream->rtp_port, rtcpAddr,
- (linphone_core_rtcp_enabled(q->getCore()->getCCore()) && !isMulticast) ? (tstream->rtcp_port ? tstream->rtcp_port : tstream->rtp_port + 1) : 0, usedPt);
- ms_filter_call_method(textStream->rttsource, MS_RTT_4103_SOURCE_SET_KEEP_ALIVE_INTERVAL, &interval);
- ms_filter_add_notify_callback(textStream->rttsink, realTimeTextCharacterReceived, this, false);
- ms_media_stream_sessions_set_encryption_mandatory(&textStream->ms.sessions, isEncryptionMandatory());
- }
- } else
- lInfo() << "No valid text stream defined";
-}
-
-void MediaSessionPrivate::startVideoStream (CallSession::State targetState) {
-#ifdef VIDEO_ENABLED
- L_Q();
- bool reusedPreview = false;
- /* Shutdown preview */
- MSFilter *source = nullptr;
- if (q->getCore()->getCCore()->previewstream) {
- if (q->getCore()->getCCore()->video_conf.reuse_preview_source)
- source = video_preview_stop_reuse_source(q->getCore()->getCCore()->previewstream);
- else
- video_preview_stop(q->getCore()->getCCore()->previewstream);
- q->getCore()->getCCore()->previewstream = nullptr;
- }
- const SalStreamDescription *vstream = sal_media_description_find_best_stream(resultDesc, SalVideo);
- if (vstream && (vstream->dir != SalStreamInactive) && (vstream->rtp_port != 0)) {
- int usedPt = -1;
- videoProfile = makeProfile(resultDesc, vstream, &usedPt);
- if (usedPt == -1)
- lWarning() << "No video stream accepted";
- else {
- getCurrentParams()->getPrivate()->setUsedVideoCodec(rtp_profile_get_payload(videoProfile, usedPt));
- getCurrentParams()->enableVideo(true);
- rtp_session_enable_rtcp_mux(videoStream->ms.sessions.rtp_session, vstream->rtcp_mux);
- media_stream_set_max_network_bitrate(&videoStream->ms, linphone_core_get_upload_bandwidth(q->getCore()->getCCore()) * 1000);
- if (q->getCore()->getCCore()->video_conf.preview_vsize.width != 0)
- video_stream_set_preview_size(videoStream, q->getCore()->getCCore()->video_conf.preview_vsize);
- video_stream_set_fps(videoStream, linphone_core_get_preferred_framerate(q->getCore()->getCCore()));
- if (lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "video", "nowebcam_uses_normal_fps", 0))
- videoStream->staticimage_webcam_fps_optimization = false;
- const LinphoneVideoDefinition *vdef = linphone_core_get_preferred_video_definition(q->getCore()->getCCore());
- MSVideoSize vsize;
- vsize.width = static_cast<int>(linphone_video_definition_get_width(vdef));
- vsize.height = static_cast<int>(linphone_video_definition_get_height(vdef));
- video_stream_set_sent_video_size(videoStream, vsize);
- video_stream_enable_self_view(videoStream, q->getCore()->getCCore()->video_conf.selfview);
- if (videoWindowId)
- video_stream_set_native_window_id(videoStream, videoWindowId);
- else if (q->getCore()->getCCore()->video_window_id)
- video_stream_set_native_window_id(videoStream, q->getCore()->getCCore()->video_window_id);
- if (q->getCore()->getCCore()->preview_window_id)
- video_stream_set_native_preview_window_id(videoStream, q->getCore()->getCCore()->preview_window_id);
- video_stream_use_preview_video_window(videoStream, q->getCore()->getCCore()->use_preview_window);
- const char *rtpAddr = (vstream->rtp_addr[0] != '\0') ? vstream->rtp_addr : resultDesc->addr;
- const char *rtcpAddr = (vstream->rtcp_addr[0] != '\0') ? vstream->rtcp_addr : resultDesc->addr;
- bool isMulticast = !!ms_is_multicast(rtpAddr);
- MediaStreamDir dir = MediaStreamSendRecv;
- bool isActive = true;
- if (isMulticast) {
- if (vstream->multicast_role == SalMulticastReceiver)
- dir = MediaStreamRecvOnly;
- else
- dir = MediaStreamSendOnly;
- } else if ((vstream->dir == SalStreamSendOnly) && q->getCore()->getCCore()->video_conf.capture)
- dir = MediaStreamSendOnly;
- else if ((vstream->dir == SalStreamRecvOnly) && q->getCore()->getCCore()->video_conf.display)
- dir = MediaStreamRecvOnly;
- else if (vstream->dir == SalStreamSendRecv) {
- if (q->getCore()->getCCore()->video_conf.display && q->getCore()->getCCore()->video_conf.capture)
- dir = MediaStreamSendRecv;
- else if (q->getCore()->getCCore()->video_conf.display)
- dir = MediaStreamRecvOnly;
- else
- dir = MediaStreamSendOnly;
- } else {
- lWarning() << "Video stream is inactive";
- /* Either inactive or incompatible with local capabilities */
- isActive = false;
- }
- MSWebCam *cam = getVideoDevice();
- if (isActive) {
- if (sal_stream_description_has_srtp(vstream)) {
- const SalStreamDescription *localStreamDesc = sal_media_description_find_stream(localDesc, vstream->proto, SalVideo);
- int cryptoIdx = Sal::findCryptoIndexFromTag(localStreamDesc->crypto, static_cast<unsigned char>(vstream->crypto_local_tag));
- if (cryptoIdx >= 0) {
- ms_media_stream_sessions_set_srtp_recv_key_b64(&videoStream->ms.sessions, vstream->crypto[0].algo, vstream->crypto[0].master_key);
- ms_media_stream_sessions_set_srtp_send_key_b64(&videoStream->ms.sessions, vstream->crypto[0].algo, localStreamDesc->crypto[cryptoIdx].master_key);
- }
- }
- configureRtpSessionForRtcpFb(vstream);
- configureRtpSessionForRtcpXr(SalVideo);
- configureAdaptiveRateControl(&videoStream->ms, getCurrentParams()->getUsedVideoCodec(), true);
- log->video_enabled = true;
- video_stream_set_direction(videoStream, dir);
- lInfo() << "startVideoStream: device_rotation=" << q->getCore()->getCCore()->device_rotation;
- video_stream_set_device_rotation(videoStream, q->getCore()->getCCore()->device_rotation);
- video_stream_set_freeze_on_error(videoStream, !!lp_config_get_int(linphone_core_get_config(q->getCore()->getCCore()), "video", "freeze_on_error", 1));
- if (isMulticast)
- rtp_session_set_multicast_ttl(videoStream->ms.sessions.rtp_session, vstream->ttl);
- video_stream_use_video_preset(videoStream, lp_config_get_string(linphone_core_get_config(q->getCore()->getCCore()), "video", "preset", nullptr));
- if (q->getCore()->getCCore()->video_conf.reuse_preview_source && source) {
- lInfo() << "video_stream_start_with_source kept: " << source;
- video_stream_start_with_source(videoStream, videoProfile, rtpAddr, vstream->rtp_port, rtcpAddr,
- linphone_core_rtcp_enabled(q->getCore()->getCCore()) ? (vstream->rtcp_port ? vstream->rtcp_port : vstream->rtp_port + 1) : 0,
- usedPt, -1, cam, source);
- reusedPreview = true;
- } else {
- bool ok = true;
- MSMediaStreamIO io = MS_MEDIA_STREAM_IO_INITIALIZER;
- if (linphone_config_get_bool(linphone_core_get_config(q->getCore()->getCCore()), "video", "rtp_io", FALSE)) {
- io.input.type = io.output.type = MSResourceRtp;
- io.input.session = io.output.session = createVideoRtpIoSession();
- if (!io.input.session) {
- ok = false;
- lWarning() << "Cannot create video RTP IO session";
- }
- } else {
- io.input.type = MSResourceCamera;
- io.input.camera = cam;
- io.output.type = MSResourceDefault;
- }
- if (ok) {
- video_stream_start_from_io(videoStream, videoProfile, rtpAddr, vstream->rtp_port, rtcpAddr,
- (linphone_core_rtcp_enabled(q->getCore()->getCCore()) && !isMulticast) ? (vstream->rtcp_port ? vstream->rtcp_port : vstream->rtp_port + 1) : 0,
- usedPt, &io);
- }
- }
-
- ms_media_stream_sessions_set_encryption_mandatory(&videoStream->ms.sessions, isEncryptionMandatory());
- if (listener)
- listener->onResetFirstVideoFrameDecoded(q->getSharedFromThis());
- /* Start ZRTP engine if needed : set here or remote have a zrtp-hash attribute */
- SalMediaDescription *remote = op->getRemoteMediaDescription();
- const SalStreamDescription *remoteStream = sal_media_description_find_best_stream(remote, SalVideo);
- if ((getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP) || (remoteStream->haveZrtpHash == 1)) {
- /* Audio stream is already encrypted and video stream is active */
- if (media_stream_secured(&audioStream->ms) && (media_stream_get_state(&videoStream->ms) == MSStreamStarted)) {
- video_stream_start_zrtp(videoStream);
- if (remoteStream->haveZrtpHash == 1) {
- int retval = ms_zrtp_setPeerHelloHash(videoStream->ms.sessions.zrtp_context, (uint8_t *)remoteStream->zrtphash, strlen((const char *)(remoteStream->zrtphash)));
- if (retval != 0)
- lError() << "Video stream ZRTP hash mismatch 0x" << hex << retval;
- }
- }
- }
-
- if (linphone_core_retransmission_on_nack_enabled(q->getCore()->getCCore())) {
- video_stream_enable_retransmission_on_nack(videoStream, TRUE);
- }
- }
- }
- } else
- lInfo() << "No valid video stream defined";
- if (!reusedPreview && source) {
- /* Destroy not-reused source filter */
- lWarning() << "Video preview (" << source << ") not reused: destroying it";
- ms_filter_destroy(source);
- }
-#endif
-}
-
-void MediaSessionPrivate::stopAudioStream () {
- L_Q();
- if (!audioStream)
- return;
-
- if (listener)
- listener->onUpdateMediaInfoForReporting(q->getSharedFromThis(), LINPHONE_CALL_STATS_AUDIO);
- media_stream_reclaim_sessions(&audioStream->ms, &sessions[mainAudioStreamIndex]);
- if (audioStream->ec) {
- char *stateStr = nullptr;
- ms_filter_call_method(audioStream->ec, MS_ECHO_CANCELLER_GET_STATE_STRING, &stateStr);
- if (stateStr) {
- lInfo() << "Writing echo canceler state, " << (int)strlen(stateStr) << " bytes";
- lp_config_write_relative_file(linphone_core_get_config(q->getCore()->getCCore()), ecStateStore.c_str(), stateStr);
- }
- }
- audio_stream_get_local_rtp_stats(audioStream, &log->local_stats);
- fillLogStats(&audioStream->ms);
- if (listener)
- listener->onCallSessionConferenceStreamStopping(q->getSharedFromThis());
- ms_bandwidth_controller_remove_stream(q->getCore()->getCCore()->bw_controller, &audioStream->ms);
- audio_stream_stop(audioStream);
- updateRtpStats(audioStats, mainAudioStreamIndex);
- audioStream = nullptr;
- handleStreamEvents(mainAudioStreamIndex);
- rtp_session_unregister_event_queue(sessions[mainAudioStreamIndex].rtp_session, audioStreamEvQueue);
- ortp_ev_queue_flush(audioStreamEvQueue);
- ortp_ev_queue_destroy(audioStreamEvQueue);
- audioStreamEvQueue = nullptr;
-
- getCurrentParams()->getPrivate()->setUsedAudioCodec(nullptr);
-
- currentCaptureCard = nullptr;
- currentPlayCard = nullptr;
-
-}
-
-void MediaSessionPrivate::stopTextStream () {
- L_Q();
- if (textStream) {
- if (listener)
- listener->onUpdateMediaInfoForReporting(q->getSharedFromThis(), LINPHONE_CALL_STATS_TEXT);
- media_stream_reclaim_sessions(&textStream->ms, &sessions[mainTextStreamIndex]);
- fillLogStats(&textStream->ms);
- text_stream_stop(textStream);
- updateRtpStats(textStats, mainTextStreamIndex);
- textStream = nullptr;
- handleStreamEvents(mainTextStreamIndex);
- rtp_session_unregister_event_queue(sessions[mainTextStreamIndex].rtp_session, textStreamEvQueue);
- ortp_ev_queue_flush(textStreamEvQueue);
- ortp_ev_queue_destroy(textStreamEvQueue);
- textStreamEvQueue = nullptr;
- getCurrentParams()->getPrivate()->setUsedRealtimeTextCodec(nullptr);
- }
-}
-
-void MediaSessionPrivate::stopVideoStream () {
-#ifdef VIDEO_ENABLED
- L_Q();
- if (videoStream) {
- if (listener)
- listener->onUpdateMediaInfoForReporting(q->getSharedFromThis(), LINPHONE_CALL_STATS_VIDEO);
- media_stream_reclaim_sessions(&videoStream->ms, &sessions[mainVideoStreamIndex]);
- fillLogStats(&videoStream->ms);
- ms_bandwidth_controller_remove_stream(q->getCore()->getCCore()->bw_controller, &videoStream->ms);
- video_stream_stop(videoStream);
- updateRtpStats(videoStats, mainVideoStreamIndex);
- videoStream = nullptr;
- handleStreamEvents(mainVideoStreamIndex);
- rtp_session_unregister_event_queue(sessions[mainVideoStreamIndex].rtp_session, videoStreamEvQueue);
- ortp_ev_queue_flush(videoStreamEvQueue);
- ortp_ev_queue_destroy(videoStreamEvQueue);
- videoStreamEvQueue = nullptr;
- getCurrentParams()->getPrivate()->setUsedVideoCodec(nullptr);
- }
-#endif
-}
-
-void MediaSessionPrivate::tryEarlyMediaForking (SalMediaDescription *md) {
- L_Q();
- lInfo() << "Early media response received from another branch, checking if media can be forked to this new destination";
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&resultDesc->streams[i]))
- continue;
- SalStreamDescription *refStream = &resultDesc->streams[i];
- SalStreamDescription *newStream = &md->streams[i];
- if ((refStream->type == newStream->type) && refStream->payloads && newStream->payloads) {
- OrtpPayloadType *refpt = reinterpret_cast<OrtpPayloadType *>(refStream->payloads->data);
- OrtpPayloadType *newpt = reinterpret_cast<OrtpPayloadType *>(newStream->payloads->data);
- if ((strcmp(refpt->mime_type, newpt->mime_type) == 0) && (refpt->clock_rate == newpt->clock_rate)
- && (payload_type_get_number(refpt) == payload_type_get_number(newpt))) {
- MediaStream *ms = nullptr;
- if (refStream->type == SalAudio)
- ms = &audioStream->ms;
- else if (refStream->type == SalVideo)
- ms = &videoStream->ms;
- if (ms) {
- RtpSession *session = ms->sessions.rtp_session;
- const char *rtpAddr = (newStream->rtp_addr[0] != '\0') ? newStream->rtp_addr : md->addr;
- const char *rtcpAddr = (newStream->rtcp_addr[0] != '\0') ? newStream->rtcp_addr : md->addr;
- if (ms_is_multicast(rtpAddr))
- lInfo() << "Multicast addr [" << rtpAddr << "/" << newStream->rtp_port << "] does not need auxiliary rtp's destination for CallSession [" << q << "]";
- else
- rtp_session_add_aux_remote_addr_full(session, rtpAddr, newStream->rtp_port, rtcpAddr, newStream->rtcp_port);
- }
- }
- }
- }
-}
-
-void MediaSessionPrivate::updateStreamFrozenPayloads (SalStreamDescription *resultDesc, SalStreamDescription *localStreamDesc) {
- L_Q();
- for (bctbx_list_t *elem = resultDesc->payloads; elem != nullptr; elem = bctbx_list_next(elem)) {
- OrtpPayloadType *pt = reinterpret_cast<OrtpPayloadType *>(bctbx_list_get_data(elem));
- if (PayloadTypeHandler::isPayloadTypeNumberAvailable(localStreamDesc->already_assigned_payloads, payload_type_get_number(pt), nullptr)) {
- /* New codec, needs to be added to the list */
- localStreamDesc->already_assigned_payloads = bctbx_list_append(localStreamDesc->already_assigned_payloads, payload_type_clone(pt));
- lInfo() << "CallSession[" << q << "] : payload type " << payload_type_get_number(pt) << " " << pt->mime_type << "/" << pt->clock_rate
- << " fmtp=" << L_C_TO_STRING(pt->recv_fmtp) << " added to frozen list";
- }
- }
-}
-
-void MediaSessionPrivate::updateFrozenPayloads (SalMediaDescription *result) {
- for (int i = 0; i < result->nb_streams; i++) {
- updateStreamFrozenPayloads(&result->streams[i], &localDesc->streams[i]);
- }
-}
-
-void MediaSessionPrivate::updateStreams (SalMediaDescription *newMd, CallSession::State targetState) {
- L_Q();
-
- if (state == CallSession::State::Connected || state == CallSession::State::Resuming ||
- (state == CallSession::State::IncomingEarlyMedia && !linphone_core_get_ring_during_incoming_early_media(q->getCore()->getCCore()))) {
- q->getCore()->getPrivate()->getToneManager()->goToCall(q->getSharedFromThis());
- }
-
- if (!newMd) {
- lError() << "updateStreams() called with null media description";
- return;
- }
-
- updateBiggestDesc(localDesc);
- sal_media_description_ref(newMd);
- SalMediaDescription *oldMd = resultDesc;
- resultDesc = newMd;
-
- if (getParams()->getMediaEncryption() == LinphoneMediaEncryptionDTLS) {
- getCurrentParams()->getPrivate()->setUpdateCallWhenIceCompleted(false);
- lInfo() << "Disabling update call when ice completed on call [" << q << "]";
- }
-
- if ((audioStream && (audioStream->ms.state == MSStreamStarted)) || (videoStream && (videoStream->ms.state == MSStreamStarted))) {
- clearEarlyMediaDestinations();
-
- /* We already started media: check if we really need to restart it */
- int mdChanged = 0;
- if (oldMd) {
- mdChanged = mediaParametersChanged(oldMd, newMd);
- /* Might not be mandatory to restart stream for each ice restart as it leads bad user experience, specially in video. See 0002495 for better background on this */
- if (mdChanged & (SAL_MEDIA_DESCRIPTION_CODEC_CHANGED
- | SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED
- | SAL_MEDIA_DESCRIPTION_ICE_RESTART_DETECTED
- | SAL_MEDIA_DESCRIPTION_FORCE_STREAM_RECONSTRUCTION)
- ) {
- lInfo() << "Media descriptions are different, need to restart the streams";
- } else {
- for(int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; ++i) {
- if (!sal_stream_description_active(&oldMd->streams[i]) && !sal_stream_description_active(&newMd->streams[i])) continue;
- string streamTypeName = sal_stream_description_get_type_as_string(&newMd->streams[i]);
-
- /* If there was a change in the streams then newMd should have more streams */
- if (mdChanged & SAL_MEDIA_DESCRIPTION_STREAMS_CHANGED && i >= oldMd->nb_streams) {
- lInfo() << "New " << streamTypeName << " stream detected, starting the stream";
-
- if (newMd->streams[i].type == SalAudio) {
- initializeAudioStream();
- } else if (newMd->streams[i].type == SalVideo) {
- initializeVideoStream();
- } else if (newMd->streams[i].type == SalText) {
- initializeTextStream();
- }
-
- if (getParams()->earlyMediaSendingEnabled() && (state == CallSession::State::OutgoingEarlyMedia)) {
- if (newMd->streams[i].type == SalAudio && audioStream)
- rtp_session_set_symmetric_rtp(audioStream->ms.sessions.rtp_session, false);
- else if (newMd->streams[i].type == SalVideo && videoStream)
- rtp_session_set_symmetric_rtp(videoStream->ms.sessions.rtp_session, false);
- }
-
- startStream(&newMd->streams[i], i, targetState);
-
- if (newMd->streams[i].type == SalAudio && audioStream) {
- if ((state == CallSession::State::Pausing) && pausedByApp && (q->getCore()->getCallCount() == 1))
- q->getCore()->getPrivate()->getToneManager()->startNamedTone(q->getSharedFromThis(), LinphoneToneCallOnHold);
- }
-
- updateStreamFrozenPayloads(&newMd->streams[i], &localDesc->streams[i]);
-
- continue;
- }
-
- int sdChanged = sal_stream_description_equals(&oldMd->streams[i], &newMd->streams[i]);
-
- if (newMd->streams[i].type == SalAudio && listener && listener->isPlayingRingbackTone(q->getSharedFromThis())) {
- lInfo() << "Playing ringback tone, will restart the audio stream";
- sdChanged |= SAL_MEDIA_DESCRIPTION_FORCE_STREAM_RECONSTRUCTION;
- }
-
- if (sdChanged & (SAL_MEDIA_DESCRIPTION_CODEC_CHANGED
- | SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED
- | SAL_MEDIA_DESCRIPTION_ICE_RESTART_DETECTED
- | SAL_MEDIA_DESCRIPTION_FORCE_STREAM_RECONSTRUCTION)
- ) {
- lInfo() << "Stream descriptions are different, need to restart the " << streamTypeName << " stream";
- restartStream(&newMd->streams[i], i, sdChanged, targetState);
- } else {
- sdChanged |= mdChanged;
-
- if (newMd->streams[i].type == SalAudio && audioMuted && (targetState == CallSession::State::StreamsRunning)) {
- lInfo() << "Early media finished, unmuting audio input...";
- /* We were in early media, now we want to enable real media */
- audioMuted = false;
-
- if (audioStream) {
- linphone_core_enable_mic(q->getCore()->getCCore(), linphone_core_mic_enabled(q->getCore()->getCCore()));
- }
- }
-
-#ifdef VIDEO_ENABLED
- if (newMd->streams[i].type == SalVideo && videoMuted && (targetState == CallSession::State::StreamsRunning)) {
- lInfo() << "Early media finished, unmuting video input...";
- /* We were in early media, now we want to enable real media */
- videoMuted = false;
-
- if (videoStream && cameraEnabled) {
- q->enableCamera(q->cameraEnabled());
- }
- }
-#endif
-
- if (sdChanged == SAL_MEDIA_DESCRIPTION_UNCHANGED) {
- /* FIXME ZRTP, might be restarted in any cases? */
- lInfo() << "No need to restart the " << streamTypeName << " stream, SDP is unchanged";
- } else {
- if (sdChanged & SAL_MEDIA_DESCRIPTION_NETWORK_CHANGED) {
- lInfo() << "Network parameters have changed for the " << streamTypeName << " stream, update it";
- if (newMd->streams[i].type == SalAudio || newMd->streams[i].type == SalVideo) {
- updateStreamDestination(newMd, &newMd->streams[i]);
- }
- }
- if (sdChanged & SAL_MEDIA_DESCRIPTION_CRYPTO_KEYS_CHANGED) {
- lInfo() << "Crypto parameters have changed for the " << streamTypeName << " stream, update it";
- updateStreamCryptoParameters(&oldMd->streams[i], &newMd->streams[i]);
- }
- }
- }
- }
-
- for (const auto &hook : postProcessHooks) {
- hook();
- }
- postProcessHooks.clear();
-
- if (!iceAgent->hasCompleted()) {
- iceAgent->startConnectivityChecks();
- }
-
- if (oldMd)
- sal_media_description_unref(oldMd);
-
- return;
- }
- }
-
- stopStreams();
- if (mdChanged & SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED) {
- lInfo() << "Media ip type has changed, destroying sessions context on CallSession [" << q << "]";
- ms_media_stream_sessions_uninit(&sessions[mainAudioStreamIndex]);
- ms_media_stream_sessions_uninit(&sessions[mainVideoStreamIndex]);
- ms_media_stream_sessions_uninit(&sessions[mainTextStreamIndex]);
- }
- initializeStreams();
- }
-
- if (!audioStream) {
- /* This happens after pausing the call locally. The streams are destroyed and then we wait the 200Ok to recreate them */
- initializeStreams();
- }
-
- if (getParams()->earlyMediaSendingEnabled() && (state == CallSession::State::OutgoingEarlyMedia))
- prepareEarlyMediaForking();
-
- startStreams(targetState);
-
- if ((state == CallSession::State::Pausing) && pausedByApp && (q->getCore()->getCallCount() == 1)) {
- q->getCore()->getPrivate()->getToneManager()->startNamedTone(q->getSharedFromThis(), LinphoneToneCallOnHold);
- }
-
- updateFrozenPayloads(newMd);
-
- upBandwidth = linphone_core_get_upload_bandwidth(q->getCore()->getCCore());
-
- if (oldMd)
- sal_media_description_unref(oldMd);
-}
-
-void MediaSessionPrivate::updateStreamDestination (SalMediaDescription *newMd, SalStreamDescription *newDesc) {
- if (!sal_stream_description_active(newDesc))
- return;
-
- if (newDesc && newDesc->type == SalAudio) {
- if (audioStream) {
- const char *rtpAddr = (newDesc->rtp_addr[0] != '\0') ? newDesc->rtp_addr : newMd->addr;
- const char *rtcpAddr = (newDesc->rtcp_addr[0] != '\0') ? newDesc->rtcp_addr : newMd->addr;
- lInfo() << "Change audio stream destination: RTP=" << rtpAddr << ":" << newDesc->rtp_port << " RTCP=" << rtcpAddr << ":" << newDesc->rtcp_port;
- rtp_session_set_remote_addr_full(audioStream->ms.sessions.rtp_session, rtpAddr, newDesc->rtp_port, rtcpAddr, newDesc->rtcp_port);
- }
- }
-#ifdef VIDEO_ENABLED
- else if (newDesc && newDesc->type == SalVideo) {
- if (videoStream) {
- const char *rtpAddr = (newDesc->rtp_addr[0] != '\0') ? newDesc->rtp_addr : newMd->addr;
- const char *rtcpAddr = (newDesc->rtcp_addr[0] != '\0') ? newDesc->rtcp_addr : newMd->addr;
- lInfo() << "Change video stream destination: RTP=" << rtpAddr << ":" << newDesc->rtp_port << " RTCP=" << rtcpAddr << ":" << newDesc->rtcp_port;
- rtp_session_set_remote_addr_full(videoStream->ms.sessions.rtp_session, rtpAddr, newDesc->rtp_port, rtcpAddr, newDesc->rtcp_port);
- }
- }
-#endif
-}
-
-void MediaSessionPrivate::updateStreamsDestinations (SalMediaDescription *oldMd, SalMediaDescription *newMd) {
- SalStreamDescription *newAudioDesc = nullptr;
-
- #ifdef VIDEO_ENABLED
- SalStreamDescription *newVideoDesc = nullptr;
- #endif
-
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&newMd->streams[i]))
- continue;
- if (newMd->streams[i].type == SalAudio)
- newAudioDesc = &newMd->streams[i];
-
- #ifdef VIDEO_ENABLED
- else if (newMd->streams[i].type == SalVideo)
- newVideoDesc = &newMd->streams[i];
- #endif
- }
-
- updateStreamDestination(newMd, newAudioDesc);
-#ifdef VIDEO_ENABLED
- updateStreamDestination(newMd, newVideoDesc);
-#endif
-}
-
-// -----------------------------------------------------------------------------
-
-bool MediaSessionPrivate::allStreamsAvpfEnabled () const {
- int nbActiveStreams = 0;
- int nbAvpfEnabledStreams = 0;
- if (audioStream && media_stream_get_state(&audioStream->ms) == MSStreamStarted) {
- nbActiveStreams++;
- if (media_stream_avpf_enabled(&audioStream->ms))
- nbAvpfEnabledStreams++;
- }
- if (videoStream && media_stream_get_state(&videoStream->ms) == MSStreamStarted) {
- nbActiveStreams++;
- if (media_stream_avpf_enabled(&videoStream->ms))
- nbAvpfEnabledStreams++;
- }
- return (nbActiveStreams > 0) && (nbActiveStreams == nbAvpfEnabledStreams);
-}
-
-bool MediaSessionPrivate::allStreamsEncrypted () const {
- int numberOfEncryptedStreams = 0;
- int numberOfActiveStreams = 0;
- if (audioStream && (media_stream_get_state(&audioStream->ms) == MSStreamStarted)) {
- numberOfActiveStreams++;
- if (media_stream_secured(&audioStream->ms))
- numberOfEncryptedStreams++;
- }
- if (videoStream && (media_stream_get_state(&videoStream->ms) == MSStreamStarted)) {
- numberOfActiveStreams++;
- if (media_stream_secured(&videoStream->ms))
- numberOfEncryptedStreams++;
- }
- if (textStream && (media_stream_get_state(&textStream->ms) == MSStreamStarted)) {
- numberOfActiveStreams++;
- if (media_stream_secured(&textStream->ms))
- numberOfEncryptedStreams++;
- }
- return (numberOfActiveStreams > 0) && (numberOfActiveStreams == numberOfEncryptedStreams);
-}
-
-bool MediaSessionPrivate::atLeastOneStreamStarted () const {
- return (audioStream && (media_stream_get_state(&audioStream->ms) == MSStreamStarted))
- || (videoStream && (media_stream_get_state(&videoStream->ms) == MSStreamStarted))
- || (textStream && (media_stream_get_state(&textStream->ms) == MSStreamStarted));
-}
-
-void MediaSessionPrivate::audioStreamAuthTokenReady (const string &authToken, bool verified) {
- this->authToken = authToken;
- authTokenVerified = verified;
- lInfo() << "Authentication token is " << authToken << "(" << (verified ? "verified" : "unverified") << ")";
-}
-
-void MediaSessionPrivate::audioStreamEncryptionChanged (bool encrypted) {
- propagateEncryptionChanged();
-
- #ifdef VIDEO_ENABLED
- L_Q();
- /* Enable video encryption */
- if ((getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP) && q->getCurrentParams()->videoEnabled()) {
- lInfo() << "Trying to start ZRTP encryption on video stream";
- video_stream_start_zrtp(videoStream);
- }
- #endif
-}
-
-uint16_t MediaSessionPrivate::getAvpfRrInterval () const {
- uint16_t rrInterval = 0;
- if (audioStream && (media_stream_get_state(&audioStream->ms) == MSStreamStarted)) {
- uint16_t streamRrInterval = media_stream_get_avpf_rr_interval(&audioStream->ms);
- if (streamRrInterval > rrInterval) rrInterval = streamRrInterval;
- }
- if (videoStream && (media_stream_get_state(&videoStream->ms) == MSStreamStarted)) {
- uint16_t streamRrInterval = media_stream_get_avpf_rr_interval(&videoStream->ms);
- if (streamRrInterval > rrInterval) rrInterval = streamRrInterval;
- }
- return rrInterval;
-}
-
-unsigned int MediaSessionPrivate::getNbActiveStreams () const {
- SalMediaDescription *md = nullptr;
- if (op)
- md = op->getRemoteMediaDescription();
- if (!md)
- return 0;
- return sal_media_description_nb_active_streams_of_type(md, SalAudio) + sal_media_description_nb_active_streams_of_type(md, SalVideo) + sal_media_description_nb_active_streams_of_type(md, SalText);
-}
-
-bool MediaSessionPrivate::isEncryptionMandatory () const {
- L_Q();
- if (getParams()->getMediaEncryption() == LinphoneMediaEncryptionDTLS) {
- lInfo() << "Forced encryption mandatory on CallSession [" << q << "] due to SRTP-DTLS";
- return true;
- }
- return getParams()->mandatoryMediaEncryptionEnabled();
-}
-
-int MediaSessionPrivate::mediaParametersChanged (SalMediaDescription *oldMd, SalMediaDescription *newMd) {
- L_Q();
- if (forceStreamsReconstruction) {
- forceStreamsReconstruction = false;
- return SAL_MEDIA_DESCRIPTION_FORCE_STREAM_RECONSTRUCTION;
- }
- if (getParams()->getPrivate()->getInConference() != getCurrentParams()->getPrivate()->getInConference())
- return SAL_MEDIA_DESCRIPTION_FORCE_STREAM_RECONSTRUCTION;
- if (upBandwidth != linphone_core_get_upload_bandwidth(q->getCore()->getCCore()))
- return SAL_MEDIA_DESCRIPTION_FORCE_STREAM_RECONSTRUCTION;
- if (localDescChanged) {
- char *differences = sal_media_description_print_differences(localDescChanged);
- lInfo() << "Local description has changed: " << differences;
- ms_free(differences);
- }
- int otherDescChanged = sal_media_description_global_equals(oldMd, newMd);
- if (otherDescChanged) {
- char *differences = sal_media_description_print_differences(otherDescChanged);
- lInfo() << "Other description has changed: " << differences;
- ms_free(differences);
- }
- return localDescChanged | otherDescChanged;
+ return getParams()->mandatoryMediaEncryptionEnabled();
}
void MediaSessionPrivate::propagateEncryptionChanged () {
L_Q();
- if (!allStreamsEncrypted()) {
+
+ string authToken = getStreamsGroup().getAuthenticationToken();
+ bool authTokenVerified = getStreamsGroup().getAuthenticationTokenVerified();
+ if (!getStreamsGroup().allStreamsEncrypted()) {
lInfo() << "Some streams are not encrypted";
getCurrentParams()->setMediaEncryption(LinphoneMediaEncryptionNone);
if (listener)
@@ -4014,7 +1584,15 @@ void MediaSessionPrivate::propagateEncryptionChanged () {
if (encryptionEngine && authTokenVerified) {
const SalAddress *remoteAddress = getOp()->getRemoteContactAddress();
peerDeviceId = sal_address_as_string_uri_only(remoteAddress);
- encryptionEngine->authenticationVerified(audioStream->ms.sessions.zrtp_context, op->getRemoteMediaDescription(), peerDeviceId);
+ Stream *stream = mainAudioStreamIndex != -1 ? getStreamsGroup().getStream(mainAudioStreamIndex) : nullptr;
+ if (stream){
+ MS2Stream *ms2s = dynamic_cast<MS2Stream*>(stream);
+ if (ms2s){
+ encryptionEngine->authenticationVerified(ms2s->getZrtpContext(), op->getRemoteMediaDescription(), peerDeviceId);
+ }else{
+ lError() << "Could not dynamic_cast to MS2Stream in propagateEncryptionChanged().";
+ }
+ }
ms_free(peerDeviceId);
}
} else {
@@ -4027,115 +1605,34 @@ void MediaSessionPrivate::propagateEncryptionChanged () {
: (q->getCurrentParams()->getMediaEncryption() == LinphoneMediaEncryptionDTLS) ? "DTLS" : "Unknown mechanism");
if (listener)
listener->onEncryptionChanged(q->getSharedFromThis(), true, authToken);
-#ifdef VIDEO_ENABLED
- if (isEncryptionMandatory() && videoStream && media_stream_started(&videoStream->ms)) {
+
+ Stream *videoStream = mainVideoStreamIndex != -1 ? getStreamsGroup().getStream(mainVideoStreamIndex) : nullptr;
+ if (isEncryptionMandatory() && videoStream && videoStream->getState() == Stream::Running) {
/* Nothing could have been sent yet so generating key frame */
- video_stream_send_vfu(videoStream);
+ VideoControlInterface *vc = dynamic_cast<VideoControlInterface*> (videoStream);
+ if (vc) vc->sendVfu();
}
-#endif
- }
-}
-
-// -----------------------------------------------------------------------------
-
-void MediaSessionPrivate::fillLogStats (MediaStream *st) {
- float quality = media_stream_get_average_quality_rating(st);
- if (quality >= 0) {
- if (static_cast<int>(log->quality) == -1)
- log->quality = quality;
- else
- log->quality *= quality / 5.0f;
}
}
-void MediaSessionPrivate::updateRtpStats (LinphoneCallStats *stats, int streamIndex) {
- if (sessions[streamIndex].rtp_session) {
- const rtp_stats_t *rtpStats = rtp_session_get_stats(sessions[streamIndex].rtp_session);
- if (rtpStats)
- _linphone_call_stats_set_rtp_stats(stats, rtpStats);
- }
+MSWebCam *MediaSessionPrivate::getVideoDevice()const{
+#ifdef VIDEO_ENABLED
+ MS2VideoStream *vs = getStreamsGroup().lookupMainStreamInterface<MS2VideoStream>(SalVideo);
+ if (vs) return vs->getVideoDevice(state);
+#endif
+ return nullptr;
}
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::executeBackgroundTasks (bool oneSecondElapsed) {
- L_Q();
- switch (state) {
- case CallSession::State::StreamsRunning:
- case CallSession::State::OutgoingEarlyMedia:
- case CallSession::State::IncomingEarlyMedia:
- case CallSession::State::PausedByRemote:
- case CallSession::State::Paused:
- if (oneSecondElapsed) {
- float audioLoad = 0.f;
- float videoLoad = 0.f;
- float textLoad = 0.f;
- if (audioStream && audioStream->ms.sessions.ticker)
- audioLoad = ms_ticker_get_average_load(audioStream->ms.sessions.ticker);
- if (videoStream && videoStream->ms.sessions.ticker)
- videoLoad = ms_ticker_get_average_load(videoStream->ms.sessions.ticker);
- if (textStream && textStream->ms.sessions.ticker)
- textLoad = ms_ticker_get_average_load(textStream->ms.sessions.ticker);
- reportBandwidth();
- lInfo() << "Thread processing load: audio=" << audioLoad << "\tvideo=" << videoLoad << "\ttext=" << textLoad;
- }
- break;
- default:
- /* No stats for other states */
- break;
- }
-
- handleStreamEvents(mainAudioStreamIndex);
- handleStreamEvents(mainVideoStreamIndex);
- handleStreamEvents(mainTextStreamIndex);
-
- if (listener)
- listener->onNoMediaTimeoutCheck(q->getSharedFromThis(), oneSecondElapsed);
-}
-
-void MediaSessionPrivate::reportBandwidth () {
- L_Q();
- reportBandwidthForStream(&audioStream->ms, LinphoneStreamTypeAudio);
- reportBandwidthForStream(&videoStream->ms, LinphoneStreamTypeVideo);
- reportBandwidthForStream(&textStream->ms, LinphoneStreamTypeText);
-
- lInfo() << "Bandwidth usage for CallSession [" << q << "]:\n" << fixed << setprecision(2) <<
- "\tRTP audio=[d=" << linphone_call_stats_get_download_bandwidth(audioStats) << ",u=" << linphone_call_stats_get_upload_bandwidth(audioStats) <<
- "], video=[d=" << linphone_call_stats_get_download_bandwidth(videoStats) << ",u=" << linphone_call_stats_get_upload_bandwidth(videoStats) << ",ed=" << linphone_call_stats_get_estimated_download_bandwidth(videoStats) <<
- "], text=[d=" << linphone_call_stats_get_download_bandwidth(textStats) << ",u=" << linphone_call_stats_get_upload_bandwidth(textStats) << "] kbits/sec\n" <<
- "\tRTCP audio=[d=" << linphone_call_stats_get_rtcp_download_bandwidth(audioStats) << ",u=" << linphone_call_stats_get_rtcp_upload_bandwidth(audioStats) <<
- "], video=[d=" << linphone_call_stats_get_rtcp_download_bandwidth(videoStats) << ",u=" << linphone_call_stats_get_rtcp_upload_bandwidth(videoStats) <<
- "], text=[d=" << linphone_call_stats_get_rtcp_download_bandwidth(textStats) << ",u=" << linphone_call_stats_get_rtcp_upload_bandwidth(textStats) << "] kbits/sec";
-}
-void MediaSessionPrivate::reportBandwidthForStream (MediaStream *ms, LinphoneStreamType type) {
- L_Q();
- LinphoneCallStats *stats = nullptr;
- if (type == LinphoneStreamTypeAudio) {
- stats = audioStats;
- } else if (type == LinphoneStreamTypeVideo) {
- stats = videoStats;
- } else if (type == LinphoneStreamTypeText) {
- stats = textStats;
- } else
- return;
- bool active = ms ? (media_stream_get_state(ms) == MSStreamStarted) : false;
- _linphone_call_stats_set_download_bandwidth(stats, active ? (float)(media_stream_get_down_bw(ms) * 1e-3) : 0.f);
- _linphone_call_stats_set_upload_bandwidth(stats, active ? (float)(media_stream_get_up_bw(ms) * 1e-3) : 0.f);
- _linphone_call_stats_set_rtcp_download_bandwidth(stats, active ? (float)(media_stream_get_rtcp_down_bw(ms) * 1e-3) : 0.f);
- _linphone_call_stats_set_rtcp_upload_bandwidth(stats, active ? (float)(media_stream_get_rtcp_up_bw(ms) * 1e-3) : 0.f);
- _linphone_call_stats_set_ip_family_of_remote(stats,
- active ? (ortp_stream_is_ipv6(&ms->sessions.rtp_session->rtp.gs) ? LinphoneAddressFamilyInet6 : LinphoneAddressFamilyInet) : LinphoneAddressFamilyUnspec);
-
- if (q->getCore()->getCCore()->send_call_stats_periodical_updates) {
- if (active)
- linphone_call_stats_update(stats, ms);
- _linphone_call_stats_set_updated(stats, _linphone_call_stats_get_updated(stats) | LINPHONE_CALL_STATS_PERIODICAL_UPDATE);
- if (listener)
- listener->onStatsUpdated(q->getSharedFromThis(), stats);
- _linphone_call_stats_set_updated(stats, 0);
- }
+// -----------------------------------------------------------------------------
+
+void MediaSessionPrivate::lossOfMediaDetected() {
+ L_Q();
+ if (listener)
+ listener->onLossOfMediaDetected(q->getSharedFromThis());
}
// -----------------------------------------------------------------------------
@@ -4161,16 +1658,6 @@ void MediaSessionPrivate::handleIncomingReceivedStateInIncomingNotification () {
acceptOrTerminateReplacedSessionInIncomingNotification();
}
-bool MediaSessionPrivate::isReadyForInvite () const {
- bool callSessionReady = CallSessionPrivate::isReadyForInvite();
- bool iceReady = false;
- if (iceAgent->hasSession()) {
- if (iceAgent->candidatesGathered())
- iceReady = true;
- } else
- iceReady = true;
- return callSessionReady && iceReady;
-}
LinphoneStatus MediaSessionPrivate::pause () {
L_Q();
@@ -4189,20 +1676,19 @@ LinphoneStatus MediaSessionPrivate::pause () {
}
broken = false;
setState(CallSession::State::Pausing, "Pausing call");
- makeLocalMediaDescription();
- op->setLocalMediaDescription(localDesc);
+ makeLocalMediaDescription(true);
op->update(subject.c_str(), false);
if (listener)
listener->onResetCurrentSession(q->getSharedFromThis());
- if (audioStream || videoStream || textStream)
- stopStreams();
+ stopStreams();
pausedByApp = false;
return 0;
}
int MediaSessionPrivate::restartInvite () {
stopStreams();
- initializeStreams();
+ getStreamsGroup().clearStreams();
+ makeLocalMediaDescription(true);
return CallSessionPrivate::restartInvite();
}
@@ -4212,16 +1698,8 @@ void MediaSessionPrivate::setTerminated () {
}
LinphoneStatus MediaSessionPrivate::startAcceptUpdate (CallSession::State nextState, const string &stateInfo) {
- if (iceAgent->hasSession() && (iceAgent->getNbLosingPairs() > 0)) {
- /* Defer the sending of the answer until there are no losing pairs left */
- return 0;
- }
- makeLocalMediaDescription();
- updateRemoteSessionIdAndVer();
- op->setLocalMediaDescription(localDesc);
op->accept();
SalMediaDescription *md = op->getFinalMediaDescription();
- iceAgent->stopIceForInactiveStreams(md);
if (md && !sal_media_description_empty(md))
updateStreams(md, nextState);
setState(nextState, stateInfo);
@@ -4230,12 +1708,8 @@ LinphoneStatus MediaSessionPrivate::startAcceptUpdate (CallSession::State nextSt
LinphoneStatus MediaSessionPrivate::startUpdate (const string &subject) {
L_Q();
- fillMulticastMediaAddresses();
- if (!getParams()->getPrivate()->getNoUserConsent())
- makeLocalMediaDescription();
- if (!q->getCore()->getCCore()->sip_conf.sdp_200_ack)
- op->setLocalMediaDescription(localDesc);
- else
+
+ if (q->getCore()->getCCore()->sip_conf.sdp_200_ack)
op->setLocalMediaDescription(nullptr);
LinphoneStatus result = CallSessionPrivate::startUpdate(subject);
if (q->getCore()->getCCore()->sip_conf.sdp_200_ack) {
@@ -4256,24 +1730,30 @@ void MediaSessionPrivate::terminate () {
void MediaSessionPrivate::updateCurrentParams () const {
CallSessionPrivate::updateCurrentParams();
- LinphoneVideoDefinition *vdef = linphone_video_definition_new(MS_VIDEO_SIZE_UNKNOWN_W, MS_VIDEO_SIZE_UNKNOWN_H, nullptr);
- getCurrentParams()->getPrivate()->setSentVideoDefinition(vdef);
- getCurrentParams()->getPrivate()->setReceivedVideoDefinition(vdef);
- linphone_video_definition_unref(vdef);
-#ifdef VIDEO_ENABLED
- if (videoStream) {
- MSVideoSize vsize = video_stream_get_sent_video_size(videoStream);
- vdef = linphone_video_definition_new(static_cast<unsigned int>(vsize.width), static_cast<unsigned int>(vsize.height), nullptr);
+
+ VideoControlInterface *i = getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i){
+ VideoControlInterface::VideoStats st;
+ LinphoneVideoDefinition *vdef;
+
+ i->getRecvStats(&st);
+ vdef = linphone_video_definition_new((unsigned)st.width, (unsigned)st.height, nullptr);
+ getCurrentParams()->getPrivate()->setReceivedVideoDefinition(vdef);
+ linphone_video_definition_unref(vdef);
+ getCurrentParams()->getPrivate()->setReceivedFps(st.fps);
+
+ i->getSendStats(&st);
+ vdef = linphone_video_definition_new((unsigned)st.width, (unsigned)st.height, nullptr);
getCurrentParams()->getPrivate()->setSentVideoDefinition(vdef);
linphone_video_definition_unref(vdef);
- vsize = video_stream_get_received_video_size(videoStream);
- vdef = linphone_video_definition_new(static_cast<unsigned int>(vsize.width), static_cast<unsigned int>(vsize.height), nullptr);
+ getCurrentParams()->getPrivate()->setSentFps(st.fps);
+
+ }else{
+ LinphoneVideoDefinition *vdef = linphone_video_definition_new(MS_VIDEO_SIZE_UNKNOWN_W, MS_VIDEO_SIZE_UNKNOWN_H, nullptr);
+ getCurrentParams()->getPrivate()->setSentVideoDefinition(vdef);
getCurrentParams()->getPrivate()->setReceivedVideoDefinition(vdef);
linphone_video_definition_unref(vdef);
- getCurrentParams()->getPrivate()->setSentFps(video_stream_get_sent_framerate(videoStream));
- getCurrentParams()->getPrivate()->setReceivedFps(video_stream_get_received_framerate(videoStream));
}
-#endif
/* REVISITED
* Previous code was buggy.
@@ -4283,7 +1763,7 @@ void MediaSessionPrivate::updateCurrentParams () const {
* mechanism (added by jehan: and encryption status from media which is much stronger than only result of offer/answer )
* Typically there can be inactive streams for which the media layer has no idea of whether they are encrypted or not.
*/
-
+ string authToken = getStreamsGroup().getAuthenticationToken();
switch (getParams()->getMediaEncryption()) {
case LinphoneMediaEncryptionZRTP:
if (atLeastOneStreamStarted()) {
@@ -4325,37 +1805,41 @@ void MediaSessionPrivate::updateCurrentParams () const {
else
getCurrentParams()->setAvpfRrInterval(0);
if (md) {
- SalStreamDescription *sd = sal_media_description_find_best_stream(md, SalAudio);
- getCurrentParams()->setAudioDirection(sd ? MediaSessionParamsPrivate::salStreamDirToMediaDirection(sd->dir) : LinphoneMediaDirectionInactive);
- if (getCurrentParams()->getAudioDirection() != LinphoneMediaDirectionInactive) {
- const char *rtpAddr = (sd->rtp_addr[0] != '\0') ? sd->rtp_addr : md->addr;
- getCurrentParams()->enableAudioMulticast(!!ms_is_multicast(rtpAddr));
- } else
- getCurrentParams()->enableAudioMulticast(false);
- sd = sal_media_description_find_best_stream(md, SalVideo);
- getCurrentParams()->getPrivate()->enableImplicitRtcpFb(sd && sal_stream_description_has_implicit_avpf(sd));
- getCurrentParams()->setVideoDirection(sd ? MediaSessionParamsPrivate::salStreamDirToMediaDirection(sd->dir) : LinphoneMediaDirectionInactive);
- if (getCurrentParams()->getVideoDirection() != LinphoneMediaDirectionInactive) {
- const char *rtpAddr = (sd->rtp_addr[0] != '\0') ? sd->rtp_addr : md->addr;
- getCurrentParams()->enableVideoMulticast(!!ms_is_multicast(rtpAddr));
- } else
- getCurrentParams()->enableVideoMulticast(false);
+ if (mainAudioStreamIndex != -1){
+ SalStreamDescription *sd = &md->streams[mainAudioStreamIndex];
+ getCurrentParams()->setAudioDirection(sd ? MediaSessionParamsPrivate::salStreamDirToMediaDirection(sd->dir) : LinphoneMediaDirectionInactive);
+ if (getCurrentParams()->getAudioDirection() != LinphoneMediaDirectionInactive) {
+ const char *rtpAddr = (sd->rtp_addr[0] != '\0') ? sd->rtp_addr : md->addr;
+ getCurrentParams()->enableAudioMulticast(!!ms_is_multicast(rtpAddr));
+ } else
+ getCurrentParams()->enableAudioMulticast(false);
+ getCurrentParams()->enableAudio(sal_stream_description_enabled(sd));
+ }
+ if (mainVideoStreamIndex != -1){
+ SalStreamDescription *sd = &md->streams[mainVideoStreamIndex];
+ getCurrentParams()->getPrivate()->enableImplicitRtcpFb(sd && sal_stream_description_has_implicit_avpf(sd));
+ getCurrentParams()->setVideoDirection(sd ? MediaSessionParamsPrivate::salStreamDirToMediaDirection(sd->dir) : LinphoneMediaDirectionInactive);
+ if (getCurrentParams()->getVideoDirection() != LinphoneMediaDirectionInactive) {
+ const char *rtpAddr = (sd->rtp_addr[0] != '\0') ? sd->rtp_addr : md->addr;
+ getCurrentParams()->enableVideoMulticast(!!ms_is_multicast(rtpAddr));
+ } else
+ getCurrentParams()->enableVideoMulticast(false);
+ getCurrentParams()->enableVideo(sal_stream_description_enabled(sd));
+ }
+ if (mainTextStreamIndex != -1){
+ SalStreamDescription *sd = &md->streams[mainTextStreamIndex];
+ // Direction and multicast are not supported for real-time text.
+ getCurrentParams()->enableRealtimeText(sal_stream_description_enabled(sd));
+ }
}
+ getCurrentParams()->getPrivate()->setUpdateCallWhenIceCompleted(getParams()->getPrivate()->getUpdateCallWhenIceCompleted());
}
// -----------------------------------------------------------------------------
-void MediaSessionPrivate::accept (const MediaSessionParams *msp, bool wasRinging) {
- L_Q();
- if (msp) {
- setParams(new MediaSessionParams(*msp));
- iceAgent->prepare(localDesc, true, false /*we don't allow gathering now, it must have been done before*/);
- makeLocalMediaDescription();
- op->setLocalMediaDescription(localDesc);
- }
-
- updateRemoteSessionIdAndVer();
+void MediaSessionPrivate::startAccept(){
+ L_Q();
/* Give a chance a set card prefered sampling frequency */
if (localDesc->streams[0].max_rate > 0) {
lInfo() << "Configuring prefered card sampling rate to [" << localDesc->streams[0].max_rate << "]";
@@ -4365,20 +1849,30 @@ void MediaSessionPrivate::accept (const MediaSessionParams *msp, bool wasRinging
ms_snd_card_set_preferred_sample_rate(q->getCore()->getCCore()->sound_conf.capt_sndcard, localDesc->streams[0].max_rate);
}
- LinphoneCore *lc = q->getCore()->getCCore();
- if (!wasRinging && (audioStream->ms.state == MSStreamInitialized) && !lc->use_files) {
- audio_stream_prepare_sound(audioStream, lc->sound_conf.play_sndcard, lc->sound_conf.capt_sndcard);
- }
-
CallSessionPrivate::accept(nullptr);
SalMediaDescription *newMd = op->getFinalMediaDescription();
- iceAgent->stopIceForInactiveStreams(newMd);
if (newMd) {
updateStreams(newMd, CallSession::State::StreamsRunning);
setState(CallSession::State::StreamsRunning, "Connected (streams running)");
} else
expectMediaInAck = true;
+ if (callAcceptanceDefered) callAcceptanceDefered = false;
+}
+
+void MediaSessionPrivate::accept (const MediaSessionParams *msp, bool wasRinging) {
+ if (msp) {
+ setParams(new MediaSessionParams(*msp));
+ }
+ if (msp || localDesc == nullptr) makeLocalMediaDescription(op->getRemoteMediaDescription() ? false : true);
+
+ updateRemoteSessionIdAndVer();
+
+ if (getStreamsGroup().prepare()){
+ callAcceptanceDefered = true;
+ return; /* Deferred until completion of ICE gathering */
+ }
+ startAccept();
}
LinphoneStatus MediaSessionPrivate::acceptUpdate (const CallSessionParams *csp, CallSession::State nextState, const string &stateInfo) {
@@ -4409,11 +1903,10 @@ LinphoneStatus MediaSessionPrivate::acceptUpdate (const CallSessionParams *csp,
lWarning() << "Video isn't supported in conference";
getParams()->enableVideo(false);
}
- /* Update multicast params according to call params */
- fillMulticastMediaAddresses();
- iceAgent->checkSession(IR_Controlled, true);
- initializeStreams(); /* So that video stream is initialized if necessary */
- if (iceAgent->prepare(localDesc, true))
+ updateRemoteSessionIdAndVer();
+ makeLocalMediaDescription(op->getRemoteMediaDescription() ? false : true);
+
+ if (getStreamsGroup().prepare())
return 0; /* Deferred until completion of ICE gathering */
startAcceptUpdate(nextState, stateInfo);
return 0;
@@ -4422,87 +1915,32 @@ LinphoneStatus MediaSessionPrivate::acceptUpdate (const CallSessionParams *csp,
// -----------------------------------------------------------------------------
void MediaSessionPrivate::refreshSockets () {
- for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- MSMediaStreamSessions *mss = &sessions[i];
- if (mss->rtp_session)
- rtp_session_refresh_sockets(mss->rtp_session);
- }
+ getStreamsGroup().refreshSockets();
}
void MediaSessionPrivate::reinviteToRecoverFromConnectionLoss () {
L_Q();
lInfo() << "MediaSession [" << q << "] is going to be updated (reINVITE) in order to recover from lost connectivity";
- if (iceAgent->hasSession())
- iceAgent->resetSession(IR_Controlling);
+ getStreamsGroup().getIceService().resetSession();
q->update(getParams());
}
void MediaSessionPrivate::repairByInviteWithReplaces () {
if ((state == CallSession::State::IncomingEarlyMedia) || (state == CallSession::State::OutgoingEarlyMedia)) {
stopStreams();
- initializeStreams();
}
CallSessionPrivate::repairByInviteWithReplaces();
}
-// -----------------------------------------------------------------------------
-
-#ifdef VIDEO_ENABLED
-void MediaSessionPrivate::videoStreamEventCb (const MSFilter *f, const unsigned int eventId, const void *args) {
- L_Q();
- switch (eventId) {
- case MS_VIDEO_DECODER_DECODING_ERRORS:
- lWarning() << "MS_VIDEO_DECODER_DECODING_ERRORS";
- if (videoStream && video_stream_is_decoding_error_to_be_reported(videoStream, 5000)) {
- video_stream_decoding_error_reported(videoStream);
- q->sendVfuRequest();
- }
- break;
- case MS_VIDEO_DECODER_RECOVERED_FROM_ERRORS:
- lInfo() << "MS_VIDEO_DECODER_RECOVERED_FROM_ERRORS";
- if (videoStream)
- video_stream_decoding_error_recovered(videoStream);
- break;
- case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
- lInfo() << "First video frame decoded successfully";
- if (listener)
- listener->onFirstVideoFrameDecoded(q->getSharedFromThis());
- break;
- case MS_VIDEO_DECODER_SEND_PLI:
- case MS_VIDEO_DECODER_SEND_SLI:
- case MS_VIDEO_DECODER_SEND_RPSI:
- /* Handled internally by mediastreamer2 */
- break;
- case MS_CAMERA_PREVIEW_SIZE_CHANGED: {
- MSVideoSize size = *(MSVideoSize *)args;
- lInfo() << "Camera video preview size changed: " << size.width << "x" << size.height;
- linphone_core_resize_video_preview(q->getCore()->getCCore(), size.width, size.height);
- break;
- }
- default:
- lWarning() << "Unhandled event " << eventId;
- break;
- }
-}
-#endif
-
-void MediaSessionPrivate::realTimeTextCharacterReceived (MSFilter *f, unsigned int id, void *arg) {
- L_Q();
- if (id == MS_RTT_4103_RECEIVED_CHAR) {
- RealtimeTextReceivedCharacter *data = reinterpret_cast<RealtimeTextReceivedCharacter *>(arg);
- if (listener)
- listener->onRealTimeTextCharacterReceived(q->getSharedFromThis(), data);
- }
-}
-
int MediaSessionPrivate::sendDtmf () {
L_Q();
LinphoneCore *lc = q->getCore()->getCCore();
// By default we send DTMF RFC2833 if we do not have enabled SIP_INFO but we can also send RFC2833 and SIP_INFO
if (linphone_core_get_use_rfc2833_for_dtmf(lc) || !linphone_core_get_use_info_for_dtmf(lc)) {
+ AudioControlInterface *i = getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
// In Band DTMF
- if (audioStream)
- audio_stream_send_dtmf(audioStream, dtmfSequence.front());
+ if (i)
+ i->sendDtmf(dtmfSequence.front());
else {
lError() << "Cannot send RFC2833 DTMF when we are not in communication";
return FALSE;
@@ -4526,12 +1964,12 @@ int MediaSessionPrivate::sendDtmf () {
// -----------------------------------------------------------------------------
int MediaSessionPrivate::resumeAfterFailedTransfer (void *userData, unsigned int) {
- MediaSession *session = reinterpret_cast<MediaSession *>(userData);
+ MediaSession *session = static_cast<MediaSession *>(userData);
return session->getPrivate()->resumeAfterFailedTransfer();
}
bool_t MediaSessionPrivate::startPendingRefer (void *userData) {
- MediaSession *session = reinterpret_cast<MediaSession *>(userData);
+ MediaSession *session = static_cast<MediaSession *>(userData);
session->getPrivate()->startPendingRefer();
return TRUE;
}
@@ -4575,6 +2013,10 @@ void MediaSessionPrivate::stunAuthRequestedCb (const char *realm, const char *no
*username = user;
}
+IceSession *MediaSessionPrivate::getIceSession()const{
+ return getIceService().getSession();
+}
+
// =============================================================================
MediaSession::MediaSession (const shared_ptr<Core> &core, shared_ptr<Participant> me, const CallSessionParams *params, CallSessionListener *listener)
@@ -4588,24 +2030,8 @@ MediaSession::MediaSession (const shared_ptr<Core> &core, shared_ptr<Participant
else
d->setParams(new MediaSessionParams());
d->setCurrentParams(new MediaSessionParams());
-
- d->audioStats = _linphone_call_stats_new();
- d->initStats(d->audioStats, LinphoneStreamTypeAudio);
- d->videoStats = _linphone_call_stats_new();
- d->initStats(d->videoStats, LinphoneStreamTypeVideo);
- d->textStats = _linphone_call_stats_new();
- d->initStats(d->textStats, LinphoneStreamTypeText);
-
- int minPort, maxPort;
- linphone_core_get_audio_port_range(getCore()->getCCore(), &minPort, &maxPort);
- d->setPortConfig(d->mainAudioStreamIndex, make_pair(minPort, maxPort));
- linphone_core_get_video_port_range(getCore()->getCCore(), &minPort, &maxPort);
- d->setPortConfig(d->mainVideoStreamIndex, make_pair(minPort, maxPort));
- linphone_core_get_text_port_range(getCore()->getCCore(), &minPort, &maxPort);
- d->setPortConfig(d->mainTextStreamIndex, make_pair(minPort, maxPort));
-
- memset(d->sessions, 0, sizeof(d->sessions));
- d->iceAgent = makeUnique<IceAgent>(*this);
+ d->streamsGroup = makeUnique<StreamsGroup>(*this);
+ d->streamsGroup->getIceService().setListener(d);
lInfo() << "New MediaSession [" << this << "] initialized (LinphoneCore version: " << linphone_core_get_version() << ")";
}
@@ -4613,14 +2039,7 @@ MediaSession::MediaSession (const shared_ptr<Core> &core, shared_ptr<Participant
MediaSession::~MediaSession () {
L_D();
cancelDtmfs();
- if (d->audioStream || d->videoStream)
- d->freeResources();
- if (d->audioStats)
- linphone_call_stats_unref(d->audioStats);
- if (d->videoStats)
- linphone_call_stats_unref(d->videoStats);
- if (d->textStats)
- linphone_call_stats_unref(d->textStats);
+ d->freeResources();
if (d->natPolicy)
linphone_nat_policy_unref(d->natPolicy);
if (d->localDesc)
@@ -4662,8 +2081,7 @@ LinphoneStatus MediaSession::acceptEarlyMedia (const MediaSessionParams *msp) {
/* If parameters are passed, update the media description */
if (msp) {
d->setParams(new MediaSessionParams(*msp));
- d->makeLocalMediaDescription();
- d->op->setLocalMediaDescription(d->localDesc);
+ d->makeLocalMediaDescription(false);
d->op->setSentCustomHeaders(d->getParams()->getPrivate()->getCustomHeaders());
}
d->op->notifyRinging(true);
@@ -4706,12 +2124,10 @@ void MediaSession::configure (LinphoneCallDir direction, LinphoneProxyConfig *cf
if (direction == LinphoneCallOutgoing) {
d->selectOutgoingIpVersion();
- d->getLocalIp(to);
- d->initializeStreams(); // Reserve the sockets immediately
- d->getCurrentParams()->getPrivate()->setUpdateCallWhenIceCompleted(d->getParams()->getPrivate()->getUpdateCallWhenIceCompleted());
- d->fillMulticastMediaAddresses();
- if (d->natPolicy && linphone_nat_policy_ice_enabled(d->natPolicy))
- d->iceAgent->checkSession(IR_Controlling, false);
+ if (!getCore()->getCCore()->sip_conf.sdp_200_ack){
+ /* Do not make a local media description when sending an empty INVITE. */
+ d->makeLocalMediaDescription(true);
+ }
d->runStunTestsIfNeeded();
d->discoverMtu(to);
} else if (direction == LinphoneCallIncoming) {
@@ -4720,22 +2136,10 @@ void MediaSession::configure (LinphoneCallDir direction, LinphoneProxyConfig *cf
* remote offer, if any. If the remote offer contains IPv4 addresses, we should propose IPv4 as well. */
Address cleanedFrom = from;
cleanedFrom.clean();
- d->getLocalIp(cleanedFrom);
d->setParams(new MediaSessionParams());
- d->params->initDefault(getCore());
+ d->params->initDefault(getCore(), LinphoneCallIncoming);
d->initializeParamsAccordingToIncomingCallParams();
- SalMediaDescription *md = d->op->getRemoteMediaDescription();
- if (d->natPolicy && linphone_nat_policy_ice_enabled(d->natPolicy)) {
- if (md) {
- /* Create the ice session now if ICE is required */
- d->iceAgent->checkSession(IR_Controlled, false);
- } else {
- linphone_nat_policy_unref(d->natPolicy);
- d->natPolicy = nullptr;
- lWarning() << "ICE not supported for incoming INVITE without SDP";
- }
- }
- d->initializeStreams(); // Reserve the sockets immediately
+ d->makeLocalMediaDescription(op->getRemoteMediaDescription() ? false : true);
if (d->natPolicy)
d->runStunTestsIfNeeded();
d->discoverMtu(cleanedFrom);
@@ -4759,31 +2163,42 @@ LinphoneStatus MediaSession::deferUpdate () {
void MediaSession::initiateIncoming () {
L_D();
CallSession::initiateIncoming();
- d->initializeStreams();
if (d->natPolicy) {
- if (linphone_nat_policy_ice_enabled(d->natPolicy))
- d->deferIncomingNotification = d->iceAgent->prepare(d->localDesc, true);
+ if (linphone_nat_policy_ice_enabled(d->natPolicy)){
+
+ d->deferIncomingNotification = d->getStreamsGroup().prepare();
+ /*
+ * If ICE gathering is done, we can update the local media description immediately.
+ * Otherwise, we'll get the ORTP_EVENT_ICE_GATHERING_FINISHED event later.
+ */
+ if (!d->deferIncomingNotification) d->updateLocalMediaDescriptionFromIce();
+ }
}
}
bool MediaSession::initiateOutgoing () {
L_D();
bool defer = CallSession::initiateOutgoing();
- d->initializeStreams();
if (linphone_nat_policy_ice_enabled(d->natPolicy)) {
if (getCore()->getCCore()->sip_conf.sdp_200_ack)
lWarning() << "ICE is not supported when sending INVITE without SDP";
else {
/* Defer the start of the call after the ICE gathering process */
- defer |= d->iceAgent->prepare(d->localDesc, false);
+ bool ice_needs_defer = d->getStreamsGroup().prepare();
+ if (!ice_needs_defer) {
+ /*
+ * If ICE gathering is done, we can update the local media description immediately.
+ * Otherwise, we'll get the ORTP_EVENT_ICE_GATHERING_FINISHED event later.
+ */
+ d->updateLocalMediaDescriptionFromIce();
+ }
+ defer |= ice_needs_defer;
}
}
return defer;
}
void MediaSession::iterate (time_t currentRealTime, bool oneSecondElapsed) {
- L_D();
- d->executeBackgroundTasks(oneSecondElapsed);
CallSession::iterate(currentRealTime, oneSecondElapsed);
}
@@ -4813,20 +2228,19 @@ LinphoneStatus MediaSession::resume () {
d->broken = false;
/* Stop playing music immediately. If remote side is a conference it
* prevents the participants to hear it while the 200OK comes back. */
- if (d->audioStream)
- audio_stream_play(d->audioStream, nullptr);
- d->makeLocalMediaDescription();
+ Stream *as = d->getStreamsGroup().lookupMainStream(SalAudio);
+ if (as) as->stop();
+ d->setState(CallSession::State::Resuming,"Resuming");
+ d->makeLocalMediaDescription(true);
sal_media_description_set_dir(d->localDesc, SalStreamSendRecv);
- if (!getCore()->getCCore()->sip_conf.sdp_200_ack)
- d->op->setLocalMediaDescription(d->localDesc);
- else
+ if (getCore()->getCCore()->sip_conf.sdp_200_ack)
d->op->setLocalMediaDescription(nullptr);
string subject = "Call resuming";
if (d->getParams()->getPrivate()->getInConference() && !getCurrentParams()->getPrivate()->getInConference())
subject = "Conference";
if (d->op->update(subject.c_str(), false) != 0)
return -1;
- d->setState(CallSession::State::Resuming,"Resuming");
+
if (!d->getParams()->getPrivate()->getInConference() && d->listener)
d->listener->onSetCurrentSession(getSharedFromThis());
if (getCore()->getCCore()->sip_conf.sdp_200_ack) {
@@ -4861,27 +2275,23 @@ LinphoneStatus MediaSession::sendDtmfs (const std::string &dtmfs) {
}
void MediaSession::sendVfuRequest () {
-#ifdef VIDEO_ENABLED
L_D();
MediaSessionParams *curParams = getCurrentParams();
- if ((curParams->avpfEnabled() || curParams->getPrivate()->implicitRtcpFbEnabled())
- && d->videoStream && media_stream_get_state(&d->videoStream->ms) == MSStreamStarted) { // || sal_media_description_has_implicit_avpf((const SalMediaDescription *)call->resultdesc)
+ if ((curParams->avpfEnabled() || curParams->getPrivate()->implicitRtcpFbEnabled())) { // || sal_media_description_has_implicit_avpf((const SalMediaDescription *)call->resultdesc)
lInfo() << "Request Full Intra Request on CallSession [" << this << "]";
- video_stream_send_fir(d->videoStream);
+ d->getStreamsGroup().forEach<VideoControlInterface>([](VideoControlInterface *i){ i->sendVfuRequest(); });
} else if (getCore()->getCCore()->sip_conf.vfu_with_info) {
lInfo() << "Request SIP INFO FIR on CallSession [" << this << "]";
if (d->state == CallSession::State::StreamsRunning)
d->op->sendVfuRequest();
} else
lInfo() << "vfu request using sip disabled from config [sip,vfu_with_info]";
-#endif
}
void MediaSession::startIncomingNotification (bool notifyRinging) {
L_D();
- d->makeLocalMediaDescription();
- d->op->setLocalMediaDescription(d->localDesc);
+
SalMediaDescription *md = d->op->getFinalMediaDescription();
if (md) {
if (sal_media_description_empty(md) || linphone_core_incompatible_security(getCore()->getCCore(), md)) {
@@ -4900,30 +2310,21 @@ void MediaSession::startIncomingNotification (bool notifyRinging) {
int MediaSession::startInvite (const Address *destination, const string &subject, const Content *content) {
L_D();
linphone_core_stop_dtmf_stream(getCore()->getCCore());
- d->makeLocalMediaDescription();
if (!getCore()->getCCore()->ringstream && getCore()->getCCore()->sound_conf.play_sndcard && getCore()->getCCore()->sound_conf.capt_sndcard) {
/* Give a chance to set card prefered sampling frequency */
- if (d->localDesc->streams[0].max_rate > 0)
+ if (d->localDesc && d->localDesc->streams[0].max_rate > 0)
ms_snd_card_set_preferred_sample_rate(getCore()->getCCore()->sound_conf.play_sndcard, d->localDesc->streams[0].max_rate);
- if (!getCore()->getCCore()->use_files)
- audio_stream_prepare_sound(d->audioStream, getCore()->getCCore()->sound_conf.play_sndcard, getCore()->getCCore()->sound_conf.capt_sndcard);
- }
- if (!getCore()->getCCore()->sip_conf.sdp_200_ack) {
- /* We are offering, set local media description before sending the call */
- d->op->setLocalMediaDescription(d->localDesc);
+ d->getStreamsGroup().prepare();
}
+ d->op->setLocalMediaDescription(d->localDesc);
+
int result = CallSession::startInvite(destination, subject, content);
if (result < 0) {
if (d->state == CallSession::State::Error)
d->stopStreams();
return result;
}
- if (getCore()->getCCore()->sip_conf.sdp_200_ack) {
- // We are NOT offering, set local media description after sending the call so that we are ready to
- // process the remote offer when it will arrive.
- d->op->setLocalMediaDescription(d->localDesc);
- }
return result;
}
@@ -4933,21 +2334,20 @@ void MediaSession::startRecording () {
lError() << "MediaSession::startRecording(): no output file specified. Use MediaSessionParams::setRecordFilePath()";
return;
}
- if (d->audioStream && !d->getParams()->getPrivate()->getInConference())
- audio_stream_mixed_record_start(d->audioStream);
- d->recordActive = true;
+ AudioControlInterface * i = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ i->startRecording();
}
void MediaSession::stopRecording () {
L_D();
- if (d->audioStream && !d->getParams()->getPrivate()->getInConference())
- audio_stream_mixed_record_stop(d->audioStream);
- d->recordActive = false;
+ AudioControlInterface * i = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ i->stopRecording();
}
bool MediaSession::isRecording () {
L_D();
- return d->recordActive;
+ AudioControlInterface * i = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ return i->isRecording();
}
void MediaSession::terminateBecauseOfLostMedia () {
@@ -4966,12 +2366,14 @@ LinphoneStatus MediaSession::update (const MediaSessionParams *msp, const string
return -1;
if (d->getCurrentParams() == msp)
lWarning() << "CallSession::update() is given the current params, this is probably not what you intend to do!";
- d->iceAgent->checkSession(IR_Controlling, true);
if (msp) {
d->broken = false;
d->setState(nextState, "Updating call");
d->setParams(new MediaSessionParams(*msp));
- if (d->iceAgent->prepare(d->localDesc, false)) {
+ if (!d->getParams()->getPrivate()->getNoUserConsent())
+ d->makeLocalMediaDescription(true);
+
+ if (d->getStreamsGroup().prepare()) {
lInfo() << "Defer CallSession update to gather ICE candidates";
return 0;
}
@@ -4984,29 +2386,18 @@ LinphoneStatus MediaSession::update (const MediaSessionParams *msp, const string
const sound_config_t &soundConfig = getCore()->getCCore()->sound_conf;
const MSSndCard *captureCard = soundConfig.capt_sndcard;
const MSSndCard *playCard = soundConfig.lsd_card ? soundConfig.lsd_card : soundConfig.play_sndcard;
-
- if (captureCard != d->currentCaptureCard || playCard != d->currentPlayCard) {
- d->forceStreamsReconstruction = true;
+
+ MS2AudioStream *as = d->getStreamsGroup().lookupMainStreamInterface<MS2AudioStream>(SalAudio);
+ if (as && ((captureCard != as->getCurrentCaptureCard()) || playCard != as->getCurrentPlaybackCard())) {
//Ideally this should use the same logic as video (See video_stream_change_camera)
//I.E. reconstruct only ms2 graphs without destroying the streams.
//For now, we just stop and restart audio stream with new playback/capture card
+ as->stop();
d->updateStreams(d->resultDesc, d->state);
+ }else{
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) i->parametersChanged();
}
-
- #ifdef VIDEO_ENABLED
- else if (d->videoStream) {
- const LinphoneVideoDefinition *vdef = linphone_core_get_preferred_video_definition(getCore()->getCCore());
- MSVideoSize vsize;
- vsize.width = static_cast<int>(linphone_video_definition_get_width(vdef));
- vsize.height = static_cast<int>(linphone_video_definition_get_height(vdef));
- video_stream_set_sent_video_size(d->videoStream, vsize);
- video_stream_set_fps(d->videoStream, linphone_core_get_preferred_framerate(getCore()->getCCore()));
- if (d->cameraEnabled && (d->videoStream->cam != getCore()->getCCore()->video_conf.device))
- video_stream_change_camera(d->videoStream, getCore()->getCCore()->video_conf.device);
- else
- video_stream_update_video_params(d->videoStream);
- }
- #endif
}
return result;
}
@@ -5015,52 +2406,22 @@ LinphoneStatus MediaSession::update (const MediaSessionParams *msp, const string
void MediaSession::requestNotifyNextVideoFrameDecoded () {
L_D();
- if (d->videoStream && d->videoStream->ms.decoder)
- ms_filter_call_method_noarg(d->videoStream->ms.decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
-}
-
-void MediaSessionPrivate::snapshotTakenCb(void *userdata, struct _MSFilter *f, unsigned int id, void *arg) {
-#ifdef VIDEO_ENABLED
- L_Q();
- if (id == MS_JPEG_WRITER_SNAPSHOT_TAKEN) {
- const char *filepath = (const char *) arg;
- listener->onSnapshotTaken(q->getSharedFromThis(), filepath);
- }
-#endif
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) i->requestNotifyNextVideoFrameDecoded();
}
-#ifdef VIDEO_ENABLED
-static void snapshot_taken(void *userdata, struct _MSFilter *f, unsigned int id, void *arg) {
- MediaSessionPrivate *d = (MediaSessionPrivate *)userdata;
- d->snapshotTakenCb(userdata, f, id, arg);
-}
-#endif
LinphoneStatus MediaSession::takePreviewSnapshot (const string& file) {
-#ifdef VIDEO_ENABLED
L_D();
- if (d->videoStream && d->videoStream->local_jpegwriter) {
- ms_filter_clear_notify_callback(d->videoStream->jpegwriter);
- const char *filepath = file.empty() ? nullptr : file.c_str();
- ms_filter_add_notify_callback(d->videoStream->local_jpegwriter, snapshot_taken, d, TRUE);
- return ms_filter_call_method(d->videoStream->local_jpegwriter, MS_JPEG_WRITER_TAKE_SNAPSHOT, (void *)filepath);
- }
- lWarning() << "Cannot take local snapshot: no currently running video stream on this call";
-#endif
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) return i->takePreviewSnapshot(file);
return -1;
}
LinphoneStatus MediaSession::takeVideoSnapshot (const string& file) {
-#ifdef VIDEO_ENABLED
L_D();
- if (d->videoStream && d->videoStream->jpegwriter) {
- ms_filter_clear_notify_callback(d->videoStream->jpegwriter);
- const char *filepath = file.empty() ? nullptr : file.c_str();
- ms_filter_add_notify_callback(d->videoStream->jpegwriter, snapshot_taken, d, TRUE);
- return ms_filter_call_method(d->videoStream->jpegwriter, MS_JPEG_WRITER_TAKE_SNAPSHOT, (void *)filepath);
- }
- lWarning() << "Cannot take snapshot: no currently running video stream on this call";
-#endif
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) return i->takeVideoSnapshot(file);
return -1;
}
@@ -5070,97 +2431,49 @@ void MediaSession::zoomVideo (float zoomFactor, float *cx, float *cy) {
void MediaSession::zoomVideo (float zoomFactor, float cx, float cy) {
L_D();
- if (d->videoStream && d->videoStream->output) {
- if (zoomFactor < 1)
- zoomFactor = 1;
- float halfsize = 0.5f * 1.0f / zoomFactor;
- if ((cx - halfsize) < 0)
- cx = 0 + halfsize;
- if ((cx + halfsize) > 1)
- cx = 1 - halfsize;
- if ((cy - halfsize) < 0)
- cy = 0 + halfsize;
- if ((cy + halfsize) > 1)
- cy = 1 - halfsize;
- float zoom[3] = { zoomFactor, cx, cy };
- ms_filter_call_method(d->videoStream->output, MS_VIDEO_DISPLAY_ZOOM, &zoom);
- } else
- lWarning() << "Could not apply zoom: video output wasn't activated";
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) i->zoomVideo(zoomFactor, cx, cy);
}
// -----------------------------------------------------------------------------
bool MediaSession::cameraEnabled () const {
L_D();
- return d->cameraEnabled;
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) return i->cameraEnabled();
+ return false;
}
-bool MediaSession::echoCancellationEnabled () const {
+void MediaSession::enableCamera (bool value) {
L_D();
- if (!d->audioStream || !d->audioStream->ec)
- return !!linphone_core_echo_cancellation_enabled(getCore()->getCCore());
-
- bool val;
- ms_filter_call_method(d->audioStream->ec, MS_ECHO_CANCELLER_GET_BYPASS_MODE, &val);
- return !val;
+ VideoControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (i) i->enableCamera(value);
}
-bool MediaSession::echoLimiterEnabled () const {
+bool MediaSession::echoCancellationEnabled () const {
L_D();
- if (d->audioStream)
- return d->audioStream->el_type !=ELInactive;
- return !!linphone_core_echo_limiter_enabled(getCore()->getCCore());
+ AudioControlInterface * i = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ return i ? i->echoCancellationEnabled() : false;
}
-void MediaSession::enableCamera (bool value) {
-#ifdef VIDEO_ENABLED
+void MediaSession::enableEchoCancellation (bool value) {
L_D();
- d->cameraEnabled = value;
- switch (d->state) {
- case CallSession::State::StreamsRunning:
- case CallSession::State::OutgoingEarlyMedia:
- case CallSession::State::IncomingEarlyMedia:
- case CallSession::State::Connected:
- if (d->videoStream && video_stream_started(d->videoStream) && (video_stream_get_camera(d->videoStream) != d->getVideoDevice())) {
- string currentCam = video_stream_get_camera(d->videoStream) ? ms_web_cam_get_name(video_stream_get_camera(d->videoStream)) : "NULL";
- string newCam = d->getVideoDevice() ? ms_web_cam_get_name(d->getVideoDevice()) : "NULL";
- lInfo() << "Switching video cam from [" << currentCam << "] to [" << newCam << "] on CallSession [" << this << "]";
- video_stream_change_camera(d->videoStream, d->getVideoDevice());
- }
- break;
- default:
- break;
- }
-#endif
+ AudioControlInterface * i = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (i) i->enableEchoCancellation(value);
}
-void MediaSession::enableEchoCancellation (bool value) {
- L_D();
- if (d->audioStream && d->audioStream->ec) {
- bool bypassMode = !value;
- ms_filter_call_method(d->audioStream->ec, MS_ECHO_CANCELLER_SET_BYPASS_MODE, &bypassMode);
- }
+bool MediaSession::echoLimiterEnabled () const {
+ lWarning() << "MediaSession::echoLimiterEnabled() unimplemented.";
+ return false;
}
void MediaSession::enableEchoLimiter (bool value) {
- L_D();
- if (d->audioStream) {
- if (value) {
- string type = lp_config_get_string(linphone_core_get_config(getCore()->getCCore()), "sound", "el_type", "mic");
- if (type == "mic")
- audio_stream_enable_echo_limiter(d->audioStream, ELControlMic);
- else if (type == "full")
- audio_stream_enable_echo_limiter(d->audioStream, ELControlFull);
- } else
- audio_stream_enable_echo_limiter(d->audioStream, ELInactive);
- }
+ lWarning() << "MediaSession::enableEchoLimiter() unimplemented.";
}
bool MediaSession::getAllMuted () const {
L_D();
- if (d->audioStream && d->videoStream) return d->audioMuted && d->videoMuted;
- if (d->audioStream) return d->audioMuted;
- return d->videoMuted;
+ return d->getStreamsGroup().isMuted();
}
LinphoneCallStats * MediaSession::getAudioStats () const {
@@ -5169,23 +2482,17 @@ LinphoneCallStats * MediaSession::getAudioStats () const {
string MediaSession::getAuthenticationToken () const {
L_D();
- return d->authToken;
+ return d->getStreamsGroup().getAuthenticationToken();
}
bool MediaSession::getAuthenticationTokenVerified () const {
L_D();
- return d->authTokenVerified;
+ return d->getStreamsGroup().getAuthenticationTokenVerified();
}
float MediaSession::getAverageQuality () const {
L_D();
- float audioRating = -1.f;
- float videoRating = -1.f;
- if (d->audioStream)
- audioRating = media_stream_get_average_quality_rating(&d->audioStream->ms) / 5.0f;
- if (d->videoStream)
- videoRating = media_stream_get_average_quality_rating(&d->videoStream->ms) / 5.0f;
- return MediaSessionPrivate::aggregateQualityRatings(audioRating, videoRating);
+ return d->getStreamsGroup().getAverageQuality();
}
MediaSessionParams * MediaSession::getCurrentParams () const {
@@ -5196,13 +2503,7 @@ MediaSessionParams * MediaSession::getCurrentParams () const {
float MediaSession::getCurrentQuality () const {
L_D();
- float audioRating = -1.f;
- float videoRating = -1.f;
- if (d->audioStream)
- audioRating = media_stream_get_quality_rating(&d->audioStream->ms) / 5.0f;
- if (d->videoStream)
- videoRating = media_stream_get_quality_rating(&d->videoStream->ms) / 5.0f;
- return MediaSessionPrivate::aggregateQualityRatings(audioRating, videoRating);
+ return d->getStreamsGroup().getCurrentQuality();
}
const MediaSessionParams * MediaSession::getMediaParams () const {
@@ -5212,46 +2513,95 @@ const MediaSessionParams * MediaSession::getMediaParams () const {
RtpTransport * MediaSession::getMetaRtcpTransport (int streamIndex) const {
L_D();
- if ((streamIndex < 0) || (streamIndex >= getStreamCount()))
+ MS2Stream *s = dynamic_cast<MS2Stream*>(d->getStreamsGroup().getStream(streamIndex));
+ if (!s){
+ lError() << "MediaSession::getMetaRtcpTransport(): no stream with index " << streamIndex;
return nullptr;
- RtpTransport *metaRtp;
- RtpTransport *metaRtcp;
- rtp_session_get_transports(d->sessions[streamIndex].rtp_session, &metaRtp, &metaRtcp);
- return metaRtcp;
+ }
+ return s->getMetaRtpTransports().second;
}
RtpTransport * MediaSession::getMetaRtpTransport (int streamIndex) const {
L_D();
- if ((streamIndex < 0) || (streamIndex >= getStreamCount()))
+ MS2Stream *s = dynamic_cast<MS2Stream*>(d->getStreamsGroup().getStream(streamIndex));
+ if (!s){
+ lError() << "MediaSession::getMetaRtcpTransport(): no stream with index " << streamIndex;
return nullptr;
- RtpTransport *metaRtp;
- RtpTransport *metaRtcp;
- rtp_session_get_transports(d->sessions[streamIndex].rtp_session, &metaRtp, &metaRtcp);
- return metaRtp;
+ }
+ return s->getMetaRtpTransports().first;
}
float MediaSession::getMicrophoneVolumeGain () const {
L_D();
- if (d->audioStream)
- return audio_stream_get_sound_card_input_gain(d->audioStream);
- else {
+ AudioControlInterface *iface = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (iface){
+ return iface->getMicGain();
+ } else {
lError() << "Could not get record volume: no audio stream";
return -1.0f;
}
}
+void MediaSession::setMicrophoneVolumeGain (float value) {
+ L_D();
+ AudioControlInterface *iface = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (iface)
+ iface->setMicGain(value);
+ else
+ lError() << "Could not set record volume: no audio stream";
+}
+
+float MediaSession::getSpeakerVolumeGain () const {
+ L_D();
+ AudioControlInterface *iface = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (iface)
+ return iface->getSpeakerGain();
+ else {
+ lError() << "Could not get playback volume: no audio stream";
+ return -1.0f;
+ }
+}
+
+void MediaSession::setSpeakerVolumeGain (float value) {
+ L_D();
+ AudioControlInterface *iface = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (iface)
+ iface->setSpeakerGain(value);
+ else
+ lError() << "Could not set playback volume: no audio stream";
+}
+
void * MediaSession::getNativeVideoWindowId () const {
L_D();
- if (d->videoWindowId) {
- /* The video id was previously set by the app */
- return d->videoWindowId;
+ auto iface = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (iface) {
+ iface->getNativeWindowId();
}
-#ifdef VIDEO_ENABLED
- else if (d->videoStream) {
- /* It was not set but we want to get the one automatically created by mediastreamer2 (desktop versions only) */
- return video_stream_get_native_window_id(d->videoStream);
+ return nullptr;
+}
+
+void MediaSession::setNativeVideoWindowId (void *id) {
+ L_D();
+ auto iface = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (iface) {
+ iface->setNativeWindowId(id);
+ }
+}
+
+void MediaSession::setNativePreviewWindowId(void *id){
+ L_D();
+ auto iface = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (iface) {
+ iface->setNativePreviewWindowId(id);
+ }
+}
+
+void * MediaSession::getNativePreviewVideoWindowId () const{
+ L_D();
+ auto iface = d->getStreamsGroup().lookupMainStreamInterface<VideoControlInterface>(SalVideo);
+ if (iface) {
+ iface->getNativePreviewWindowId();
}
-#endif
return nullptr;
}
@@ -5262,20 +2612,17 @@ const CallSessionParams * MediaSession::getParams () const {
float MediaSession::getPlayVolume () const {
L_D();
- if (d->audioStream && d->audioStream->volrecv) {
- float vol = 0;
- ms_filter_call_method(d->audioStream->volrecv, MS_VOLUME_GET, &vol);
- return vol;
- }
+ AudioControlInterface *iface = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (iface) return iface->getPlayVolume();
return LINPHONE_VOLUME_DB_LOWEST;
}
float MediaSession::getRecordVolume () const {
L_D();
- if (d->audioStream && d->audioStream->volsend && !d->microphoneMuted && (d->state == CallSession::State::StreamsRunning)) {
- float vol = 0;
- ms_filter_call_method(d->audioStream->volsend, MS_VOLUME_GET, &vol);
- return vol;
+
+ if (d->state == CallSession::State::StreamsRunning){
+ AudioControlInterface *iface = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (iface) return iface->getRecordVolume();
}
return LINPHONE_VOLUME_DB_LOWEST;
}
@@ -5284,109 +2631,90 @@ const MediaSessionParams * MediaSession::getRemoteParams () {
L_D();
if (d->op){
SalMediaDescription *md = d->op->getRemoteMediaDescription();
+ MediaSessionParams * params = nullptr;
if (md) {
- d->setRemoteParams(new MediaSessionParams());
- unsigned int nbAudioStreams = sal_media_description_nb_active_streams_of_type(md, SalAudio);
- for (unsigned int i = 0; i < nbAudioStreams; i++) {
- SalStreamDescription *sd = sal_media_description_get_active_stream_of_type(md, SalAudio, i);
- if (sal_stream_description_has_srtp(sd))
- d->getRemoteParams()->setMediaEncryption(LinphoneMediaEncryptionSRTP);
- }
- unsigned int nbVideoStreams = sal_media_description_nb_active_streams_of_type(md, SalVideo);
- for (unsigned int i = 0; i < nbVideoStreams; i++) {
- SalStreamDescription *sd = sal_media_description_get_active_stream_of_type(md, SalVideo, i);
- if (sal_stream_description_active(sd))
- d->getRemoteParams()->enableVideo(true);
- if (sal_stream_description_has_srtp(sd))
- d->getRemoteParams()->setMediaEncryption(LinphoneMediaEncryptionSRTP);
- }
- unsigned int nbTextStreams = sal_media_description_nb_active_streams_of_type(md, SalText);
- for (unsigned int i = 0; i < nbTextStreams; i++) {
- SalStreamDescription *sd = sal_media_description_get_active_stream_of_type(md, SalText, i);
- if (sal_stream_description_has_srtp(sd))
- d->getRemoteParams()->setMediaEncryption(LinphoneMediaEncryptionSRTP);
- d->getRemoteParams()->enableRealtimeText(true);
- }
- if (!d->getRemoteParams()->videoEnabled()) {
+ SalStreamDescription *sd;
+ params = new MediaSessionParams();
+
+ if (d->mainAudioStreamIndex != -1 && d->mainAudioStreamIndex < md->nb_streams){
+ sd = &md->streams[d->mainAudioStreamIndex];
+ params->enableAudio(sal_stream_description_enabled(sd));
+ params->setMediaEncryption(sal_stream_description_has_srtp(sd) ? LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone);
+ params->getPrivate()->setCustomSdpMediaAttributes(LinphoneStreamTypeAudio, md->streams[d->mainAudioStreamIndex].custom_sdp_attributes);
+ }else params->enableAudio(false);
+
+ if (d->mainVideoStreamIndex != -1 && d->mainVideoStreamIndex < md->nb_streams){
+ sd = &md->streams[d->mainVideoStreamIndex];
+ params->enableVideo(sal_stream_description_enabled(sd));
+ params->setMediaEncryption(sal_stream_description_has_srtp(sd) ? LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone);
+ params->getPrivate()->setCustomSdpMediaAttributes(LinphoneStreamTypeVideo, md->streams[d->mainVideoStreamIndex].custom_sdp_attributes);
+ }else params->enableVideo(false);
+
+ if (d->mainTextStreamIndex != -1 && d->mainTextStreamIndex < md->nb_streams){
+ sd = &md->streams[d->mainTextStreamIndex];
+ params->enableRealtimeText(sal_stream_description_enabled(sd));
+ params->setMediaEncryption(sal_stream_description_has_srtp(sd) ? LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone);
+ params->getPrivate()->setCustomSdpMediaAttributes(LinphoneStreamTypeText, md->streams[d->mainTextStreamIndex].custom_sdp_attributes);
+ }else params->enableRealtimeText(false);
+
+ if (!params->videoEnabled()) {
if ((md->bandwidth > 0) && (md->bandwidth <= linphone_core_get_edge_bw(getCore()->getCCore())))
- d->getRemoteParams()->enableLowBandwidth(true);
+ params->enableLowBandwidth(true);
}
if (md->name[0] != '\0')
- d->getRemoteParams()->setSessionName(md->name);
-
- d->getRemoteParams()->getPrivate()->setCustomSdpAttributes(md->custom_sdp_attributes);
- d->getRemoteParams()->getPrivate()->setCustomSdpMediaAttributes(LinphoneStreamTypeAudio, md->streams[d->mainAudioStreamIndex].custom_sdp_attributes);
- d->getRemoteParams()->getPrivate()->setCustomSdpMediaAttributes(LinphoneStreamTypeVideo, md->streams[d->mainVideoStreamIndex].custom_sdp_attributes);
- d->getRemoteParams()->getPrivate()->setCustomSdpMediaAttributes(LinphoneStreamTypeText, md->streams[d->mainTextStreamIndex].custom_sdp_attributes);
+ params->setSessionName(md->name);
+ params->getPrivate()->setCustomSdpAttributes(md->custom_sdp_attributes);
+ params->enableRtpBundle(md->bundles != nullptr);
}
const SalCustomHeader *ch = d->op->getRecvCustomHeaders();
if (ch) {
- /* Instanciate a remote_params only if a SIP message was received before (custom headers indicates this) */
- if (!d->remoteParams)
- d->setRemoteParams(new MediaSessionParams());
- d->getRemoteParams()->getPrivate()->setCustomHeaders(ch);
+ if (!params) params = new MediaSessionParams();
+ params->getPrivate()->setCustomHeaders(ch);
}
-
- const list<Content> additionnalContents = d->op->getAdditionalRemoteBodies();
- for (auto& content : additionnalContents)
- d->remoteParams->addCustomContent(content);
-
- return d->getRemoteParams();
+ const list<Content> &additionnalContents = d->op->getAdditionalRemoteBodies();
+ for (auto& content : additionnalContents){
+ if (!params) params = new MediaSessionParams();
+ params->addCustomContent(content);
+ }
+ d->setRemoteParams(params);
+ return params;
}
return nullptr;
}
-float MediaSession::getSpeakerVolumeGain () const {
- L_D();
- if (d->audioStream)
- return audio_stream_get_sound_card_output_gain(d->audioStream);
- else {
- lError() << "Could not get playback volume: no audio stream";
- return -1.0f;
- }
-}
-
LinphoneCallStats * MediaSession::getStats (LinphoneStreamType type) const {
L_D();
if (type == LinphoneStreamTypeUnknown)
return nullptr;
LinphoneCallStats *stats = nullptr;
- LinphoneCallStats *statsCopy = _linphone_call_stats_new();
- if (type == LinphoneStreamTypeAudio)
- stats = d->audioStats;
- else if (type == LinphoneStreamTypeVideo)
- stats = d->videoStats;
- else if (type == LinphoneStreamTypeText)
- stats = d->textStats;
- MediaStream *ms = d->getMediaStream(type);
- if (ms && stats)
- linphone_call_stats_update(stats, ms);
- _linphone_call_stats_clone(statsCopy, stats);
+ LinphoneCallStats *statsCopy = nullptr;
+ Stream *s = d->getStream(type);
+ if (s && (stats = s->getStats())) {
+ statsCopy = (LinphoneCallStats*) belle_sip_object_clone((belle_sip_object_t*)stats);
+ }
return statsCopy;
}
int MediaSession::getStreamCount () const {
- /* TODO: Revisit when multiple media streams will be implemented */
-#ifdef VIDEO_ENABLED
- if (getCurrentParams()->realtimeTextEnabled())
- return 3;
- return 2;
-#else
- if (getCurrentParams()->realtimeTextEnabled())
- return 2;
- return 1;
-#endif
+ L_D();
+ return (int)d->getStreamsGroup().size();
}
MSFormatType MediaSession::getStreamType (int streamIndex) const {
L_D();
- /* TODO: Revisit when multiple media streams will be implemented */
- if (streamIndex == d->mainVideoStreamIndex)
- return MSVideo;
- else if (streamIndex == d->mainTextStreamIndex)
- return MSText;
- else if (streamIndex == d->mainAudioStreamIndex)
- return MSAudio;
+ Stream *s = d->getStreamsGroup().getStream(streamIndex);
+ if (s){
+ switch(s->getType()){
+ case SalAudio:
+ return MSAudio;
+ case SalVideo:
+ return MSVideo;
+ case SalText:
+ return MSText;
+ case SalOther:
+ break;
+ }
+ }
return MSUnknownMedia;
}
@@ -5400,35 +2728,25 @@ LinphoneCallStats * MediaSession::getVideoStats () const {
bool MediaSession::mediaInProgress () const {
L_D();
- if ((linphone_call_stats_get_ice_state(d->audioStats) == LinphoneIceStateInProgress)
- || (linphone_call_stats_get_ice_state(d->videoStats) == LinphoneIceStateInProgress)
- || (linphone_call_stats_get_ice_state(d->textStats) == LinphoneIceStateInProgress))
- return true;
- /* TODO: could check zrtp state */
+ for(auto &stream : d->getStreamsGroup().getStreams()){
+ LinphoneCallStats *stats = stream->getStats();
+ if (stats && linphone_call_stats_get_ice_state(stats) == LinphoneIceStateInProgress){
+ return true;
+ }
+ }
return false;
}
void MediaSession::setAudioRoute (LinphoneAudioRoute route) {
L_D();
- if (d->audioStream)
- audio_stream_set_audio_route(d->audioStream, (MSAudioRoute)route);
+ AudioControlInterface *i = d->getStreamsGroup().lookupMainStreamInterface<AudioControlInterface>(SalAudio);
+ if (i) i->setRoute(route);
}
void MediaSession::setAuthenticationTokenVerified (bool value) {
L_D();
- if (!d->audioStream || !media_stream_started(&d->audioStream->ms)) {
- lError() << "MediaSession::setAuthenticationTokenVerified(): No audio stream or not started";
- return;
- }
- if (!d->audioStream->ms.sessions.zrtp_context) {
- lError() << "MediaSession::setAuthenticationTokenVerified(): No zrtp context";
- return;
- }
- // SAS verified
- if (value) {
- ms_zrtp_sas_verified(d->audioStream->ms.sessions.zrtp_context);
- } else { // SAS rejected
- ms_zrtp_sas_reset_verified(d->audioStream->ms.sessions.zrtp_context);
+ d->getStreamsGroup().setAuthTokenVerified(value);
+ if (!value) {
char *peerDeviceId = nullptr;
auto encryptionEngine = getCore()->getEncryptionEngine();
if (encryptionEngine) { //inform lime that zrtp no longuer guaranty the trust
@@ -5438,41 +2756,26 @@ void MediaSession::setAuthenticationTokenVerified (bool value) {
ms_free(peerDeviceId);
}
}
- d->authTokenVerified = value;
d->propagateEncryptionChanged();
}
-void MediaSession::setMicrophoneVolumeGain (float value) {
- L_D();
- if(d->audioStream)
- audio_stream_set_sound_card_input_gain(d->audioStream, value);
- else
- lError() << "Could not set record volume: no audio stream";
-}
-
-void MediaSession::setNativeVideoWindowId (void *id) {
- L_D();
- d->videoWindowId = id;
-#ifdef VIDEO_ENABLED
- if (d->videoStream)
- video_stream_set_native_window_id(d->videoStream, id);
-#endif
-}
-
void MediaSession::setParams (const MediaSessionParams *msp) {
L_D();
- if ((d->state == CallSession::State::OutgoingInit) || (d->state == CallSession::State::IncomingReceived))
- d->setParams(msp ? new MediaSessionParams(*msp) : nullptr);
- else
- lError() << "MediaSession::setParams(): Invalid state %s", Utils::toString(d->state);
+
+ switch(d->state){
+ case CallSession::State::OutgoingInit:
+ case CallSession::State::IncomingReceived:
+ d->setParams(msp ? new MediaSessionParams(*msp) : nullptr);
+ // Update the local media description.
+ d->makeLocalMediaDescription(d->state == CallSession::State::OutgoingInit ?
+ !getCore()->getCCore()->sip_conf.sdp_200_ack : false);
+ break;
+ default:
+ lError() << "MediaSession::setParams(): Invalid state %s", Utils::toString(d->state);
+ break;
+ }
}
-void MediaSession::setSpeakerVolumeGain (float value) {
- L_D();
- if (d->audioStream)
- audio_stream_set_sound_card_output_gain(d->audioStream, value);
- else
- lError() << "Could not set playback volume: no audio stream";
-}
+
LINPHONE_END_NAMESPACE
diff --git a/src/conference/session/media-session.h b/src/conference/session/media-session.h
index ff39c48c1ec89e13cfa4929cc88d415984102301..52f465ead44324e2feec076ed10980cdae6e7ee2 100644
--- a/src/conference/session/media-session.h
+++ b/src/conference/session/media-session.h
@@ -38,6 +38,8 @@ class LINPHONE_PUBLIC MediaSession : public CallSession {
friend class CallPrivate;
friend class IceAgent;
friend class ToneManager;
+ friend class Stream;
+ friend class StreamsGroup;
public:
MediaSession (const std::shared_ptr<Core> &core, std::shared_ptr<Participant> me, const CallSessionParams *params, CallSessionListener *listener);
@@ -90,6 +92,7 @@ public:
RtpTransport * getMetaRtpTransport (int streamIndex) const;
float getMicrophoneVolumeGain () const;
void * getNativeVideoWindowId () const;
+ void * getNativePreviewVideoWindowId () const;
const CallSessionParams *getParams () const override;
float getPlayVolume () const;
float getRecordVolume () const;
@@ -105,9 +108,9 @@ public:
void setAuthenticationTokenVerified (bool value);
void setMicrophoneVolumeGain (float value);
void setNativeVideoWindowId (void *id);
+ void setNativePreviewWindowId (void *id);
void setParams (const MediaSessionParams *msp);
void setSpeakerVolumeGain (float value);
-
private:
L_DECLARE_PRIVATE(MediaSession);
L_DISABLE_COPY(MediaSession);
diff --git a/src/conference/session/ms2-stream.cpp b/src/conference/session/ms2-stream.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..2fbac7d2afb059e78558cf9fb2f9a161bf407c8d
--- /dev/null
+++ b/src/conference/session/ms2-stream.cpp
@@ -0,0 +1,1104 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <bctoolbox/defs.h>
+
+#include "ms2-streams.h"
+#include "media-session.h"
+#include "media-session-p.h"
+#include "core/core.h"
+#include "c-wrapper/c-wrapper.h"
+#include "call/call.h"
+#include "call/call-p.h"
+#include "conference/participant.h"
+#include "utils/payload-type-handler.h"
+#include "conference/params/media-session-params-p.h"
+#include "nat/ice-service.h"
+
+#include "linphone/core.h"
+
+
+using namespace::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+/*
+ * MS2Stream implementation
+ */
+
+MS2Stream::MS2Stream(StreamsGroup &sg, const OfferAnswerContext ¶ms) : Stream(sg, params){
+ memset(&mSessions, 0, sizeof(mSessions));
+ initMulticast(params);
+ mStats = _linphone_call_stats_new();
+ _linphone_call_stats_set_type(mStats, (LinphoneStreamType)getType());
+ _linphone_call_stats_set_received_rtcp(mStats, nullptr);
+ _linphone_call_stats_set_sent_rtcp(mStats, nullptr);
+ _linphone_call_stats_set_ice_state(mStats, LinphoneIceStateNotActivated);
+}
+
+void MS2Stream::removeFromBundle(){
+ if (mRtpBundle){
+ rtp_bundle_remove_session(mRtpBundle, mSessions.rtp_session);
+ if (mOwnsBundle){
+ RtpBundle *bundle = mRtpBundle;
+ getGroup().addPostRenderHook( [bundle](){
+ rtp_bundle_delete(bundle);
+ });
+ mOwnsBundle = false;
+ getMediaSessionPrivate().getCurrentParams()->enableRtpBundle(false);
+ }
+ mRtpBundle = nullptr;
+ mBundleOwner = nullptr;
+ }
+}
+
+void MS2Stream::initRtpBundle(const OfferAnswerContext ¶ms){
+ int index = sal_media_description_get_index_of_transport_owner(params.resultMediaDescription, params.resultStreamDescription);
+ if (index == -1) {
+ lInfo() << *this << " is not part of any bundle";
+ removeFromBundle();
+ return ; /*No bundle to handle */
+ }
+
+ mBundleOwner = dynamic_cast<MS2Stream*>(getGroup().getStream((size_t)index));
+ if (!mBundleOwner){
+ lError() << "Could not locate the stream owning the bundle's transport.";
+ removeFromBundle();
+ return;
+ }
+ RtpBundle * bundle = mBundleOwner->createOrGetRtpBundle(params.resultStreamDescription);
+ if (bundle && mBundleOwner != this && mRtpBundle == nullptr){
+ lInfo() << "Stream " << *this << " added to rtp bundle " << bundle << " with mid '" << params.resultStreamDescription->mid << "'";
+ rtp_bundle_add_session(bundle, params.resultStreamDescription->mid, mSessions.rtp_session);
+ mRtpBundle = bundle;
+ mOwnsBundle = false;
+ getMediaSessionPrivate().getCurrentParams()->enableRtpBundle(true);
+ }
+
+ // It is necessary to call this function after adding the session to the bundle so the SDES contains the MID item
+ string userAgent = linphone_core_get_user_agent(getCCore());
+ rtp_session_set_source_description(mSessions.rtp_session, getMediaSessionPrivate().getMe()->getAddress().asString().c_str(), NULL, NULL, NULL, NULL, userAgent.c_str(), NULL);
+}
+
+RtpBundle *MS2Stream::createOrGetRtpBundle(const SalStreamDescription *sd){
+ if (!mRtpBundle){
+ mRtpBundle = rtp_bundle_new();
+ lInfo() << "Stream " << *this << " with mid '" << sd->mid << "'is the owner of rtp bundle " << mRtpBundle;
+ rtp_bundle_add_session(mRtpBundle, sd->mid, mSessions.rtp_session);
+ rtp_bundle_set_mid_extension_id(mRtpBundle, sd->mid_rtp_ext_header_id);
+ mOwnsBundle = true;
+ getMediaSessionPrivate().getCurrentParams()->enableRtpBundle(true);
+ }
+ return mRtpBundle;
+}
+
+void MS2Stream::setIceCheckList(IceCheckList *cl){
+ mIceCheckList = cl;
+ MediaStream *stream = getMediaStream();
+ if (stream){
+ rtp_session_set_pktinfo(mSessions.rtp_session, cl != nullptr);
+ rtp_session_set_symmetric_rtp(mSessions.rtp_session, (cl == nullptr) ? linphone_core_symmetric_rtp_enabled(getCCore()) : false);
+ media_stream_set_ice_check_list(stream, cl);
+ }
+ if (!cl){
+ updateIceInStats();
+ }
+}
+
+string MS2Stream::getBindIp(){
+ string bindIp = lp_config_get_string(linphone_core_get_config(getCCore()), "rtp", "bind_address", "");
+
+ if (!mPortConfig.multicastIp.empty()){
+ if (mPortConfig.multicastRole == SalMulticastSender) {
+ /* As multicast sender, we must decide a local interface to use to send multicast, and bind to it */
+ char multicastBindIp[LINPHONE_IPADDR_SIZE] = {0};
+ linphone_core_get_local_ip_for((mPortConfig.multicastIp.find_first_of(':') == string::npos) ? AF_INET : AF_INET6, nullptr, multicastBindIp);
+ bindIp = mPortConfig.multicastBindIp = multicastBindIp;
+ } else {
+ /* Otherwise we shall use an address family of the same family of the multicast address, because
+ * dual stack socket and multicast don't work well on Mac OS (linux is OK, as usual). */
+ bindIp = (mPortConfig.multicastIp.find_first_of(':') == string::npos) ? "0.0.0.0" : "::0";
+ }
+ }else if (bindIp.empty()){
+ /*If ipv6 is not enabled, listen to 0.0.0.0. The default behavior of mediastreamer when no IP is passed is to try ::0, and in
+ * case of failure try 0.0.0.0 . But we don't want this if IPv6 is explicitely disabled.*/
+ if (!linphone_core_ipv6_enabled(getCCore())){
+ bindIp = "0.0.0.0";
+ }
+ }
+ return bindIp;
+}
+
+void MS2Stream::fillLocalMediaDescription(OfferAnswerContext & ctx){
+ SalStreamDescription *localDesc = ctx.localStreamDescription;
+ strncpy(localDesc->rtp_addr, getPublicIp().c_str(), sizeof(localDesc->rtp_addr) - 1);
+ strncpy(localDesc->rtcp_addr, getPublicIp().c_str(), sizeof(localDesc->rtcp_addr) -1);
+
+ if (localDesc->rtp_port == SAL_STREAM_DESCRIPTION_PORT_TO_BE_DETERMINED && localDesc->payloads != nullptr){
+ /* Don't fill ports if no codecs are defined. The stream is not valid and should be disabled.*/
+ localDesc->rtp_port = mPortConfig.rtpPort;
+ localDesc->rtcp_port = mPortConfig.rtcpPort;
+ }
+ if (!isTransportOwner()){
+ /* A secondary stream part of a bundle must set port to zero and add the bundle-only attribute. */
+ localDesc->rtp_port = 0;
+ localDesc->bundle_only = TRUE;
+ }
+
+ localDesc->rtp_ssrc = rtp_session_get_send_ssrc(mSessions.rtp_session);
+
+ if (linphone_core_media_encryption_supported(getCCore(), LinphoneMediaEncryptionZRTP)) {
+ /* set the hello hash */
+ if (mSessions.zrtp_context) {
+ ms_zrtp_getHelloHash(mSessions.zrtp_context, localDesc->zrtphash, 128);
+ /* Turn on the flag to use it if ZRTP is set */
+ localDesc->haveZrtpHash = (getMediaSessionPrivate().getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP);
+ } else
+ localDesc->haveZrtpHash = 0;
+ }
+ if (sal_stream_description_has_dtls(localDesc)) {
+ /* Get the self fingerprint from call (it's computed at stream init) */
+ strncpy(localDesc->dtls_fingerprint, mDtlsFingerPrint.c_str(), sizeof(localDesc->dtls_fingerprint) - 1);
+ /* If we are offering, SDP will have actpass setup attribute when role is unset, if we are responding the result mediadescription will be set to SalDtlsRoleIsClient */
+ localDesc->dtls_role = SalDtlsRoleUnset;
+ } else {
+ localDesc->dtls_fingerprint[0] = '\0';
+ localDesc->dtls_role = SalDtlsRoleInvalid;
+ }
+ /* In case we were offered multicast, we become multicast receiver. The local media description must reflect this. */
+ localDesc->multicast_role = mPortConfig.multicastRole;
+ Stream::fillLocalMediaDescription(ctx);
+}
+
+void MS2Stream::refreshSockets(){
+ rtp_session_refresh_sockets(mSessions.rtp_session);
+}
+
+void MS2Stream::initMulticast(const OfferAnswerContext ¶ms) {
+ mPortConfig.multicastRole = params.localStreamDescription->multicast_role;
+
+ if (mPortConfig.multicastRole == SalMulticastReceiver){
+ mPortConfig.multicastIp = params.remoteStreamDescription->rtp_addr;
+ mPortConfig.rtpPort = params.remoteStreamDescription->rtp_port;
+ mPortConfig.rtcpPort = 0; /*RTCP deactivated in multicast*/
+ }
+
+ lInfo() << *this << ": multicast role is ["
+ << sal_multicast_role_to_string(mPortConfig.multicastRole) << "]";
+}
+
+void MS2Stream::configureRtpSessionForRtcpFb (const OfferAnswerContext ¶ms) {
+ if (getType() != SalAudio && getType() != SalVideo) return; //No AVPF for other than audio/video
+
+ rtp_session_enable_avpf_feature(mSessions.rtp_session, ORTP_AVPF_FEATURE_GENERIC_NACK, !!params.resultStreamDescription->rtcp_fb.generic_nack_enabled);
+ rtp_session_enable_avpf_feature(mSessions.rtp_session, ORTP_AVPF_FEATURE_TMMBR, !!params.resultStreamDescription->rtcp_fb.tmmbr_enabled);
+}
+
+void MS2Stream::configureRtpSessionForRtcpXr(const OfferAnswerContext ¶ms) {
+ OrtpRtcpXrConfiguration currentConfig;
+ const OrtpRtcpXrConfiguration *remoteConfig = ¶ms.remoteStreamDescription->rtcp_xr;
+ if (params.localStreamDescription->dir == SalStreamInactive)
+ return;
+ else if (params.localStreamDescription->dir == SalStreamRecvOnly) {
+ /* Use local config for unilateral parameters and remote config for collaborative parameters */
+ memcpy(¤tConfig, ¶ms.localStreamDescription->rtcp_xr, sizeof(currentConfig));
+ currentConfig.rcvr_rtt_mode = remoteConfig->rcvr_rtt_mode;
+ currentConfig.rcvr_rtt_max_size = remoteConfig->rcvr_rtt_max_size;
+ } else
+ memcpy(¤tConfig, remoteConfig, sizeof(currentConfig));
+
+ rtp_session_configure_rtcp_xr(mSessions.rtp_session, ¤tConfig);
+}
+
+void MS2Stream::configureAdaptiveRateControl (const OfferAnswerContext ¶ms) {
+ if (getState() == Stream::Running){
+ return; // If stream is already running, these things are not expected to change.
+ }
+ bool videoWillBeUsed = false;
+ MediaStream *ms = getMediaStream();
+ const SalStreamDescription *vstream = sal_media_description_find_best_stream(const_cast<SalMediaDescription*>(params.resultMediaDescription), SalVideo);
+ if (vstream && (vstream->dir != SalStreamInactive) && vstream->payloads) {
+ /* When video is used, do not make adaptive rate control on audio, it is stupid */
+ videoWillBeUsed = true;
+ }
+ bool enabled = !!linphone_core_adaptive_rate_control_enabled(getCCore());
+ if (!enabled) {
+ media_stream_enable_adaptive_bitrate_control(ms, false);
+ return;
+ }
+ bool isAdvanced = true;
+ string algo = linphone_core_get_adaptive_rate_algorithm(getCCore());
+ if (algo == "basic")
+ isAdvanced = false;
+ else if (algo == "advanced")
+ isAdvanced = true;
+
+ if (isAdvanced && !params.resultStreamDescription->rtcp_fb.tmmbr_enabled) {
+ lWarning() << "Advanced adaptive rate control requested but avpf-tmmbr is not activated in this stream. Reverting to basic rate control instead";
+ isAdvanced = false;
+ }
+ if (isAdvanced) {
+ lInfo() << "Setting up advanced rate control";
+ ms_bandwidth_controller_add_stream(getCCore()->bw_controller, ms);
+ media_stream_enable_adaptive_bitrate_control(ms, false);
+ } else {
+ media_stream_set_adaptive_bitrate_algorithm(ms, MSQosAnalyzerAlgorithmSimple);
+ if (getType() == SalAudio && videoWillBeUsed) {
+ /* If this is an audio stream but video is going to be used, there is no need to perform
+ * basic rate control on the audio stream, just the video stream. */
+ enabled = false;
+ }
+ media_stream_enable_adaptive_bitrate_control(ms, enabled);
+ }
+}
+
+void MS2Stream::tryEarlyMediaForking(const OfferAnswerContext &ctx){
+ RtpSession *session = mSessions.rtp_session;
+ const SalStreamDescription *newStream = ctx.remoteStreamDescription;
+ const char *rtpAddr = (newStream->rtp_addr[0] != '\0') ? newStream->rtp_addr : ctx.remoteMediaDescription->addr;
+ const char *rtcpAddr = (newStream->rtcp_addr[0] != '\0') ? newStream->rtcp_addr : ctx.remoteMediaDescription->addr;
+ if (!ms_is_multicast(rtpAddr)){
+ rtp_session_set_symmetric_rtp(session, false); // Disable symmetric RTP when auxiliary destinations are added.
+ rtp_session_add_aux_remote_addr_full(session, rtpAddr, newStream->rtp_port, rtcpAddr, newStream->rtcp_port);
+ mUseAuxDestinations = true;
+ }
+ Stream::tryEarlyMediaForking(ctx);
+}
+
+void MS2Stream::finishEarlyMediaForking(){
+ if (mUseAuxDestinations){
+ rtp_session_set_symmetric_rtp(mSessions.rtp_session, linphone_core_symmetric_rtp_enabled(getCCore()));
+ rtp_session_clear_aux_remote_addr(mSessions.rtp_session);
+ mUseAuxDestinations = false;
+ }
+}
+
+/*
+ * This function is used by derived implementations that need to extract the destination of RTP/RTCP streams
+ * from the result media description.
+ * Indeed, when RTP bundle mode is ON, this information is to be taken in the transport owner stream.
+ */
+void MS2Stream::getRtpDestination(const OfferAnswerContext ¶ms, RtpAddressInfo *info){
+ const SalStreamDescription *stream = params.resultStreamDescription;
+ if (mRtpBundle && !mOwnsBundle){
+ if (!mBundleOwner){
+ lError() << "Bundle owner shall be set !";
+ }else{
+ stream = ¶ms.resultMediaDescription->streams[mBundleOwner->getIndex()];
+ }
+ }
+
+ info->rtpAddr = stream->rtp_addr[0] != '\0' ? stream->rtp_addr : params.resultMediaDescription->addr;
+ bool isMulticast = !!ms_is_multicast(info->rtpAddr.c_str());
+ info->rtpPort = stream->rtp_port;
+ info->rtcpAddr = stream->rtcp_addr[0] != '\0' ? stream->rtcp_addr : info->rtpAddr;
+ info->rtcpPort = (linphone_core_rtcp_enabled(getCCore()) && !isMulticast) ? (stream->rtcp_port ? stream->rtcp_port : stream->rtp_port + 1) : 0;
+}
+
+/*
+ * Handle some basic session changes.
+ * Return true everything was handled, false otherwise, in which case the caller will have to restart the stream.
+ */
+bool MS2Stream::handleBasicChanges(const OfferAnswerContext ¶ms, CallSession::State targetState){
+ const SalStreamDescription *stream = params.resultStreamDescription;
+
+ if (stream->dir == SalStreamInactive || !sal_stream_description_enabled(stream)){
+ /* In this case all we have to do is to ensure that the stream is stopped. */
+ if (getState() != Stopped) stop();
+ return true;
+ }
+ if (getState() == Stream::Running){
+ int changesToHandle = params.resultStreamDescriptionChanges;
+ if (params.resultStreamDescriptionChanges & SAL_MEDIA_DESCRIPTION_NETWORK_CHANGED){
+ updateDestinations(params);
+ changesToHandle &= ~SAL_MEDIA_DESCRIPTION_NETWORK_CHANGED;
+ }
+ if (params.resultStreamDescriptionChanges & SAL_MEDIA_DESCRIPTION_CRYPTO_KEYS_CHANGED){
+ updateCryptoParameters(params);
+ changesToHandle &= ~SAL_MEDIA_DESCRIPTION_CRYPTO_KEYS_CHANGED;
+ }
+ // SAL_MEDIA_DESCRIPTION_STREAMS_CHANGED monitors the number of streams, it is ignored here.
+ changesToHandle &= ~SAL_MEDIA_DESCRIPTION_STREAMS_CHANGED;
+
+ if (changesToHandle == 0){
+ // We've handled everything.
+ if (params.resultStreamDescriptionChanges){
+ lInfo() << "Stream updated, no need to restart.";
+ }
+ return true;
+ }
+ }else if (getState() == Stream::Stopped){
+ /* Already stopped, nothing to do.*/
+ return false;
+ }
+ /* Otherwise these changes shall be handled by a full restart of the stream. */
+ stop();
+ return false;
+}
+
+void MS2Stream::render(const OfferAnswerContext ¶ms, CallSession::State targetState){
+ const SalStreamDescription *stream = params.resultStreamDescription;
+ const char *rtpAddr = (stream->rtp_addr[0] != '\0') ? stream->rtp_addr : params.resultMediaDescription->addr;
+ bool isMulticast = !!ms_is_multicast(rtpAddr);
+ MediaStream *ms = getMediaStream();
+
+ if (getIceService().isActive() || (getMediaSessionPrivate().getParams()->earlyMediaSendingEnabled()
+ && (targetState == CallSession::State::OutgoingEarlyMedia))) {
+ rtp_session_set_symmetric_rtp(mSessions.rtp_session, false);
+ }
+ media_stream_set_max_network_bitrate(getMediaStream(), linphone_core_get_upload_bandwidth(getCCore()) * 1000);
+ if (isMulticast)
+ rtp_session_set_multicast_ttl(mSessions.rtp_session, stream->ttl);
+ rtp_session_enable_rtcp_mux(mSessions.rtp_session, stream->rtcp_mux);
+ // Valid local tags are > 0
+ if (sal_stream_description_has_srtp(stream)) {
+ int cryptoIdx = Sal::findCryptoIndexFromTag(params.localStreamDescription->crypto, static_cast<unsigned char>(stream->crypto_local_tag));
+ if (cryptoIdx >= 0) {
+ ms_media_stream_sessions_set_srtp_recv_key_b64(&ms->sessions, stream->crypto[0].algo, stream->crypto[0].master_key);
+ ms_media_stream_sessions_set_srtp_send_key_b64(&ms->sessions, stream->crypto[0].algo,
+ params.localStreamDescription->crypto[cryptoIdx].master_key);
+ } else
+ lWarning() << "Failed to find local crypto algo with tag: " << stream->crypto_local_tag;
+ }
+ ms_media_stream_sessions_set_encryption_mandatory(&ms->sessions, getMediaSessionPrivate().isEncryptionMandatory());
+ configureRtpSessionForRtcpFb(params);
+ configureRtpSessionForRtcpXr(params);
+ configureAdaptiveRateControl(params);
+
+ if (stream->dtls_role != SalDtlsRoleInvalid){ /* If DTLS is available at both end points */
+ /* Give the peer certificate fingerprint to dtls context */
+ ms_dtls_srtp_set_peer_fingerprint(ms->sessions.dtls_context, params.remoteStreamDescription->dtls_fingerprint);
+ }
+
+ switch(targetState){
+ case CallSession::State::IncomingEarlyMedia:
+ BCTBX_NO_BREAK;
+ case CallSession::State::OutgoingEarlyMedia:
+ /* don't accept to send real-live media in early media stage by default.*/
+ if (!getMediaSessionPrivate().getParams()->earlyMediaSendingEnabled()) {
+ lInfo() << "Early media sending not allowed, will send silence and dummy video instead.";
+ mMuted = true;
+ }else{
+ lInfo() << "Early media sending allowed, will send real live sound and video.";
+ }
+ break;
+ case CallSession::State::StreamsRunning:
+ mMuted = false;
+ finishEarlyMediaForking();
+ break;
+ default:
+ break;
+ }
+ startEventHandling();
+ initRtpBundle(params);
+ setIceCheckList(mIceCheckList); // do it after enabling bundles
+ Stream::render(params, targetState);
+}
+
+
+OrtpJitterBufferAlgorithm MS2Stream::jitterBufferNameToAlgo (const string &name) {
+ if (name == "basic") return OrtpJitterBufferBasic;
+ if (name == "rls") return OrtpJitterBufferRecursiveLeastSquare;
+ lError() << "Invalid jitter buffer algorithm: " << name;
+ return OrtpJitterBufferRecursiveLeastSquare;
+}
+
+void MS2Stream::applyJitterBufferParams (RtpSession *session) {
+ LinphoneConfig *config = linphone_core_get_config(getCCore());
+ JBParameters params;
+ rtp_session_get_jitter_buffer_params(session, ¶ms);
+ params.min_size = lp_config_get_int(config, "rtp", "jitter_buffer_min_size", 40);
+ params.max_size = lp_config_get_int(config, "rtp", "jitter_buffer_max_size", 500);
+ params.max_packets = params.max_size * 200 / 1000; /* Allow 200 packet per seconds, quite large */
+ const char *algo = lp_config_get_string(config, "rtp", "jitter_buffer_algorithm", "rls");
+ params.buffer_algorithm = jitterBufferNameToAlgo(algo ? algo : "");
+ params.refresh_ms = lp_config_get_int(config, "rtp", "jitter_buffer_refresh_period", 5000);
+ params.ramp_refresh_ms = lp_config_get_int(config, "rtp", "jitter_buffer_ramp_refresh_period", 5000);
+ params.ramp_step_ms = lp_config_get_int(config, "rtp", "jitter_buffer_ramp_step", 20);
+ params.ramp_threshold = lp_config_get_int(config, "rtp", "jitter_buffer_ramp_threshold", 70);
+
+ switch (getType()) {
+ case SalAudio:
+ case SalText: /* Let's use the same params for text as for audio */
+ params.nom_size = linphone_core_get_audio_jittcomp(getCCore());
+ params.adaptive = linphone_core_audio_adaptive_jittcomp_enabled(getCCore());
+ break;
+ case SalVideo:
+ params.nom_size = linphone_core_get_video_jittcomp(getCCore());
+ params.adaptive = linphone_core_video_adaptive_jittcomp_enabled(getCCore());
+ break;
+ default:
+ lError() << "applyJitterBufferParams(): should not happen";
+ break;
+ }
+ params.enabled = params.nom_size > 0;
+ if (params.enabled) {
+ if (params.min_size > params.nom_size)
+ params.min_size = params.nom_size;
+ if (params.max_size < params.nom_size)
+ params.max_size = params.nom_size;
+ }
+ rtp_session_set_jitter_buffer_params(session, ¶ms);
+}
+
+void MS2Stream::configureRtpSession(RtpSession *session){
+ rtp_session_enable_network_simulation(session, &getCCore()->net_conf.netsim_params);
+ applyJitterBufferParams(session);
+ string userAgent = linphone_core_get_user_agent(getCCore());
+ rtp_session_set_source_description(session, getMediaSessionPrivate().getMe()->getAddress().asString().c_str(), NULL, NULL, NULL, NULL, userAgent.c_str(), NULL);
+ rtp_session_set_symmetric_rtp(session, linphone_core_symmetric_rtp_enabled(getCCore()));
+
+ if (getType() == SalVideo){
+ int videoRecvBufSize = lp_config_get_int(linphone_core_get_config(getCCore()), "video", "recv_buf_size", 0);
+ if (videoRecvBufSize > 0)
+ rtp_session_set_recv_buf_size(session, videoRecvBufSize);
+ }
+}
+
+void MS2Stream::setupDtlsParams (MediaStream *ms) {
+ if (getMediaSessionPrivate().getParams()->getMediaEncryption() == LinphoneMediaEncryptionDTLS) {
+ MSDtlsSrtpParams dtlsParams = { 0 };
+
+ /* TODO : search for a certificate with CNAME=sip uri(retrieved from variable me) or default : linphone-dtls-default-identity */
+ /* This will parse the directory to find a matching fingerprint or generate it if not found */
+ /* returned string must be freed */
+ char *certificate = nullptr;
+ char *key = nullptr;
+ char *fingerprint = nullptr;
+
+ sal_certificates_chain_parse_directory(&certificate, &key, &fingerprint,
+ linphone_core_get_user_certificates_path(getCCore()), "linphone-dtls-default-identity", SAL_CERTIFICATE_RAW_FORMAT_PEM, true, true);
+ if (fingerprint) {
+ if (getMediaSessionPrivate().getDtlsFingerprint().empty()){
+ getMediaSessionPrivate().setDtlsFingerprint(fingerprint);
+ }
+ mDtlsFingerPrint = fingerprint;
+ ms_free(fingerprint);
+ }
+ if (key && certificate) {
+ dtlsParams.pem_certificate = certificate;
+ dtlsParams.pem_pkey = key;
+ dtlsParams.role = MSDtlsSrtpRoleUnset; /* Default is unset, then check if we have a result SalMediaDescription */
+ media_stream_enable_dtls(ms, &dtlsParams);
+ ms_free(certificate);
+ ms_free(key);
+ } else {
+ lError() << "Unable to retrieve or generate DTLS certificate and key - DTLS disabled";
+ /* TODO : check if encryption forced, if yes, stop call */
+ }
+ }
+}
+
+void MS2Stream::startDtls(const OfferAnswerContext ¶ms){
+ if (mDtlsStarted) return;
+ if (!sal_stream_description_has_dtls(params.resultStreamDescription)) return;
+
+ if (params.resultStreamDescription->dtls_role == SalDtlsRoleInvalid){
+ lWarning() << "Unable to start DTLS engine on stream session [" << &mSessions << "], Dtls role in resulting media description is invalid";
+ }else {
+ if (!isTransportOwner()){
+ /* RTP bundle mode: there must be only one DTLS association per transport. */
+ return;
+ }
+ /* Workaround for buggy openssl versions that send DTLS packets bigger than the MTU. We need to increase the recv buf size of the RtpSession.*/
+ int recv_buf_size = lp_config_get_int(linphone_core_get_config(getCCore()),"rtp", "dtls_recv_buf_size", 5000);
+ rtp_session_set_recv_buf_size(mSessions.rtp_session, recv_buf_size);
+
+ /* If DTLS is available at both end points */
+ /* Give the peer certificate fingerprint to dtls context */
+ ms_dtls_srtp_set_peer_fingerprint(mSessions.dtls_context, params.remoteStreamDescription->dtls_fingerprint);
+ ms_dtls_srtp_set_role(mSessions.dtls_context, (params.resultStreamDescription->dtls_role == SalDtlsRoleIsClient) ? MSDtlsSrtpRoleIsClient : MSDtlsSrtpRoleIsServer); /* Set the role to client */
+ ms_dtls_srtp_start(mSessions.dtls_context); /* Then start the engine, it will send the DTLS client Hello */
+ mDtlsStarted = true;
+ }
+}
+
+void MS2Stream::initializeSessions(MediaStream *stream){
+ if (mPortConfig.multicastRole == SalMulticastReceiver){
+ if (!mPortConfig.multicastIp.empty())
+ media_stream_join_multicast_group(stream, mPortConfig.multicastIp.c_str());
+ else
+ lError() << "Cannot join multicast group if multicast ip is not set";
+ }
+
+ configureRtpSession(stream->sessions.rtp_session);
+ setupDtlsParams(stream);
+
+ if (mPortConfig.rtpPort == -1){
+ // Case where we requested random ports from the system. Now that they are allocated, get them.
+ mPortConfig.rtpPort = rtp_session_get_local_port(stream->sessions.rtp_session);
+ mPortConfig.rtcpPort = rtp_session_get_local_rtcp_port(stream->sessions.rtp_session);
+ }
+ int dscp = -1;
+ switch(getType()){
+ case SalAudio:
+ dscp = linphone_core_get_audio_dscp(getCCore());
+ break;
+ case SalVideo:
+ dscp = linphone_core_get_video_dscp(getCCore());
+ break;
+ default:
+ break;
+
+ }
+ if (dscp != -1)
+ media_stream_set_dscp(stream, dscp);
+
+ mOrtpEvQueue = ortp_ev_queue_new();
+ rtp_session_register_event_queue(stream->sessions.rtp_session, mOrtpEvQueue);
+
+ media_stream_reclaim_sessions(stream, &mSessions);
+
+}
+
+void MS2Stream::updateCryptoParameters(const OfferAnswerContext ¶ms) {
+ const SalStreamDescription *localStreamDesc = params.localStreamDescription;
+ const SalStreamDescription *newStream = params.resultStreamDescription;
+ MediaStream * ms = getMediaStream();
+
+ if (newStream->proto == SalProtoRtpSavpf || newStream->proto == SalProtoRtpSavp){
+ int cryptoIdx = Sal::findCryptoIndexFromTag(localStreamDesc->crypto, static_cast<unsigned char>(newStream->crypto_local_tag));
+ if (cryptoIdx >= 0) {
+ if (params.localStreamDescriptionChanges & SAL_MEDIA_DESCRIPTION_CRYPTO_KEYS_CHANGED){
+ ms_media_stream_sessions_set_srtp_send_key_b64(&ms->sessions, newStream->crypto[0].algo, localStreamDesc->crypto[cryptoIdx].master_key);
+ }
+ ms_media_stream_sessions_set_srtp_recv_key_b64(&ms->sessions, newStream->crypto[0].algo, newStream->crypto[0].master_key);
+ } else
+ lWarning() << "Failed to find local crypto algo with tag: " << newStream->crypto_local_tag;
+ }
+ startDtls(params);
+}
+
+void MS2Stream::updateDestinations(const OfferAnswerContext ¶ms) {
+ if (params.resultStreamDescription->rtp_port == 0 && params.resultStreamDescription->bundle_only){
+ /* we can ignore */
+ return;
+ }
+
+ const char *rtpAddr = (params.resultStreamDescription->rtp_addr[0] != '\0') ? params.resultStreamDescription->rtp_addr : params.resultMediaDescription->addr;
+ const char *rtcpAddr = (params.resultStreamDescription->rtcp_addr[0] != '\0') ? params.resultStreamDescription->rtcp_addr : params.resultMediaDescription->addr;
+ lInfo() << "Change audio stream destination: RTP=" << rtpAddr << ":" << params.resultStreamDescription->rtp_port << " RTCP=" << rtcpAddr << ":" << params.resultStreamDescription->rtcp_port;
+ rtp_session_set_remote_addr_full(mSessions.rtp_session, rtpAddr, params.resultStreamDescription->rtp_port, rtcpAddr, params.resultStreamDescription->rtcp_port);
+}
+
+void MS2Stream::startEventHandling(){
+ if (mTimer) return;
+ mTimer = getCore().createTimer([this](){
+ handleEvents();
+ return true;
+ }, sEventPollIntervalMs, "Stream event processing timer");
+}
+
+void MS2Stream::stopEventHandling(){
+ if (mTimer){
+ getCore().destroyTimer(mTimer);
+ mTimer = nullptr;
+ }
+}
+
+bool MS2Stream::prepare(){
+ if (getCCore()->rtptf) {
+ RtpTransport *meta_rtp;
+ RtpTransport *meta_rtcp;
+ rtp_session_get_transports(mSessions.rtp_session, &meta_rtp, &meta_rtcp);
+ LinphoneCoreRtpTransportFactoryFunc rtpFunc = nullptr, rtcpFunc = nullptr;
+ void *rtpFuncData = nullptr, *rtcpFuncData = nullptr;
+
+ switch(getType()){
+ case SalAudio:
+ rtpFunc = getCCore()->rtptf->audio_rtp_func;
+ rtpFuncData = getCCore()->rtptf->audio_rtp_func_data;
+ rtcpFunc = getCCore()->rtptf->audio_rtcp_func;
+ rtcpFuncData = getCCore()->rtptf->audio_rtcp_func_data;
+ break;
+ case SalVideo:
+ rtpFunc = getCCore()->rtptf->video_rtp_func;
+ rtpFuncData = getCCore()->rtptf->video_rtp_func_data;
+ rtcpFunc = getCCore()->rtptf->video_rtcp_func;
+ rtcpFuncData = getCCore()->rtptf->video_rtcp_func_data;
+ break;
+ case SalText:
+ break;
+ case SalOther:
+ break;
+ }
+
+ if (!meta_rtp_transport_get_endpoint(meta_rtp)) {
+ lInfo() << this << " using custom RTP transport endpoint";
+ meta_rtp_transport_set_endpoint(meta_rtp, rtpFunc(rtpFuncData, mPortConfig.rtpPort));
+ }
+ if (!meta_rtp_transport_get_endpoint(meta_rtcp))
+ meta_rtp_transport_set_endpoint(meta_rtcp, rtcpFunc(rtcpFuncData, mPortConfig.rtcpPort));
+ }
+ setIceCheckList(mIceCheckList);
+ startEventHandling();
+ Stream::prepare();
+ return false;
+}
+
+void MS2Stream::finishPrepare(){
+ Stream::finishPrepare();
+ stopEventHandling();
+}
+
+int MS2Stream::getIdealAudioBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc) {
+ int remoteBandwidth = 0;
+ if (desc->bandwidth > 0)
+ remoteBandwidth = desc->bandwidth;
+ else if (md->bandwidth > 0) {
+ /* Case where b=AS is given globally, not per stream */
+ remoteBandwidth = md->bandwidth;
+ }
+ int uploadBandwidth = 0;
+ bool forced = false;
+ if (getMediaSessionPrivate().getParams()->getPrivate()->getUpBandwidth() > 0) {
+ forced = true;
+ uploadBandwidth = getMediaSessionPrivate().getParams()->getPrivate()->getUpBandwidth();
+ } else
+ uploadBandwidth = linphone_core_get_upload_bandwidth(getCCore());
+ uploadBandwidth = PayloadTypeHandler::getMinBandwidth(uploadBandwidth, remoteBandwidth);
+ if (!linphone_core_media_description_contains_video_stream(md) || forced)
+ return uploadBandwidth;
+
+ /*
+ * This a default heuristic to choose a target upload bandwidth for an audio stream, the
+ * remaining can then be allocated for video.
+ */
+ if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 512))
+ uploadBandwidth = 100;
+ else if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 256))
+ uploadBandwidth = 64;
+ else if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 128))
+ uploadBandwidth = 40;
+ else if (PayloadTypeHandler::bandwidthIsGreater(uploadBandwidth, 0))
+ uploadBandwidth = 24;
+ return uploadBandwidth;
+}
+
+RtpProfile * MS2Stream::makeProfile(const SalMediaDescription *md, const SalStreamDescription *desc, int *usedPt) {
+ if (mRtpProfile){
+ rtp_profile_destroy(mRtpProfile);
+ mRtpProfile = nullptr;
+ }
+ *usedPt = -1;
+ int bandwidth = 0;
+ if (desc->type == SalAudio)
+ bandwidth = getIdealAudioBandwidth(md, desc);
+ else if (desc->type == SalVideo)
+ bandwidth = getGroup().getVideoBandwidth(md, desc);
+
+ bool first = true;
+ RtpProfile *profile = rtp_profile_new("Call profile");
+ for (const bctbx_list_t *elem = desc->payloads; elem != nullptr; elem = bctbx_list_next(elem)) {
+ OrtpPayloadType *pt = reinterpret_cast<OrtpPayloadType *>(bctbx_list_get_data(elem));
+ /* Make a copy of the payload type, so that we left the ones from the SalStreamDescription unchanged.
+ * If the SalStreamDescription is freed, this will have no impact on the running streams. */
+ pt = payload_type_clone(pt);
+ int upPtime = 0;
+ if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
+ /* First codec in list is the selected one */
+ if (desc->type == SalAudio) {
+ bandwidth = getGroup().updateAllocatedAudioBandwidth(pt, bandwidth);
+ upPtime = getMediaSessionPrivate().getParams()->getPrivate()->getUpPtime();
+ if (!upPtime)
+ upPtime = linphone_core_get_upload_ptime(getCCore());
+ }
+ first = false;
+ }
+ if (*usedPt == -1) {
+ /* Don't select telephone-event as a payload type */
+ if (strcasecmp(pt->mime_type, "telephone-event") != 0)
+ *usedPt = payload_type_get_number(pt);
+ }
+ if (pt->flags & PAYLOAD_TYPE_BITRATE_OVERRIDE) {
+ lInfo() << "Payload type [" << pt->mime_type << "/" << pt->clock_rate << "] has explicit bitrate [" << (pt->normal_bitrate / 1000) << "] kbit/s";
+ pt->normal_bitrate = PayloadTypeHandler::getMinBandwidth(pt->normal_bitrate, bandwidth * 1000);
+ } else
+ pt->normal_bitrate = bandwidth * 1000;
+ if (desc->maxptime > 0) {// follow the same schema for maxptime as for ptime. (I.E add it to fmtp)
+ ostringstream os;
+ os << "maxptime=" << desc->maxptime;
+ payload_type_append_send_fmtp(pt, os.str().c_str());
+ }
+ if (desc->ptime > 0)
+ upPtime = desc->ptime;
+ if (upPtime > 0) {
+ ostringstream os;
+ os << "ptime=" << upPtime;
+ payload_type_append_send_fmtp(pt, os.str().c_str());
+ }
+ int number = payload_type_get_number(pt);
+ if (rtp_profile_get_payload(profile, number))
+ lWarning() << "A payload type with number " << number << " already exists in profile!";
+ else
+ rtp_profile_set_payload(profile, number, pt);
+ }
+ mRtpProfile = profile;
+ return profile;
+}
+
+
+void MS2Stream::updateStats(){
+ if (mSessions.rtp_session) {
+ const rtp_stats_t *rtpStats = rtp_session_get_stats(mSessions.rtp_session);
+ if (rtpStats)
+ _linphone_call_stats_set_rtp_stats(mStats, rtpStats);
+ }
+ float quality = media_stream_get_average_quality_rating(getMediaStream());
+ LinphoneCallLog *log = getMediaSession().getLog();
+ if (quality >= 0) {
+ if (log->quality == -1.0)
+ log->quality = quality;
+ else
+ log->quality *= quality / 5.0f;
+ }
+}
+
+LinphoneCallStats *MS2Stream::getStats(){
+ MediaStream *ms = getMediaStream();
+ if (ms) linphone_call_stats_update(mStats, ms);
+ return mStats;
+}
+
+void MS2Stream::stop(){
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+
+ if (listener){
+ int statsType = -1;
+ switch(getType()){
+ case SalAudio: statsType = LINPHONE_CALL_STATS_AUDIO; break;
+ case SalVideo: statsType = LINPHONE_CALL_STATS_VIDEO; break;
+ case SalText: statsType = LINPHONE_CALL_STATS_TEXT; break;
+ default:
+ break;
+
+ }
+
+ if (statsType != -1) listener->onUpdateMediaInfoForReporting(getMediaSession().getSharedFromThis(), statsType);
+
+ /*
+ * FIXME : very very ugly way to manage the conference. Worse, it can remove from a conference a stream that has never been part
+ * of any conference.
+ * Solution: let the Conference object manage the StreamsGroups that are part of a conference.
+ */
+ if (getType() == SalAudio) listener->onCallSessionConferenceStreamStopping(getMediaSession().getSharedFromThis());
+ }
+ ms_bandwidth_controller_remove_stream(getCCore()->bw_controller, getMediaStream());
+ updateStats();
+ handleEvents();
+ stopEventHandling();
+ media_stream_reclaim_sessions(getMediaStream(), &mSessions);
+ rtp_session_set_profile(mSessions.rtp_session, &av_profile);
+ Stream::stop();
+
+ /* At this time the derived class hasn't yet stopped it streams.
+ * the RTP Profile objects can't be destroyed until the stream is completely stopped.
+ * As a result we do it later*/
+ RtpProfile *rtpProfile = mRtpProfile;
+ RtpProfile *rtpIoProfile = mRtpIoProfile;
+ getCore().doLater( [rtpProfile, rtpIoProfile](){
+ if (rtpProfile) rtp_profile_destroy(rtpProfile);
+ if (rtpIoProfile) rtp_profile_destroy(rtpIoProfile);
+ });
+ mRtpProfile = nullptr;
+ mRtpIoProfile = nullptr;
+}
+
+void MS2Stream::notifyStatsUpdated () {
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+ if (_linphone_call_stats_get_updated(mStats)) {
+ switch (_linphone_call_stats_get_updated(mStats)) {
+ case LINPHONE_CALL_STATS_RECEIVED_RTCP_UPDATE:
+ case LINPHONE_CALL_STATS_SENT_RTCP_UPDATE:
+ if (listener) {
+ listener->onRtcpUpdateForReporting(getMediaSession().getSharedFromThis(), getType());
+ }
+ break;
+ default:
+ break;
+ }
+ if (listener)
+ listener->onStatsUpdated(getMediaSession().getSharedFromThis(), mStats);
+ _linphone_call_stats_set_updated(mStats, 0);
+ }
+}
+
+void MS2Stream::iceStateChanged(){
+ updateIceInStats();
+}
+
+void MS2Stream::updateIceInStats(LinphoneIceState state){
+ lInfo() << "ICE state is " << linphone_ice_state_to_string(state) << " for " << *this;
+ _linphone_call_stats_set_ice_state(mStats, state);
+}
+
+void MS2Stream::updateIceInStats(){
+ /* Special case for rtp bundle: we report the ice state of the transport owner. */
+ if (mRtpBundle && !mOwnsBundle && mBundleOwner && mBundleOwner->mStats){
+ updateIceInStats(linphone_call_stats_get_ice_state(mBundleOwner->mStats));
+ return;
+ }
+
+ if (!mIceCheckList){
+ updateIceInStats(LinphoneIceStateNotActivated);
+ return;
+ }
+ if (ice_check_list_state(mIceCheckList) == ICL_Failed) {
+ updateIceInStats(LinphoneIceStateFailed);
+ return;
+ }
+ if (ice_check_list_state(mIceCheckList) == ICL_Running) {
+ updateIceInStats(LinphoneIceStateInProgress);
+ return;
+ }
+ /* Otherwise we are in ICL_Completed state. */
+
+ switch (ice_check_list_selected_valid_candidate_type(mIceCheckList)) {
+ case ICT_HostCandidate:
+ updateIceInStats(LinphoneIceStateHostConnection);
+ break;
+ case ICT_ServerReflexiveCandidate:
+ case ICT_PeerReflexiveCandidate:
+ updateIceInStats(LinphoneIceStateReflexiveConnection);
+ break;
+ case ICT_RelayedCandidate:
+ updateIceInStats(LinphoneIceStateRelayConnection);
+ break;
+ case ICT_CandidateInvalid:
+ case ICT_CandidateTypeMax:
+ // Shall not happen.
+ L_ASSERT(false);
+ break;
+ }
+}
+
+void MS2Stream::dtlsEncryptionChanged(){
+ getGroup().propagateEncryptionChanged();
+}
+
+void MS2Stream::handleEvents () {
+ MediaStream *ms = getMediaStream();
+ if (ms) {
+ switch(ms->type){
+ case MSAudio:
+ audio_stream_iterate((AudioStream *)ms);
+ break;
+ case MSVideo:
+#ifdef VIDEO_ENABLED
+ video_stream_iterate((VideoStream *)ms);
+#endif
+ break;
+ case MSText:
+ text_stream_iterate((TextStream *)ms);
+ break;
+ default:
+ lError() << "handleStreamEvents(): unsupported stream type";
+ return;
+ }
+ }
+ OrtpEvent *ev;
+
+ while ((ev = ortp_ev_queue_get(mOrtpEvQueue)) != nullptr) {
+ OrtpEventType evt = ortp_event_get_type(ev);
+ OrtpEventData *evd = ortp_event_get_data(ev);
+
+ /*This MUST be done before any call to "linphone_call_stats_fill" since it has ownership over evd->packet*/
+ if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
+ do {
+ if (evd->packet && rtcp_is_RTPFB(evd->packet)) {
+ if (rtcp_RTPFB_get_type(evd->packet) == RTCP_RTPFB_TMMBR) {
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+ listener->onTmmbrReceived(getMediaSession().getSharedFromThis(), (int)getIndex(), (int)rtcp_RTPFB_tmmbr_get_max_bitrate(evd->packet));
+ }
+ }
+ } while (rtcp_next_packet(evd->packet));
+ rtcp_rewind(evd->packet);
+ }
+
+ if (ms)
+ linphone_call_stats_fill(mStats, ms, ev);
+ switch(evt){
+ case ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED:
+ if (getType() != SalAudio || !isMain()){
+ getGroup().propagateEncryptionChanged();
+ }
+ break;
+ case ORTP_EVENT_DTLS_ENCRYPTION_CHANGED:
+ dtlsEncryptionChanged();
+ break;
+ case ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED:
+ case ORTP_EVENT_ICE_GATHERING_FINISHED:
+ case ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED:
+ case ORTP_EVENT_ICE_RESTART_NEEDED:
+ /* ICE events are notified directly to the IceService. */
+ getIceService().handleIceEvent(ev);
+ break;
+ }
+ notifyStatsUpdated();
+
+ /* Let subclass handle the event.*/
+ handleEvent(ev);
+ ortp_event_destroy(ev);
+ }
+}
+
+bool MS2Stream::isEncrypted() const{
+ if (!isTransportOwner()){
+ if (mBundleOwner){
+ return mBundleOwner->isEncrypted(); /* We must refer to the stream that owns the Rtp bundle.*/
+ }else{
+ lError() << "MS2Stream::isEncrypted(): no bundle owner !";
+ }
+ return false;
+ }
+ return media_stream_secured(getMediaStream());
+}
+
+bool MS2Stream::isMuted()const{
+ return mMuted;
+}
+
+RtpSession* MS2Stream::createRtpIoSession() {
+ LinphoneConfig *config = linphone_core_get_config(getCCore());
+ const char *config_section = getType() == SalAudio ? "sound" : "video";
+ const char *rtpmap = lp_config_get_string(config, config_section, "rtp_map", getType() == SalAudio ? "pcmu/8000/1" : "vp8/90000");
+ OrtpPayloadType *pt = rtp_profile_get_payload_from_rtpmap(mRtpProfile, rtpmap);
+ if (!pt)
+ return nullptr;
+ string profileName = string("RTP IO ") + string(config_section) + string(" profile");
+ mRtpIoProfile = rtp_profile_new(profileName.c_str());
+ int ptnum = lp_config_get_int(config, config_section, "rtp_ptnum", 0);
+ rtp_profile_set_payload(mRtpIoProfile, ptnum, payload_type_clone(pt));
+ const char *localIp = lp_config_get_string(config, config_section, "rtp_local_addr", "127.0.0.1");
+ int localPort = lp_config_get_int(config, config_section, "rtp_local_port", 17076);
+ RtpSession *rtpSession = ms_create_duplex_rtp_session(localIp, localPort, -1, ms_factory_get_mtu(getCCore()->factory));
+ rtp_session_set_profile(rtpSession, mRtpIoProfile);
+ const char *remoteIp = lp_config_get_string(config, config_section, "rtp_remote_addr", "127.0.0.1");
+ int remotePort = lp_config_get_int(config, config_section, "rtp_remote_port", 17078);
+ rtp_session_set_remote_addr_and_port(rtpSession, remoteIp, remotePort, -1);
+ rtp_session_enable_rtcp(rtpSession, false);
+ rtp_session_set_payload_type(rtpSession, ptnum);
+ int jittcomp = lp_config_get_int(config, config_section, "rtp_jittcomp", 0); /* 0 means no jitter buffer */
+ rtp_session_set_jitter_compensation(rtpSession, jittcomp);
+ rtp_session_enable_jitter_buffer(rtpSession, (jittcomp > 0));
+ bool symmetric = !!lp_config_get_int(config, config_section, "rtp_symmetric", 0);
+ rtp_session_set_symmetric_rtp(rtpSession, symmetric);
+ return rtpSession;
+}
+
+std::pair<RtpTransport*, RtpTransport*> MS2Stream::getMetaRtpTransports(){
+ RtpTransport *metaRtp = nullptr;
+ RtpTransport *metaRtcp = nullptr;
+ rtp_session_get_transports(mSessions.rtp_session, &metaRtp, &metaRtcp);
+ return make_pair(metaRtp, metaRtcp);
+}
+
+MSZrtpContext *MS2Stream::getZrtpContext()const{
+ return mSessions.zrtp_context;
+}
+
+float MS2Stream::getAverageQuality(){
+ MediaStream *ms = getMediaStream();
+ if (!ms) {
+ lError() << "MS2Stream::getAverageQuality(): no stream.";
+ return 0.0;
+ }
+ return media_stream_get_average_quality_rating(ms);
+}
+
+float MS2Stream::getCurrentQuality(){
+ MediaStream *ms = getMediaStream();
+ if (!ms) {
+ lError() << "MS2Stream::getCurrentQuality(): no stream.";
+ return 0.0;
+ }
+ return media_stream_get_quality_rating(getMediaStream());
+}
+
+void MS2Stream::updateBandwidthReports(){
+ MediaStream * ms = getMediaStream();
+ bool active = ms ? (media_stream_get_state(ms) == MSStreamStarted) : false;
+ _linphone_call_stats_set_download_bandwidth(mStats, active ? (float)(media_stream_get_down_bw(ms) * 1e-3) : 0.f);
+ _linphone_call_stats_set_upload_bandwidth(mStats, active ? (float)(media_stream_get_up_bw(ms) * 1e-3) : 0.f);
+ _linphone_call_stats_set_rtcp_download_bandwidth(mStats, active ? (float)(media_stream_get_rtcp_down_bw(ms) * 1e-3) : 0.f);
+ _linphone_call_stats_set_rtcp_upload_bandwidth(mStats, active ? (float)(media_stream_get_rtcp_up_bw(ms) * 1e-3) : 0.f);
+ _linphone_call_stats_set_ip_family_of_remote(mStats,
+ active ? (ortp_stream_is_ipv6(&mSessions.rtp_session->rtp.gs) ? LinphoneAddressFamilyInet6 : LinphoneAddressFamilyInet) : LinphoneAddressFamilyUnspec);
+
+ if (getCCore()->send_call_stats_periodical_updates) {
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+ if (active)
+ linphone_call_stats_update(mStats, ms);
+ _linphone_call_stats_set_updated(mStats, _linphone_call_stats_get_updated(mStats) | LINPHONE_CALL_STATS_PERIODICAL_UPDATE);
+ if (listener)
+ listener->onStatsUpdated(getMediaSession().getSharedFromThis(), mStats);
+ _linphone_call_stats_set_updated(mStats, 0);
+ }
+}
+
+float MS2Stream::getCpuUsage()const{
+ MediaStream *ms = getMediaStream();
+ if (ms->sessions.ticker == nullptr) return 0.0f;
+ return ms_ticker_get_average_load(ms->sessions.ticker);
+}
+
+void MS2Stream::finish(){
+ if (mRtpBundle && mOwnsBundle){
+ rtp_bundle_delete(mRtpBundle);
+ mRtpBundle = nullptr;
+ }
+ if (mOrtpEvQueue){
+ rtp_session_unregister_event_queue(mSessions.rtp_session, mOrtpEvQueue);
+ ortp_ev_queue_flush(mOrtpEvQueue);
+ ortp_ev_queue_destroy(mOrtpEvQueue);
+ mOrtpEvQueue = nullptr;
+ }
+ ms_media_stream_sessions_uninit(&mSessions);
+ Stream::finish();
+}
+
+bool MS2Stream::avpfEnabled() const{
+ return media_stream_avpf_enabled(getMediaStream());
+}
+
+bool MS2Stream::bundleEnabled() const{
+ return mRtpBundle != nullptr;
+}
+
+bool MS2Stream::isTransportOwner() const{
+ bool ret = mRtpBundle == nullptr || mOwnsBundle;
+ return ret;
+}
+
+int MS2Stream::getAvpfRrInterval()const{
+ MediaStream *ms = getMediaStream();
+ return media_stream_get_state(ms) == MSStreamStarted ? media_stream_get_avpf_rr_interval(ms) : 0;
+}
+
+MS2Stream::~MS2Stream(){
+ finish();
+ linphone_call_stats_unref(mStats);
+ mStats = nullptr;
+
+}
+
+
+
+LINPHONE_END_NAMESPACE
+
diff --git a/src/conference/session/ms2-streams.h b/src/conference/session/ms2-streams.h
new file mode 100644
index 0000000000000000000000000000000000000000..8f9e49ea188966e0e476433c72588219107c642e
--- /dev/null
+++ b/src/conference/session/ms2-streams.h
@@ -0,0 +1,258 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef ms2_streams_h
+#define ms2_streams_h
+
+#include "streams.h"
+
+LINPHONE_BEGIN_NAMESPACE
+
+/**
+ * Derived class for streams commonly handly through mediastreamer2 library.
+ */
+class MS2Stream : public Stream, public RtpInterface {
+public:
+ virtual void fillLocalMediaDescription(OfferAnswerContext & ctx) override;
+ virtual bool prepare() override;
+ virtual void finishPrepare() override;
+ virtual void render(const OfferAnswerContext & ctx, CallSession::State targetState) override;
+ virtual void stop() override;
+ virtual void finish() override;
+ virtual bool isEncrypted() const override;
+ MSZrtpContext *getZrtpContext()const;
+ std::pair<RtpTransport*, RtpTransport*> getMetaRtpTransports();
+ virtual MediaStream *getMediaStream()const = 0;
+ virtual void tryEarlyMediaForking(const OfferAnswerContext &ctx) override;
+ virtual void finishEarlyMediaForking() override;
+ virtual float getCurrentQuality() override;
+ virtual float getAverageQuality() override;
+ virtual LinphoneCallStats *getStats() override;
+ virtual void startDtls(const OfferAnswerContext ¶ms) override;
+ virtual bool isMuted()const override;
+ virtual void refreshSockets() override;
+ virtual void updateBandwidthReports() override;
+ virtual float getCpuUsage()const override;
+ virtual void setIceCheckList(IceCheckList *cl) override;
+ virtual void iceStateChanged() override;
+
+ /* RtpInterface */
+ virtual bool avpfEnabled() const override;
+ virtual bool bundleEnabled() const override;
+ virtual int getAvpfRrInterval() const override;
+ virtual bool isTransportOwner() const override;
+
+ virtual ~MS2Stream();
+protected:
+ virtual void handleEvent(const OrtpEvent *ev) = 0;
+ MS2Stream(StreamsGroup &sm, const OfferAnswerContext ¶ms);
+ void startEventHandling();
+ void stopEventHandling();
+ std::string getBindIp();
+ int getBindPort();
+ void initializeSessions(MediaStream *stream);
+ RtpProfile * makeProfile(const SalMediaDescription *md, const SalStreamDescription *desc, int *usedPt);
+ int getIdealAudioBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc);
+ RtpSession* createRtpIoSession();
+ void updateCryptoParameters(const OfferAnswerContext ¶ms);
+ void updateDestinations(const OfferAnswerContext ¶ms);
+ bool handleBasicChanges(const OfferAnswerContext ¶ms, CallSession::State targetState);
+ struct RtpAddressInfo{
+ std::string rtpAddr;
+ std::string rtcpAddr;
+ int rtpPort, rtcpPort;
+ };
+ void getRtpDestination(const OfferAnswerContext ¶ms, RtpAddressInfo *info);
+ void dtlsEncryptionChanged();
+ std::string mDtlsFingerPrint;
+ RtpProfile *mRtpProfile = nullptr;
+ RtpProfile *mRtpIoProfile = nullptr;
+ MSMediaStreamSessions mSessions;
+ OrtpEvQueue *mOrtpEvQueue = nullptr;
+ LinphoneCallStats *mStats = nullptr;
+ bool mUseAuxDestinations = false;
+ bool mMuted = false; /* to handle special cases where we want the audio to be muted - not related with linphone_core_enable_mic().*/
+ bool mDtlsStarted = false;
+private:
+ void initRtpBundle(const OfferAnswerContext ¶ms);
+ RtpBundle *createOrGetRtpBundle(const SalStreamDescription *sd);
+ void removeFromBundle();
+ void notifyStatsUpdated();
+ void handleEvents();
+ void updateStats();
+ void initMulticast(const OfferAnswerContext ¶ms);
+ void configureRtpSession(RtpSession *session);
+ void applyJitterBufferParams (RtpSession *session);
+ void setupDtlsParams(MediaStream *ms);
+ void configureRtpSessionForRtcpFb (const OfferAnswerContext ¶ms);
+ void configureRtpSessionForRtcpXr(const OfferAnswerContext ¶ms);
+ void configureAdaptiveRateControl(const OfferAnswerContext ¶ms);
+ void updateIceInStats(LinphoneIceState state);
+ void updateIceInStats();
+ belle_sip_source_t *mTimer = nullptr;
+ IceCheckList *mIceCheckList = nullptr;
+ RtpBundle *mRtpBundle = nullptr;
+ MS2Stream *mBundleOwner = nullptr;
+ bool mOwnsBundle = false;
+ static OrtpJitterBufferAlgorithm jitterBufferNameToAlgo(const std::string &name);
+ static constexpr const int sEventPollIntervalMs = 20;
+};
+
+class MS2AudioStream : public MS2Stream, public AudioControlInterface{
+ friend class MS2VideoStream;
+public:
+ MS2AudioStream(StreamsGroup &sg, const OfferAnswerContext ¶ms);
+ virtual bool prepare() override;
+ virtual void finishPrepare() override;
+ virtual void render(const OfferAnswerContext &ctx, CallSession::State targetState) override;
+ virtual void sessionConfirmed(const OfferAnswerContext &ctx) override;
+ virtual void stop() override;
+ virtual void finish() override;
+
+ /* AudioControlInterface */
+ virtual void enableMic(bool value) override;
+ virtual void enableSpeaker(bool value) override;
+ virtual bool micEnabled()const override;
+ virtual bool speakerEnabled()const override;
+ virtual void startRecording() override;
+ virtual void stopRecording() override;
+ virtual bool isRecording() override{
+ return mRecordActive;
+ }
+ virtual float getPlayVolume() override;
+ virtual float getRecordVolume() override;
+ virtual float getMicGain() override;
+ virtual void setMicGain(float value) override;
+ virtual float getSpeakerGain() override;
+ virtual void setSpeakerGain(float value) override;
+ virtual void setRoute(LinphoneAudioRoute route) override;
+ virtual void sendDtmf(int dtmf) override;
+ virtual void enableEchoCancellation(bool value) override;
+ virtual bool echoCancellationEnabled()const override;
+
+ virtual MediaStream *getMediaStream()const override;
+ virtual ~MS2AudioStream();
+
+ /* Yeah quite ugly: this function is used externally to configure raw mediastreamer2 AudioStreams.*/
+ static void postConfigureAudioStream(AudioStream *as, LinphoneCore *lc, bool muted);
+ MSSndCard *getCurrentPlaybackCard()const{ return mCurrentPlaybackCard; }
+ MSSndCard *getCurrentCaptureCard()const{ return mCurrentCaptureCard; }
+
+protected:
+ VideoStream *getPeerVideoStream();
+private:
+ virtual void handleEvent(const OrtpEvent *ev) override;
+ void setupMediaLossCheck();
+ void setPlaybackGainDb (float gain);
+ void setZrtpCryptoTypesParameters(MSZrtpParams *params, bool haveZrtpHash);
+ void startZrtpPrimaryChannel(const OfferAnswerContext ¶ms);
+ static void parameterizeEqualizer(AudioStream *as, LinphoneCore *lc);
+ void forceSpeakerMuted(bool muted);
+ void postConfigureAudioStream(bool muted);
+ void setupRingbackPlayer();
+ void telephoneEventReceived (int event);
+ void configureAudioStream();
+ AudioStream *mStream = nullptr;
+ MSSndCard *mCurrentCaptureCard = nullptr;
+ MSSndCard *mCurrentPlaybackCard = nullptr;
+ belle_sip_source_t *mMediaLostCheckTimer = nullptr;
+ bool mMicMuted = false;
+ bool mSpeakerMuted = false;
+ bool mRecordActive = false;
+ bool mStartZrtpLater = false;
+ static constexpr const int ecStateMaxLen = 1048576; /* 1Mo */
+ static constexpr const char * ecStateStore = ".linphone.ecstate";
+};
+
+class MS2VideoStream : public MS2Stream, public VideoControlInterface{
+public:
+ MS2VideoStream(StreamsGroup &sg, const OfferAnswerContext ¶m);
+ virtual bool prepare() override;
+ virtual void finishPrepare() override;
+ virtual void render(const OfferAnswerContext &ctx, CallSession::State targetState) override;
+ virtual void stop() override;
+ virtual void finish() override;
+
+ /* VideoControlInterface methods */
+ virtual void sendVfu() override;
+ virtual void sendVfuRequest() override;
+ virtual void enableCamera(bool value) override;
+ virtual bool cameraEnabled() const override;
+ virtual void setNativeWindowId(void *w) override;
+ virtual void * getNativeWindowId() const override;
+ virtual void setNativePreviewWindowId(void *w) override;
+ virtual void * getNativePreviewWindowId() const override;
+ virtual void tryEarlyMediaForking(const OfferAnswerContext &ctx) override;
+ virtual void parametersChanged() override;
+ virtual void requestNotifyNextVideoFrameDecoded () override;
+ virtual int takePreviewSnapshot (const std::string& file) override;
+ virtual int takeVideoSnapshot (const std::string& file) override;
+ virtual void zoomVideo (float zoomFactor, float cx, float cy) override;
+ virtual void getRecvStats(VideoStats *s) const override;
+ virtual void getSendStats(VideoStats *s) const override;
+
+ virtual MediaStream *getMediaStream()const override;
+
+ void oglRender();
+ MSWebCam * getVideoDevice(CallSession::State targetState)const;
+
+ virtual ~MS2VideoStream();
+protected:
+ AudioStream *getPeerAudioStream();
+
+private:
+ virtual void handleEvent(const OrtpEvent *ev) override;
+ virtual void zrtpStarted(Stream *mainZrtpStream) override;
+ static void sSnapshotTakenCb(void *userdata, struct _MSFilter *f, unsigned int id, void *arg);
+ void snapshotTakenCb(void *userdata, struct _MSFilter *f, unsigned int id, void *arg);
+ void videoStreamEventCb(const MSFilter *f, const unsigned int eventId, const void *args);
+ static void sVideoStreamEventCb (void *userData, const MSFilter *f, const unsigned int eventId, const void *args);
+ void activateZrtp();
+ VideoStream *mStream = nullptr;
+ void *mNativeWindowId = nullptr;
+ void *mNativePreviewWindowId = nullptr;
+ bool mCameraEnabled = true;
+
+};
+
+/*
+ * Real time text stream.
+ */
+class MS2RTTStream : public MS2Stream{
+public:
+ MS2RTTStream(StreamsGroup &sm, const OfferAnswerContext ¶m);
+ virtual bool prepare() override;
+ virtual void finishPrepare() override;
+ virtual void render(const OfferAnswerContext &ctx, CallSession::State targetState) override;
+ virtual void stop() override;
+ virtual void finish() override;
+ virtual ~MS2RTTStream();
+private:
+ void realTimeTextCharacterReceived(MSFilter *f, unsigned int id, void *arg);
+ static void sRealTimeTextCharacterReceived(void *userData, MSFilter *f, unsigned int id, void *arg);
+ virtual MediaStream *getMediaStream()const override;
+ virtual void handleEvent(const OrtpEvent *ev) override;
+ TextStream *mStream = nullptr;
+};
+
+
+LINPHONE_END_NAMESPACE
+
+#endif
+
diff --git a/src/conference/session/port-config.h b/src/conference/session/port-config.h
index 06a2b2b65ed264e941623b438c0b87505113e02a..00f00ccb5c12fd31ae753c4c25e11f32a3c90c54 100644
--- a/src/conference/session/port-config.h
+++ b/src/conference/session/port-config.h
@@ -23,12 +23,14 @@
#include <string>
#include "linphone/utils/general.h"
+#include "c-wrapper/internal/c-sal.h"
// =============================================================================
LINPHONE_BEGIN_NAMESPACE
struct PortConfig {
+ SalMulticastRole multicastRole = SalMulticastInactive;
std::string multicastIp;
std::string multicastBindIp;
int rtpPort = -1;
diff --git a/src/conference/session/rtt-stream.cpp b/src/conference/session/rtt-stream.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..55591ea25f7e2dc061c3fba6e75ac5d5a2946275
--- /dev/null
+++ b/src/conference/session/rtt-stream.cpp
@@ -0,0 +1,133 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+#include "bctoolbox/defs.h"
+
+#include "ms2-streams.h"
+#include "media-session.h"
+#include "media-session-p.h"
+#include "core/core.h"
+#include "c-wrapper/c-wrapper.h"
+#include "call/call.h"
+#include "call/call-p.h"
+#include "conference/participant.h"
+#include "conference/params/media-session-params-p.h"
+
+#include "linphone/core.h"
+
+using namespace ::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+/*
+ * MS2RTTStream implementation.
+ */
+
+MS2RTTStream::MS2RTTStream(StreamsGroup &sg, const OfferAnswerContext ¶ms) : MS2Stream(sg, params){
+ string bindIp = getBindIp();
+ mStream = text_stream_new2(getCCore()->factory, bindIp.empty() ? nullptr : bindIp.c_str(), mPortConfig.rtpPort, mPortConfig.rtcpPort);
+ initializeSessions(&mStream->ms);
+}
+
+void MS2RTTStream::realTimeTextCharacterReceived (MSFilter *f, unsigned int id, void *arg) {
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+
+ if (id == MS_RTT_4103_RECEIVED_CHAR) {
+ RealtimeTextReceivedCharacter *data = static_cast<RealtimeTextReceivedCharacter *>(arg);
+ if (listener)
+ listener->onRealTimeTextCharacterReceived(getMediaSession().getSharedFromThis(), data);
+ }
+}
+
+void MS2RTTStream::sRealTimeTextCharacterReceived (void *userData, MSFilter *f, unsigned int id, void *arg) {
+ MS2RTTStream *zis = static_cast<MS2RTTStream *>(userData);
+ zis->realTimeTextCharacterReceived(f, id, arg);
+}
+
+bool MS2RTTStream::prepare(){
+ MS2Stream::prepare();
+ text_stream_prepare_text(mStream);
+ return false;
+}
+
+void MS2RTTStream::finishPrepare(){
+ MS2Stream::finishPrepare();
+ text_stream_unprepare_text(mStream);
+}
+
+void MS2RTTStream::render(const OfferAnswerContext ¶ms, CallSession::State targetState){
+ const SalStreamDescription *tstream = params.resultStreamDescription;
+ bool basicChangesHandled = handleBasicChanges(params, targetState);
+
+ if (basicChangesHandled) {
+ if (getState() == Running) MS2Stream::render(params, targetState);
+ return;
+ }
+
+ MS2Stream::render(params, targetState);
+ RtpAddressInfo dest;
+ getRtpDestination(params, &dest);
+ int usedPt = -1;
+ RtpProfile * textProfile = makeProfile(params.resultMediaDescription, tstream, &usedPt);
+ if (usedPt == -1){
+ lError() << "No payload type was accepted for text stream.";
+ stop();
+ return;
+ }
+ getMediaSessionPrivate().getCurrentParams()->getPrivate()->setUsedRealtimeTextCodec(rtp_profile_get_payload(textProfile, usedPt));
+ getMediaSessionPrivate().getCurrentParams()->enableRealtimeText(true);
+
+ unsigned int interval = getMediaSessionPrivate().getParams()->realtimeTextKeepaliveInterval();
+ getMediaSessionPrivate().getCurrentParams()->setRealtimeTextKeepaliveInterval(interval);
+
+ text_stream_start(mStream, textProfile, dest.rtpAddr.c_str(), dest.rtpPort, dest.rtcpAddr.c_str(), dest.rtcpPort, usedPt);
+ ms_filter_add_notify_callback(mStream->rttsink, sRealTimeTextCharacterReceived, this, false);
+ ms_filter_call_method(mStream->rttsource, MS_RTT_4103_SOURCE_SET_KEEP_ALIVE_INTERVAL, &interval);
+ mStartCount++;
+}
+
+void MS2RTTStream::stop(){
+ MS2Stream::stop();
+ text_stream_stop(mStream);
+ /* In mediastreamer2, stop actually stops and destroys. We immediately need to recreate the stream object for later use, keeping the
+ * sessions (for RTP, SRTP, ZRTP etc) that were setup at the beginning. */
+ mStream = text_stream_new_with_sessions(getCCore()->factory, &mSessions);
+}
+
+void MS2RTTStream::finish(){
+ if (mStream){
+ text_stream_stop(mStream);
+ mStream = nullptr;
+ }
+}
+
+MS2RTTStream::~MS2RTTStream(){
+ finish();
+}
+
+MediaStream *MS2RTTStream::getMediaStream()const{
+ return &mStream->ms;
+}
+
+void MS2RTTStream::handleEvent(const OrtpEvent *ev){
+}
+
+
+
+
+LINPHONE_END_NAMESPACE
diff --git a/src/conference/session/stream.cpp b/src/conference/session/stream.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..bd973f13b56451270eaefefd93aa823110bab189
--- /dev/null
+++ b/src/conference/session/stream.cpp
@@ -0,0 +1,216 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "bctoolbox/defs.h"
+
+#include "streams.h"
+#include "media-session.h"
+#include "media-session-p.h"
+#include "core/core.h"
+#include "c-wrapper/c-wrapper.h"
+#include "call/call.h"
+#include "call/call-p.h"
+#include "conference/participant.h"
+#include "utils/payload-type-handler.h"
+#include "conference/params/media-session-params-p.h"
+
+#include "linphone/core.h"
+
+
+using namespace::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+
+/*
+ * Stream implementation.
+ */
+
+
+Stream::Stream(StreamsGroup &sg, const OfferAnswerContext ¶ms) : mStreamsGroup(sg), mStreamType(params.localStreamDescription->type), mIndex(params.streamIndex){
+ setPortConfig();
+ fillMulticastMediaAddresses();
+}
+
+void Stream::setMain(){
+ mIsMain = true;
+}
+
+LinphoneCore *Stream::getCCore()const{
+ return getCore().getCCore();
+}
+
+Core &Stream::getCore()const{
+ return *mStreamsGroup.getMediaSession().getCore();
+}
+
+MediaSession &Stream::getMediaSession()const{
+ return mStreamsGroup.getMediaSession();
+}
+
+MediaSessionPrivate &Stream::getMediaSessionPrivate()const{
+ return *getMediaSession().getPrivate();
+}
+
+void Stream::fillLocalMediaDescription(OfferAnswerContext & ctx){
+}
+
+bool Stream::prepare(){
+ mState = Preparing;
+ return false;
+}
+
+void Stream::finishPrepare(){
+ mState = Stopped;
+}
+
+void Stream::tryEarlyMediaForking(const OfferAnswerContext &ctx){
+}
+
+void Stream::render(const OfferAnswerContext & ctx, CallSession::State targetState){
+ mState = Running;
+}
+
+void Stream::sessionConfirmed(const OfferAnswerContext &ctx){
+}
+
+void Stream::stop(){
+ mState = Stopped;
+}
+
+void Stream::setIceCheckList(IceCheckList *cl){
+}
+
+void Stream::iceStateChanged(){
+}
+
+void Stream::setRandomPortConfig () {
+ mPortConfig.rtpPort = -1;
+ mPortConfig.rtcpPort = -1;
+}
+
+int Stream::selectRandomPort (pair<int, int> portRange) {
+ unsigned int rangeSize = static_cast<unsigned int>(portRange.second - portRange.first);
+
+ for (int nbTries = 0; nbTries < 100; nbTries++) {
+ bool alreadyUsed = false;
+ unsigned int randomInRangeSize = (bctbx_random() % rangeSize) & (unsigned int)~0x1; /* Select an even number */
+ int triedPort = ((int)randomInRangeSize) + portRange.first;
+ /*If portRange.first is even, the triedPort will be even too. The one who configures a port range that starts with an odd number will
+ * get odd RTP port numbers.*/
+
+ for (const bctbx_list_t *elem = linphone_core_get_calls(getCCore()); elem != nullptr; elem = bctbx_list_next(elem)) {
+ LinphoneCall *lcall = reinterpret_cast<LinphoneCall *>(bctbx_list_get_data(elem));
+ shared_ptr<MediaSession> session = static_pointer_cast<MediaSession>(L_GET_CPP_PTR_FROM_C_OBJECT(lcall)->getPrivate()->getActiveSession());
+ if (session->getPrivate()->getStreamsGroup().isPortUsed(triedPort)) {
+ alreadyUsed = true;
+ break;
+ }
+ }
+ if (!alreadyUsed){
+ lInfo() << "Port " << triedPort << " randomly taken from range [ " << portRange.first << " , " << portRange.second << "]";
+ return triedPort;
+ }
+ }
+
+ lError() << "Could not find any free port!";
+ return -1;
+}
+
+int Stream::selectFixedPort (pair<int, int> portRange) {
+ for (int triedPort = portRange.first; triedPort < (portRange.first + 100); triedPort += 2) {
+ bool alreadyUsed = false;
+ for (const bctbx_list_t *elem = linphone_core_get_calls(getCCore()); elem != nullptr; elem = bctbx_list_next(elem)) {
+ LinphoneCall *lcall = reinterpret_cast<LinphoneCall *>(bctbx_list_get_data(elem));
+ shared_ptr<MediaSession> session = static_pointer_cast<MediaSession>(L_GET_CPP_PTR_FROM_C_OBJECT(lcall)->getPrivate()->getActiveSession());
+ if (session->getPrivate()->getStreamsGroup().isPortUsed(triedPort)) {
+ alreadyUsed = true;
+ break;
+ }
+ }
+ if (!alreadyUsed)
+ return triedPort;
+ }
+
+ lError() << "Could not find any free port !";
+ return -1;
+}
+
+void Stream::setPortConfig(pair<int, int> portRange) {
+ if ((portRange.first <= 0) && (portRange.second <= 0)) {
+ setRandomPortConfig();
+ } else {
+ if (portRange.first == portRange.second) {
+ /* Fixed port */
+ mPortConfig.rtpPort = selectFixedPort(portRange);
+ } else {
+ /* Select random port in the specified range */
+ mPortConfig.rtpPort = selectRandomPort(portRange);
+ }
+ }
+ if (mPortConfig.rtpPort == -1) setRandomPortConfig();
+ else mPortConfig.rtcpPort = mPortConfig.rtpPort + 1;
+}
+
+void Stream::setPortConfig(){
+ int minPort = 0, maxPort = 0;
+ switch(getType()){
+ case SalAudio:
+ linphone_core_get_audio_port_range(getCCore(), &minPort, &maxPort);
+ break;
+ case SalVideo:
+ linphone_core_get_video_port_range(getCCore(), &minPort, &maxPort);
+ break;
+ case SalText:
+ linphone_core_get_text_port_range(getCCore(), &minPort, &maxPort);
+ break;
+ case SalOther:
+ break;
+ }
+ setPortConfig(make_pair(minPort, maxPort));
+}
+
+void Stream::fillMulticastMediaAddresses () {
+ mPortConfig.multicastIp.clear();
+ if (getType() == SalAudio && getMediaSession().getPrivate()->getParams()->audioMulticastEnabled()){
+ mPortConfig.multicastIp = linphone_core_get_audio_multicast_addr(getCCore());
+ } else if (getType() == SalVideo && getMediaSession().getPrivate()->getParams()->videoMulticastEnabled()){
+ mPortConfig.multicastIp = linphone_core_get_video_multicast_addr(getCCore());
+ }
+}
+
+bool Stream::isPortUsed(int port)const{
+ return port == mPortConfig.rtpPort || port == mPortConfig.rtcpPort;
+}
+
+IceService & Stream::getIceService()const{
+ return mStreamsGroup.getIceService();
+}
+
+const string & Stream::getPublicIp() const{
+ if (!mPortConfig.multicastIp.empty()){
+ return mPortConfig.multicastIp;
+ }
+ return getMediaSessionPrivate().getMediaLocalIp();
+}
+
+void Stream::finish(){
+}
+
+LINPHONE_END_NAMESPACE
diff --git a/src/conference/session/streams-group.cpp b/src/conference/session/streams-group.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..28b6ccfe701df552775c941ffb52b6a745aea98b
--- /dev/null
+++ b/src/conference/session/streams-group.cpp
@@ -0,0 +1,509 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <bctoolbox/defs.h>
+
+#include "streams.h"
+#include "media-session.h"
+#include "media-session-p.h"
+#include "core/core.h"
+#include "c-wrapper/c-wrapper.h"
+#include "call/call.h"
+#include "call/call-p.h"
+#include "conference/participant.h"
+#include "utils/payload-type-handler.h"
+#include "conference/params/media-session-params-p.h"
+#include "nat/ice-service.h"
+#include "linphone/core.h"
+
+#include <iomanip>
+
+using namespace::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+
+
+/*
+ * StreamsGroup implementation
+ */
+
+StreamsGroup::StreamsGroup(MediaSession &session) : mMediaSession(session){
+ mIceService.reset(new IceService(*this));
+}
+
+StreamsGroup::~StreamsGroup(){
+ finish();
+}
+
+IceService & StreamsGroup::getIceService()const{
+ return *mIceService;
+}
+
+Stream * StreamsGroup::createStream(const OfferAnswerContext ¶ms){
+ Stream *ret = nullptr;
+ SalStreamType type = params.localStreamDescription->type;
+ switch(type){
+ case SalAudio:
+ ret = new MS2AudioStream(*this, params);
+ break;
+ case SalVideo:
+#ifdef VIDEO_ENABLED
+ ret = new MS2VideoStream(*this, params);
+#endif
+ break;
+ case SalText:
+ ret = new MS2RTTStream(*this, params);
+ break;
+ case SalOther:
+ break;
+ }
+ if (!ret){
+ lError() << "Could not create Stream of type " << sal_stream_type_to_string(type);
+ return nullptr;
+ }
+ lInfo() << "Created " << *ret;
+
+ if ((decltype(mStreams)::size_type)params.streamIndex >= mStreams.size()) mStreams.resize(params.streamIndex + 1);
+ if (mStreams[params.streamIndex] != nullptr){
+ lInfo() << "Stream at index " << params.streamIndex << " is being replaced.";
+ }
+ mStreams[params.streamIndex].reset(ret);
+ return ret;
+}
+
+void StreamsGroup::fillLocalMediaDescription(OfferAnswerContext & params){
+ for (auto &stream : mStreams){
+ params.scopeStreamToIndex(stream->getIndex());
+ stream->fillLocalMediaDescription(params);
+ }
+ mIceService->fillLocalMediaDescription(params);
+}
+
+void StreamsGroup::createStreams(const OfferAnswerContext ¶ms){
+ size_t index;
+ for(index = 0; index < (size_t)params.localMediaDescription->nb_streams; ++index){
+ Stream *s;
+ params.scopeStreamToIndexWithDiff(index, mCurrentOfferAnswerState);
+
+ if (params.localStreamDescriptionChanges) {
+ char *differences = sal_media_description_print_differences(params.localStreamDescriptionChanges);
+ lInfo() << "Local stream description has changed: " << differences;
+ ms_free(differences);
+ }
+ if (index >= mStreams.size() || (s = mStreams[index].get()) == nullptr){
+ s = createStream(params);
+ }else{
+ if (s->getType() != params.localStreamDescription->type){
+ lError() << "Inconsistency detected while creating streams. Type has changed from " <<
+ sal_stream_type_to_string(s->getType()) << " to " <<
+ sal_stream_type_to_string(params.localStreamDescription->type) << "!";
+ }else if (params.localStreamDescriptionChanges & SAL_MEDIA_DESCRIPTION_NETWORK_XXXCAST_CHANGED ){
+ /*
+ * Special case: due to implementation constraint, it is necessary to instanciate a new Stream when changing
+ * the cast (uni or multi).
+ */
+ s->stop();
+ s = createStream(params);
+ }
+ }
+ }
+ mIceService->createStreams(params);
+}
+
+bool StreamsGroup::prepare(){
+ if (mFinished){
+ lError() << "StreamsGroup finished, cannot be used anymore.";
+ return false;
+ }
+ for (auto &stream : mStreams){
+ if (stream->getState() == Stream::Stopped){
+ stream->prepare();
+ }
+ }
+ return mIceService->prepare();
+}
+
+void StreamsGroup::finishPrepare(){
+ for (auto &stream : mStreams){
+ if (stream->getState() == Stream::Preparing){
+ stream->finishPrepare();
+ }
+ }
+ mIceService->finishPrepare();
+}
+
+void StreamsGroup::render(const OfferAnswerContext &constParams, CallSession::State targetState){
+ if (mFinished){
+ lError() << "StreamsGroup finished, cannot be used anymore.";
+ return;
+ }
+ OfferAnswerContext params;
+ params.copyFrom(constParams);
+
+ if (params.remoteMediaDescription == nullptr){
+ /* This can happen when we receive a 200Ok without SDP, after early media. In this case we use the previously
+ * provided remote media description.*/
+ params.remoteMediaDescription = mCurrentOfferAnswerState.remoteMediaDescription;
+ }
+
+ for(auto &stream : mStreams){
+ Stream *streamPtr = stream.get();
+ lInfo() << "StreamsGroup " << this << " rendering " << *stream;
+ params.scopeStreamToIndexWithDiff(stream->getIndex(), mCurrentOfferAnswerState);
+
+ if (params.localStreamDescriptionChanges) {
+ char *differences = sal_media_description_print_differences(params.localStreamDescriptionChanges);
+ lInfo() << "Local stream description has changed: " << differences;
+ ms_free(differences);
+ }
+ if (params.resultStreamDescriptionChanges) {
+ char *differences = sal_media_description_print_differences(params.resultStreamDescriptionChanges);
+ lInfo() << "Result stream description has changed: " << differences;
+ ms_free(differences);
+ }
+ if (streamPtr->getState() == Stream::Preparing)
+ streamPtr->finishPrepare();
+ streamPtr->render(params, targetState);
+ }
+ if (!mBandwidthReportTimer){
+ mBandwidthReportTimer = getCore().createTimer([this](){ this->computeAndReportBandwidth(); return true; }, 1000 , "StreamsGroup timer");
+ }
+
+ for(auto &hook : mPostRenderHooks){
+ hook();
+ }
+ mPostRenderHooks.clear();
+
+ mIceService->render(params, targetState);
+
+ if (getIceService().hasCompleted()){
+ /* Should not start dtls until ice is completed */
+ startDtls(params);
+ }
+ /* Save the state of the offer-answer, so that we are later able to monitor differences in next render() calls. */
+ mCurrentOfferAnswerState.dupFrom(params);
+}
+
+void StreamsGroup::sessionConfirmed(const OfferAnswerContext ¶ms){
+ for (auto &stream : mStreams){
+ mCurrentOfferAnswerState.scopeStreamToIndex(stream->getIndex());
+ stream->sessionConfirmed(mCurrentOfferAnswerState);
+ }
+}
+
+void StreamsGroup::stop(){
+ if (mFinished){
+ lError() << "StreamsGroup finished, cannot be used anymore.";
+ abort();
+ return;
+ }
+ if (mBandwidthReportTimer){
+ getCore().destroyTimer(mBandwidthReportTimer);
+ mBandwidthReportTimer = nullptr;
+ }
+ for(auto &stream : mStreams){
+ if (stream && stream->getState() != Stream::Stopped)
+ stream->stop();
+ }
+ mIceService->stop();
+}
+
+Stream * StreamsGroup::getStream(size_t index){
+ if (index >= mStreams.size()){
+ lFatal() << "Bad stream index " << index;
+ return nullptr;
+ }
+ return mStreams[index].get();
+}
+
+bool StreamsGroup::isPortUsed(int port)const{
+ if (port == -1) return false;
+ for(auto &stream : mStreams){
+ if (stream && stream->isPortUsed(port)) return true;
+ }
+ return false;
+}
+
+LinphoneCore *StreamsGroup::getCCore()const{
+ return mMediaSession.getCore()->getCCore();
+}
+
+Core & StreamsGroup::getCore()const{
+ return *mMediaSession.getCore();
+}
+
+MediaSessionPrivate &StreamsGroup::getMediaSessionPrivate()const{
+ return *getMediaSession().getPrivate();
+}
+
+int StreamsGroup::updateAllocatedAudioBandwidth (const PayloadType *pt, int maxbw) {
+ mAudioBandwidth = PayloadTypeHandler::getAudioPayloadTypeBandwidth(pt, maxbw);
+ lInfo() << "Audio bandwidth for StreamsGroup [" << this << "] is " << mAudioBandwidth;
+ return mAudioBandwidth;
+}
+
+int StreamsGroup::getVideoBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc) {
+ int remoteBandwidth = 0;
+ if (desc->bandwidth > 0)
+ remoteBandwidth = desc->bandwidth;
+ else if (md->bandwidth > 0) {
+ /* Case where b=AS is given globally, not per stream */
+ remoteBandwidth = PayloadTypeHandler::getRemainingBandwidthForVideo(md->bandwidth, mAudioBandwidth);
+ }
+ return PayloadTypeHandler::getMinBandwidth(
+ PayloadTypeHandler::getRemainingBandwidthForVideo(linphone_core_get_upload_bandwidth(getCCore()), mAudioBandwidth), remoteBandwidth);
+}
+
+
+void StreamsGroup::zrtpStarted(Stream *mainZrtpStream){
+ for (auto &stream : mStreams){
+ if (stream && stream.get() != mainZrtpStream) stream->zrtpStarted(mainZrtpStream);
+ }
+ propagateEncryptionChanged();
+}
+
+bool StreamsGroup::allStreamsEncrypted () const {
+ int activeStreamsCount = 0;
+ for (auto &stream : mStreams){
+ if (stream->getState() == Stream::Running){
+ ++activeStreamsCount;
+ if (!stream->isEncrypted()){
+ return false;
+ }
+ }
+ }
+ return activeStreamsCount > 0;
+}
+
+
+void StreamsGroup::propagateEncryptionChanged () {
+ getMediaSessionPrivate().propagateEncryptionChanged();
+}
+
+void StreamsGroup::authTokenReady(const string &authToken, bool verified) {
+ mAuthToken = authToken;
+ mAuthTokenVerified = verified;
+ lInfo() << "Authentication token is " << mAuthToken << "(" << (mAuthTokenVerified ? "verified" : "unverified") << ")";
+}
+
+void StreamsGroup::setAuthTokenVerified(bool value){
+ MS2Stream *s = lookupMainStreamInterface<MS2Stream>(SalAudio);
+ if (!s || s->getState() != Stream::Running){
+ lError() << "StreamsGroup::setAuthTokenVerified(): No audio stream or not started";
+ return;
+ }
+ MSZrtpContext *zrtp_context = s->getZrtpContext();
+ if (!zrtp_context) {
+ lError() << "StreamsGroup::setAuthenticationTokenVerified(): No zrtp context";
+ return;
+ }
+ // SAS verified
+ if (value) {
+ ms_zrtp_sas_verified(zrtp_context);
+ } else { // SAS rejected
+ ms_zrtp_sas_reset_verified(zrtp_context);
+ }
+ mAuthTokenVerified = value;
+}
+
+Stream * StreamsGroup::lookupMainStream(SalStreamType type){
+ for (auto &stream : mStreams){
+ if (stream->isMain() && stream->getType() == type){
+ return stream.get();
+ }
+ }
+ return nullptr;
+}
+
+
+void StreamsGroup::tryEarlyMediaForking(const OfferAnswerContext ¶ms) {
+ for (auto & s : mStreams) {
+ params.scopeStreamToIndex(s->getIndex());
+ if (!sal_stream_description_enabled(params.resultStreamDescription) || params.resultStreamDescription->dir == SalStreamInactive)
+ continue;
+
+ const SalStreamDescription *refStream = params.resultStreamDescription;
+ const SalStreamDescription *newStream = params.remoteStreamDescription;
+
+ if ((refStream->type == newStream->type) && refStream->payloads && newStream->payloads) {
+ OrtpPayloadType *refpt = static_cast<OrtpPayloadType *>(refStream->payloads->data);
+ OrtpPayloadType *newpt = static_cast<OrtpPayloadType *>(newStream->payloads->data);
+ if ((strcmp(refpt->mime_type, newpt->mime_type) == 0) && (refpt->clock_rate == newpt->clock_rate)
+ && (payload_type_get_number(refpt) == payload_type_get_number(newpt))) {
+ s->tryEarlyMediaForking(params);
+ }
+ }
+ }
+}
+
+void StreamsGroup::finishEarlyMediaForking(){
+ for (auto &stream : mStreams){
+ if (stream) stream->finishEarlyMediaForking();
+ }
+}
+
+bool StreamsGroup::isStarted()const{
+ for( auto & stream : mStreams){
+ if (stream->getState() == Stream::Running) return true;
+ }
+ return false;
+}
+
+void StreamsGroup::clearStreams(){
+ stop();
+ mIceService.reset(new IceService(*this));
+ mStreams.clear();
+ mCurrentOfferAnswerState.clear();
+}
+
+size_t StreamsGroup::getActiveStreamsCount() const{
+ size_t ret = 0;
+ for( auto & stream : mStreams){
+ if (stream->getState() == Stream::Running) ++ret;
+ }
+ return ret;
+}
+
+bool StreamsGroup::isMuted() const{
+ for (auto & stream : mStreams){
+ if (stream->getState() == Stream::Running){
+ if (stream->isMuted() == false) return false;
+ }
+ }
+ return true;
+}
+
+template< typename _functor>
+float StreamsGroup::computeOverallQuality(_functor func){
+ float globalRating = -1.0f;
+ int countedStreams = 0;
+ for (auto &stream : mStreams){
+ float streamRating = func(stream.get());
+ if (streamRating != -1.0f){
+ if (globalRating == -1.0f){
+ globalRating = streamRating;
+ }else{
+ globalRating += streamRating;
+ }
+ countedStreams++;
+ }
+ }
+ return globalRating / (float)countedStreams;
+}
+
+float StreamsGroup::getAverageQuality(){
+ return computeOverallQuality(mem_fun(&Stream::getAverageQuality));
+}
+
+float StreamsGroup::getCurrentQuality(){
+ return computeOverallQuality(mem_fun(&Stream::getCurrentQuality));
+}
+
+void StreamsGroup::startDtls(const OfferAnswerContext ¶ms){
+ for( auto & stream : mStreams){
+ params.scopeStreamToIndex(stream->getIndex());
+ stream->startDtls(params);
+ }
+}
+
+int StreamsGroup::getAvpfRrInterval()const{
+ int interval = 0;
+ for( auto & stream : mStreams){
+ RtpInterface *i = dynamic_cast<MS2Stream*>(stream.get());
+ if (i && i->getAvpfRrInterval() > interval)
+ interval = i->getAvpfRrInterval();
+ }
+ return interval;
+}
+
+bool StreamsGroup::avpfEnabled() const{
+ bool ret = false;
+ for( auto & stream : mStreams){
+ RtpInterface *i = dynamic_cast<MS2Stream*>(stream.get());
+ if (i && stream->getState() == Stream::Running){
+ if (!i->avpfEnabled()){
+ return false;
+ }
+ ret = true;
+ }
+ }
+ return ret;
+}
+
+void StreamsGroup::refreshSockets(){
+ forEach<Stream>(mem_fun(&Stream::refreshSockets));
+}
+
+void StreamsGroup::computeAndReportBandwidth(){
+ forEach<Stream>(mem_fun(&Stream::updateBandwidthReports));
+
+ if (!bctbx_log_level_enabled(BCTBX_LOG_DOMAIN, BCTBX_LOG_MESSAGE)) return;
+
+ ostringstream ostr;
+ bool introDone = false;
+
+ for (auto &stream : mStreams){
+ if (!stream) continue;
+ if (stream->getState() != Stream::Running) continue;
+ LinphoneCallStats *stats = stream->getStats();
+ if (!introDone){
+ ostr << "Bandwidth usage for CallSession [" << &getMediaSession() << "]:" << endl << fixed << setprecision(2);
+ introDone = true;
+ }
+ ostr << "\tStream #" << stream->getIndex() << " (" << sal_stream_type_to_string(stream->getType()) << ") | cpu: " << stream->getCpuUsage() << "% |" << " RTP : [d="
+ << linphone_call_stats_get_download_bandwidth(stats) << ",u=" << linphone_call_stats_get_upload_bandwidth(stats) << "] "
+ << "RTCP: [d=" << linphone_call_stats_get_rtcp_download_bandwidth(stats) << ",u=" << linphone_call_stats_get_rtcp_upload_bandwidth(stats) << "] ";
+ float est_bw = linphone_call_stats_get_estimated_download_bandwidth(stats);
+ if (est_bw > 0.0) ostr << "Est max d=" << est_bw;
+ ostr << " (kbits/sec)" << endl;
+ }
+ lInfo() << ostr.str();
+}
+
+void StreamsGroup::addPostRenderHook(const std::function<void()> &l){
+ mPostRenderHooks.push_back(l);
+}
+
+void StreamsGroup::setStreamMain(size_t index){
+ Stream *s = getStream(index);
+ if (s){
+ SalStreamType type = s->getType();
+ // Make sure there is not already a "main" stream; which would be a programmer fault.
+ Stream *other = lookupMainStream(type);
+ if (other != nullptr && other != s){
+ lError() << "StreamsGroup::setStreamMain(): error, the main attribute has already been set on another stream.";
+ return;
+ }
+ s->setMain();
+ }
+}
+
+void StreamsGroup::finish(){
+ if (mFinished) return;
+ lInfo() << "StreamsGroup::finish() called.";
+ stop(); //For the paranoid: normally it should be done already.
+ mIceService->finish(); // finish ICE first, as it has actions on the streams.
+ forEach<Stream>(mem_fun(&Stream::finish));
+ mFinished = true;
+}
+
+
+LINPHONE_END_NAMESPACE
+
diff --git a/src/conference/session/streams.h b/src/conference/session/streams.h
new file mode 100644
index 0000000000000000000000000000000000000000..fabb81f72b7e89785cab4ce4b6f185977a5e8e08
--- /dev/null
+++ b/src/conference/session/streams.h
@@ -0,0 +1,386 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef streams_h
+#define streams_h
+
+#include <vector>
+#include <memory>
+
+#include "port-config.h"
+#include "call-session.h"
+#include "media-description-renderer.h"
+
+LINPHONE_BEGIN_NAMESPACE
+
+
+class StreamsGroup;
+class MediaSession;
+class MediaSessionPrivate;
+class MediaSessionParams;
+class IceService;
+
+/**
+ * Base class for any kind of stream that may be setup with SDP.
+ */
+class Stream : public MediaDescriptionRenderer{
+ friend class StreamsGroup;
+public:
+ enum State{
+ Stopped,
+ Preparing,
+ Running
+ };
+
+ virtual void fillLocalMediaDescription(OfferAnswerContext & ctx) override;
+ /**
+ * Ask the stream to prepare to run. This may include configuration steps, ICE gathering etc.
+ */
+ virtual bool prepare() override;
+
+ /**
+ * Request the stream to finish the prepare step (such as ICE gathering).
+ */
+ virtual void finishPrepare() override;
+ /**
+ * Ask the stream to render according to the supplied offer-answer context and target state.
+ * render() may be called multiple times according to changes made in the offer answer.
+ */
+ virtual void render(const OfferAnswerContext & ctx, CallSession::State targetState) override;
+ /**
+ * Notifies that session is confirmed (called by signaling).
+ */
+ virtual void sessionConfirmed(const OfferAnswerContext &ctx) override;
+
+ /**
+ * Ask the stream to stop. A call to prepare() is necessary before doing a future render() operation, if any.
+ */
+ virtual void stop() override;
+
+ /**
+ * Notifies the stream that it will no longer be used (called in render() ).
+ * This gives the opportunity to free any useless resource immediately.
+ * Statistics (LinphoneCallStats ) must remain until destruction.
+ */
+ virtual void finish() override;
+ virtual LinphoneCallStats *getStats(){
+ return nullptr;
+ }
+ /**
+ * Called by the IceService to setup the check list to run with the stream.
+ */
+ virtual void setIceCheckList(IceCheckList *cl);
+ /**
+ * Called by the IceService to notify the stream of a state change in the ICE check list or the ICE session.
+ */
+ virtual void iceStateChanged();
+ virtual bool isEncrypted() const = 0;
+ virtual void tryEarlyMediaForking(const OfferAnswerContext &ctx) = 0;
+ virtual void finishEarlyMediaForking() = 0;
+ virtual float getCurrentQuality() = 0;
+ virtual float getAverageQuality() = 0;
+ virtual void startDtls(const OfferAnswerContext ¶ms) = 0;
+ virtual bool isMuted()const = 0;
+ virtual void refreshSockets() = 0;
+ virtual void updateBandwidthReports() = 0;
+ virtual float getCpuUsage()const = 0;
+ size_t getIndex()const { return mIndex; }
+ SalStreamType getType()const{ return mStreamType;}
+ LinphoneCore *getCCore()const;
+ Core &getCore()const;
+ MediaSession &getMediaSession()const;
+ MediaSessionPrivate &getMediaSessionPrivate()const;
+ bool isPortUsed(int port) const;
+ IceService & getIceService()const;
+ State getState()const{ return mState;}
+ StreamsGroup &getGroup()const{ return mStreamsGroup;}
+ // Returns whether this stream is the "main" one of its own type, in constrat to secondary streams.
+ bool isMain()const{ return mIsMain;}
+ int getStartCount()const{ return mStartCount; }
+ const PortConfig &getPortConfig()const{ return mPortConfig; }
+ virtual ~Stream() = default;
+ static std::string stateToString(State st){
+ switch(st){
+ case Stopped:
+ return "Stopped";
+ case Running:
+ return "Running";
+ case Preparing:
+ return "Preparing";
+ }
+ return "undefined";
+ }
+
+protected:
+ Stream(StreamsGroup &ms, const OfferAnswerContext ¶ms);
+ /**
+ * Notifies that zrtp primary stream is now secured.
+ */
+ virtual void zrtpStarted(Stream *mainZrtpStream){};
+ const std::string & getPublicIp() const;
+ PortConfig mPortConfig;
+ int mStartCount = 0; /* The number of time of the underlying stream has been started (or restarted). To be maintained by implementations. */
+private:
+ void setMain();
+ void setPortConfig(std::pair<int, int> portRange);
+ int selectFixedPort(std::pair<int, int> portRange);
+ int selectRandomPort(std::pair<int, int> portRange);
+ void setPortConfig();
+ void setRandomPortConfig();
+ void fillMulticastMediaAddresses();
+ StreamsGroup & mStreamsGroup;
+ const SalStreamType mStreamType;
+ const size_t mIndex;
+ State mState = Stopped;
+ bool mIsMain = false;
+};
+
+inline std::ostream &operator<<(std::ostream & ostr, SalStreamType type){
+ ostr << sal_stream_type_to_string(type);
+ return ostr;
+}
+
+inline std::ostream & operator<<(std::ostream & ostr, const Stream& stream){
+ ostr << "stream#" << stream.getIndex() << " [" << stream.getType() << "] in state [" << Stream::stateToString(stream.getState()) << "]";
+ return ostr;
+}
+
+
+class AudioControlInterface{
+public:
+ virtual void enableMic(bool value) = 0;
+ virtual void enableSpeaker(bool value) = 0;
+ virtual bool micEnabled()const = 0;
+ virtual bool speakerEnabled()const = 0;
+ virtual void startRecording() = 0;
+ virtual void stopRecording() = 0;
+ virtual bool isRecording() = 0;
+ virtual float getPlayVolume() = 0; /* Measured playback volume */
+ virtual float getRecordVolume() = 0; /* Measured record volume */
+ virtual float getMicGain() = 0;
+ virtual void setMicGain(float value) = 0;
+ virtual float getSpeakerGain() = 0;
+ virtual void setSpeakerGain(float value) = 0;
+ virtual void setRoute(LinphoneAudioRoute route) = 0;
+ virtual void sendDtmf(int dtmf) = 0;
+ virtual void enableEchoCancellation(bool value) = 0;
+ virtual bool echoCancellationEnabled()const = 0;
+ virtual ~AudioControlInterface() = default;
+};
+
+class VideoControlInterface{
+public:
+ struct VideoStats{
+ float fps;
+ int width, height;
+ };
+ virtual void sendVfu() = 0;
+ virtual void sendVfuRequest() = 0;
+ virtual void enableCamera(bool value) = 0;
+ virtual bool cameraEnabled() const = 0;
+ virtual void setNativeWindowId(void *w) = 0;
+ virtual void * getNativeWindowId() const = 0;
+ virtual void setNativePreviewWindowId(void *w) = 0;
+ virtual void * getNativePreviewWindowId() const = 0;
+ virtual void parametersChanged() = 0;
+ virtual void requestNotifyNextVideoFrameDecoded () = 0;
+ virtual int takePreviewSnapshot (const std::string& file) = 0;
+ virtual int takeVideoSnapshot (const std::string& file) = 0;
+ virtual void zoomVideo (float zoomFactor, float cx, float cy) = 0;
+ virtual void getRecvStats(VideoStats *s) const = 0;
+ virtual void getSendStats(VideoStats *s) const = 0;
+ virtual ~VideoControlInterface() = default;
+};
+
+/*
+ * Interface to query RTP-related information.
+ */
+class RtpInterface{
+public:
+ virtual bool avpfEnabled() const = 0;
+ virtual bool bundleEnabled() const = 0;
+ virtual int getAvpfRrInterval() const = 0;
+ /*
+ * Returns true if the stream has its own transport interface.
+ * This is always true unless rtp bundle mode is on, in which case a stream that is using the transport from another
+ * stream will return false.
+ */
+ virtual bool isTransportOwner() const = 0;
+ virtual ~RtpInterface() = default;
+};
+
+
+
+/**
+ * The StreamsGroup takes in charge the initialization and rendering of a group of streams defined
+ * according to a local media description, and a media description resulted from the offer/answer model.
+ * When the offer is received from remote, the local description must be compatible with the remote offer.
+ * The StreamsGroup is not in charge of offer/answer model logic: just the creation, rendering, and destruction of the
+ * streams.
+ */
+class StreamsGroup : public MediaDescriptionRenderer{
+ friend class Stream;
+ friend class MS2Stream;
+ friend class MS2AudioStream;
+public:
+ StreamsGroup(MediaSession &session);
+ ~StreamsGroup();
+ /**
+ * Create the streams according to the specified local and remote description.
+ * The port and transport addresses are filled into the local description in return.
+ * The local media description must not be null, the remote media description must not be null only
+ * when the offer was received from remote side.
+ */
+ void createStreams(const OfferAnswerContext ¶ms);
+
+ /**
+ * Set the "main" attribute to a stream index.
+ * There can be only one main stream per type (audio, video, text...).
+ * This attribute is useful to know whether certains tasks must be done on these streams.
+ */
+ void setStreamMain(size_t index);
+
+ /**
+ * Once the streams are created, update the local media description to fill mainly
+ * transport addresses, which are usually provided by the media layer.
+ */
+ virtual void fillLocalMediaDescription(OfferAnswerContext & ctx) override;
+ /*
+ * Request the streams to prepare (configuration steps, ice gathering.
+ * Returns false if ready, true if prepare() requires more time, in which case
+ * ICE events will be submitted to the MediaSession to inform when ready to proceed.
+ */
+ virtual bool prepare() override;
+ /**
+ * Request the stream to finish the prepare step (such as ICE gathering).
+ */
+ virtual void finishPrepare() override;
+ /**
+ * Render the streams according to the supplied offer answer parameters and target session state.
+ * Local, remote and result must all be non-null.
+ */
+ virtual void render(const OfferAnswerContext ¶ms, CallSession::State targetState) override;
+ /**
+ * Used by signaling to notify that the session is confirmed (typically, when an ACK is received.
+ */
+ virtual void sessionConfirmed(const OfferAnswerContext ¶ms) override;
+
+ /**
+ * Stop streams.
+ */
+ virtual void stop() override;
+ /**
+ * Notifies the stream that it will no longer be used (called in render() ).
+ * This gives the opportunity to free any useless resource immediately.
+ * Statistics (LinphoneCallStats ) must remain until destruction.
+ */
+ virtual void finish() override;
+ Stream * getStream(size_t index);
+ Stream * getStream(int index){
+ return getStream((size_t) index);
+ }
+ Stream * lookupMainStream(SalStreamType type);
+ template <typename _interface>
+ _interface * lookupMainStreamInterface(SalStreamType type){
+ Stream *s = lookupMainStream(type);
+ if (s){
+ _interface *iface = dynamic_cast<_interface*>(s);
+ if (iface == nullptr){
+ lError() << "lookupMainStreamInterface(): stream " << s << " cannot be casted to " << typeid(_interface).name();
+ }
+ return iface;
+ }
+ return nullptr;
+ }
+ const std::vector<std::unique_ptr<Stream>> & getStreams(){
+ return mStreams;
+ }
+ MediaSession &getMediaSession()const{
+ return mMediaSession;
+ }
+ bool isPortUsed(int port)const;
+ IceService &getIceService()const;
+ bool allStreamsEncrypted () const;
+ // Returns true if at least one stream was started.
+ bool isStarted()const;
+ // Returns true if all streams are muted (from local source standpoint).
+ bool isMuted() const;
+ // Returns true if all streams have avpf enabled.
+ bool avpfEnabled() const;
+ int getAvpfRrInterval()const;
+ void startDtls(const OfferAnswerContext ¶ms);
+ void tryEarlyMediaForking(const OfferAnswerContext &ctx);
+ void finishEarlyMediaForking();
+ /*
+ * Iterates over streams, trying to cast them to the _requestedInterface type. If they do cast,
+ * invoke the lambda expression on them.
+ */
+ template <typename _requestedInterface, typename _lambda>
+ void forEach(const _lambda &l){
+ for (auto & stream : mStreams){
+ _requestedInterface * iface = dynamic_cast<_requestedInterface*>(stream.get());
+ if (iface) l(iface);
+ }
+ }
+ void clearStreams();
+ float getCurrentQuality();
+ float getAverageQuality();
+ const std::string &getAuthToken()const{ return mAuthToken; };
+ void setAuthTokenVerified(bool value);
+ size_t getActiveStreamsCount() const;
+ size_t size()const{ return mStreams.size(); }
+ void refreshSockets();
+ const std::string & getAuthenticationToken()const{ return mAuthToken; }
+ bool getAuthenticationTokenVerified() const{ return mAuthTokenVerified; }
+ const OfferAnswerContext & getCurrentOfferAnswerContext()const{ return mCurrentOfferAnswerState; };
+ MediaSessionPrivate &getMediaSessionPrivate()const;
+ LinphoneCore *getCCore()const;
+ Core & getCore()const;
+protected:
+
+ int updateAllocatedAudioBandwidth (const PayloadType *pt, int maxbw);
+ int getVideoBandwidth (const SalMediaDescription *md, const SalStreamDescription *desc);
+ void zrtpStarted(Stream *mainZrtpStream);
+ void propagateEncryptionChanged();
+ void authTokenReady(const std::string &token, bool verified);
+ void addPostRenderHook(const std::function<void()> &l);
+private:
+ template< typename _functor>
+ float computeOverallQuality(_functor func);
+ Stream * createStream(const OfferAnswerContext ¶m);
+ MediaSession &mMediaSession;
+ std::unique_ptr<IceService> mIceService;
+ std::vector<std::unique_ptr<Stream>> mStreams;
+ void computeAndReportBandwidth();
+ // Upload bandwidth used by audio.
+ int mAudioBandwidth = 0;
+ // Zrtp auth token
+ std::string mAuthToken;
+ belle_sip_source_t *mBandwidthReportTimer = nullptr;
+ std::list<std::function<void()>> mPostRenderHooks;
+ OfferAnswerContext mCurrentOfferAnswerState;
+ bool mAuthTokenVerified = false;
+ bool mFinished = false;
+
+};
+
+LINPHONE_END_NAMESPACE
+
+#endif
+
diff --git a/src/conference/session/tone-manager.cpp b/src/conference/session/tone-manager.cpp
index 3f7ab66084ceaf257340271bae9803322a15e8fc..f5217a47cf5a59a511cd9477affba8fe02df8fc2 100644
--- a/src/conference/session/tone-manager.cpp
+++ b/src/conference/session/tone-manager.cpp
@@ -236,7 +236,7 @@ void ToneManager::createTimerToCleanTonePlayer(unsigned int delay) {
return true;
};
- mTimer = getCore()->createTimer(callback, delay);
+ mTimer = getCore()->createTimer(callback, delay, "Tone player cleanup");
}
}
@@ -406,12 +406,6 @@ void ToneManager::doStartRingbackTone(const std::shared_ptr<CallSession> &sessio
return;
MSSndCard *ringCard = lc->sound_conf.lsd_card ? lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
- int maxRate = std::static_pointer_cast<MediaSession>(session)->getPrivate()->getLocalDesc()->streams[0].max_rate;
- if (maxRate > 0) ms_snd_card_set_preferred_sample_rate(ringCard, maxRate);
-
- /* We release sound before playing ringback tone */
- AudioStream *as = reinterpret_cast<AudioStream *>(std::static_pointer_cast<MediaSession>(session)->getPrivate()->getMediaStream(LinphoneStreamTypeAudio));
- if (as) audio_stream_unprepare_sound(as);
if (lc->sound_conf.remote_ring) {
ms_snd_card_set_stream_type(ringCard, MS_SND_CARD_STREAM_VOICE);
diff --git a/src/conference/session/video-stream.cpp b/src/conference/session/video-stream.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..0d9051a61fde9ca9e6c6082e62ed40cb850d8765
--- /dev/null
+++ b/src/conference/session/video-stream.cpp
@@ -0,0 +1,513 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+
+#ifdef VIDEO_ENABLED
+
+#include "bctoolbox/defs.h"
+
+#include "ms2-streams.h"
+#include "media-session.h"
+#include "media-session-p.h"
+#include "core/core.h"
+#include "c-wrapper/c-wrapper.h"
+#include "call/call.h"
+#include "call/call-p.h"
+#include "conference/participant.h"
+#include "conference/params/media-session-params-p.h"
+
+#include "mediastreamer2/msjpegwriter.h"
+#include "mediastreamer2/msogl.h"
+
+#include "linphone/core.h"
+
+using namespace::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+/*
+ * MS2VideoStream implemenation
+ */
+
+MS2VideoStream::MS2VideoStream(StreamsGroup &sg, const OfferAnswerContext ¶ms) : MS2Stream(sg, params){
+ string bindIp = getBindIp();
+ mStream = video_stream_new2(getCCore()->factory, bindIp.empty() ? nullptr : bindIp.c_str(), mPortConfig.rtpPort, mPortConfig.rtcpPort);
+ initializeSessions(&mStream->ms);
+}
+
+void MS2VideoStream::sVideoStreamEventCb (void *userData, const MSFilter *f, const unsigned int eventId, const void *args) {
+ MS2VideoStream *zis = static_cast<MS2VideoStream*>(userData);
+ zis->videoStreamEventCb(f, eventId, args);
+}
+
+
+void MS2VideoStream::videoStreamEventCb (const MSFilter *f, const unsigned int eventId, const void *args) {
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+
+ switch (eventId) {
+ case MS_VIDEO_DECODER_DECODING_ERRORS:
+ lWarning() << "MS_VIDEO_DECODER_DECODING_ERRORS";
+ if (mStream && video_stream_is_decoding_error_to_be_reported(mStream, 5000)) {
+ video_stream_decoding_error_reported(mStream);
+ sendVfu();
+ }
+ break;
+ case MS_VIDEO_DECODER_RECOVERED_FROM_ERRORS:
+ lInfo() << "MS_VIDEO_DECODER_RECOVERED_FROM_ERRORS";
+ if (mStream)
+ video_stream_decoding_error_recovered(mStream);
+ break;
+ case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
+ lInfo() << "First video frame decoded successfully";
+ if (listener)
+ listener->onFirstVideoFrameDecoded(getMediaSession().getSharedFromThis());
+ break;
+ case MS_VIDEO_DECODER_SEND_PLI:
+ case MS_VIDEO_DECODER_SEND_SLI:
+ case MS_VIDEO_DECODER_SEND_RPSI:
+ /* Handled internally by mediastreamer2 */
+ break;
+ case MS_CAMERA_PREVIEW_SIZE_CHANGED: {
+ MSVideoSize size = *(MSVideoSize *)args;
+ lInfo() << "Camera video preview size changed: " << size.width << "x" << size.height;
+ linphone_core_resize_video_preview(getCCore(), size.width, size.height);
+ break;
+ }
+ default:
+ lWarning() << "Unhandled event " << eventId;
+ break;
+ }
+}
+
+MediaStream *MS2VideoStream::getMediaStream()const{
+ return &mStream->ms;
+}
+
+void MS2VideoStream::sendVfu(){
+ video_stream_send_vfu(mStream);
+}
+
+void MS2VideoStream::sendVfuRequest(){
+ video_stream_send_fir(mStream);
+}
+
+void MS2VideoStream::zoomVideo (float zoomFactor, float cx, float cy){
+ if (mStream->output) {
+ if (zoomFactor < 1)
+ zoomFactor = 1;
+ float halfsize = 0.5f * 1.0f / zoomFactor;
+ if ((cx - halfsize) < 0)
+ cx = 0 + halfsize;
+ if ((cx + halfsize) > 1)
+ cx = 1 - halfsize;
+ if ((cy - halfsize) < 0)
+ cy = 0 + halfsize;
+ if ((cy + halfsize) > 1)
+ cy = 1 - halfsize;
+ float zoom[3] = { zoomFactor, cx, cy };
+ ms_filter_call_method(mStream->output, MS_VIDEO_DISPLAY_ZOOM, &zoom);
+ } else
+ lWarning() << "Could not apply zoom: video output wasn't activated";
+}
+
+void MS2VideoStream::parametersChanged(){
+ if (getState() != Stream::Running) return;
+ const LinphoneVideoDefinition *vdef = linphone_core_get_preferred_video_definition(getCCore());
+ MSVideoSize vsize;
+ vsize.width = static_cast<int>(linphone_video_definition_get_width(vdef));
+ vsize.height = static_cast<int>(linphone_video_definition_get_height(vdef));
+ video_stream_set_sent_video_size(mStream, vsize);
+ video_stream_set_fps(mStream, linphone_core_get_preferred_framerate(getCCore()));
+ if (mCameraEnabled && (mStream->cam != getCCore()->video_conf.device))
+ video_stream_change_camera(mStream, getCCore()->video_conf.device);
+ else
+ video_stream_update_video_params(mStream);
+}
+
+void MS2VideoStream::setNativeWindowId(void *w){
+ mNativeWindowId = w;
+ video_stream_set_native_window_id(mStream, w);
+}
+
+void * MS2VideoStream::getNativeWindowId() const{
+ if (mNativeWindowId){
+ return mNativeWindowId;
+ }
+ /* It was not set but we want to get the one automatically created by mediastreamer2 (desktop versions only) */
+ return video_stream_get_native_window_id(mStream);
+}
+
+void MS2VideoStream::setNativePreviewWindowId(void *w){
+ mNativePreviewWindowId = w;
+ video_stream_set_native_preview_window_id(mStream, w);
+}
+
+void * MS2VideoStream::getNativePreviewWindowId() const{
+ return mNativePreviewWindowId;
+}
+
+void MS2VideoStream::enableCamera(bool value){
+ mCameraEnabled = value;
+ MSWebCam *videoDevice = getVideoDevice(getMediaSession().getState());
+ if (video_stream_started(mStream) && (video_stream_get_camera(mStream) != videoDevice)) {
+ string currentCam = video_stream_get_camera(mStream) ? ms_web_cam_get_name(video_stream_get_camera(mStream)) : "NULL";
+ string newCam = videoDevice ? ms_web_cam_get_name(videoDevice) : "NULL";
+ lInfo() << "Switching video cam from [" << currentCam << "] to [" << newCam << "]";
+ video_stream_change_camera(mStream, videoDevice);
+ }
+}
+
+MSWebCam * MS2VideoStream::getVideoDevice(CallSession::State targetState) const {
+ bool paused = (targetState == CallSession::State::Pausing) || (targetState == CallSession::State::Paused);
+ if (paused || mMuted || !mCameraEnabled)
+ return ms_web_cam_manager_get_cam(ms_factory_get_web_cam_manager(getCCore()->factory),
+ "StaticImage: Static picture");
+ else
+ return getCCore()->video_conf.device;
+}
+
+void MS2VideoStream::activateZrtp(){
+ if (linphone_core_media_encryption_supported(getCCore(), LinphoneMediaEncryptionZRTP)){
+ Stream *audioStream = getGroup().lookupMainStream(SalAudio);
+ if (audioStream){
+ MS2AudioStream *msa = dynamic_cast<MS2AudioStream*>(audioStream);
+ video_stream_enable_zrtp(mStream, (AudioStream*)msa->getMediaStream());
+ // Since the zrtp session is now initialized, make sure it is retained for future use.
+ media_stream_reclaim_sessions((MediaStream*)mStream, &mSessions);
+ video_stream_start_zrtp(mStream);
+ }else{
+ lError() << "Error while enabling zrtp on video stream: the audio stream isn't known. This is unsupported.";
+ }
+ }
+}
+
+
+bool MS2VideoStream::prepare(){
+
+ MS2Stream::prepare();
+ video_stream_prepare_video(mStream);
+ return false;
+}
+
+void MS2VideoStream::finishPrepare(){
+ MS2Stream::finishPrepare();
+ video_stream_unprepare_video(mStream);
+}
+
+void MS2VideoStream::render(const OfferAnswerContext & ctx, CallSession::State targetState){
+ bool reusedPreview = false;
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+
+ /* Shutdown preview */
+ MSFilter *source = nullptr;
+ if (getCCore()->previewstream) {
+ if (getCCore()->video_conf.reuse_preview_source)
+ source = video_preview_stop_reuse_source(getCCore()->previewstream);
+ else
+ video_preview_stop(getCCore()->previewstream);
+ getCCore()->previewstream = nullptr;
+ }
+ const SalStreamDescription *vstream = ctx.resultStreamDescription;
+
+ bool basicChangesHandled = handleBasicChanges(ctx, targetState);
+
+ if (basicChangesHandled) {
+ bool muted = mMuted;
+ if (getState() == Running) {
+ MS2Stream::render(ctx, targetState); // MS2Stream::render() may decide to unmute.
+ if (muted && !mMuted) {
+ lInfo() << "Early media finished, unmuting video input...";
+ /* We were in early media, now we want to enable real media */
+ mMuted = false;
+ enableCamera(mCameraEnabled);
+ }
+ }
+ return;
+ }
+
+ int usedPt = -1;
+ RtpProfile *videoProfile = makeProfile(ctx.resultMediaDescription, vstream, &usedPt);
+ if (usedPt == -1){
+ lError() << "No payload types accepted for video stream !";
+ stop();
+ return;
+ }
+
+
+ video_stream_enable_display_filter_auto_rotate(mStream,
+ !!lp_config_get_int(linphone_core_get_config(getCCore()), "video", "display_filter_auto_rotate", 0)
+ );
+
+ const char *displayFilter = linphone_core_get_video_display_filter(getCCore());
+ if (displayFilter)
+ video_stream_set_display_filter_name(mStream, displayFilter);
+ video_stream_set_event_callback(mStream, sVideoStreamEventCb, this);
+ if (isMain()){
+ getMediaSessionPrivate().getCurrentParams()->getPrivate()->setUsedVideoCodec(rtp_profile_get_payload(videoProfile, usedPt));
+ }
+
+ if (getCCore()->video_conf.preview_vsize.width != 0)
+ video_stream_set_preview_size(mStream, getCCore()->video_conf.preview_vsize);
+ video_stream_set_fps(mStream, linphone_core_get_preferred_framerate(getCCore()));
+ if (lp_config_get_int(linphone_core_get_config(getCCore()), "video", "nowebcam_uses_normal_fps", 0))
+ mStream->staticimage_webcam_fps_optimization = false;
+ const LinphoneVideoDefinition *vdef = linphone_core_get_preferred_video_definition(getCCore());
+ MSVideoSize vsize;
+ vsize.width = static_cast<int>(linphone_video_definition_get_width(vdef));
+ vsize.height = static_cast<int>(linphone_video_definition_get_height(vdef));
+ video_stream_set_sent_video_size(mStream, vsize);
+ video_stream_enable_self_view(mStream, getCCore()->video_conf.selfview);
+ if (mNativeWindowId)
+ video_stream_set_native_window_id(mStream, mNativeWindowId);
+ else if (getCCore()->video_window_id)
+ video_stream_set_native_window_id(mStream, getCCore()->video_window_id);
+ if (getCCore()->preview_window_id)
+ video_stream_set_native_preview_window_id(mStream, getCCore()->preview_window_id);
+ video_stream_use_preview_video_window(mStream, getCCore()->use_preview_window);
+
+ MS2Stream::render(ctx, targetState);
+
+ RtpAddressInfo dest;
+ getRtpDestination(ctx, &dest);
+ MediaStreamDir dir = MediaStreamSendRecv;
+
+ if ((vstream->dir == SalStreamSendOnly) && getCCore()->video_conf.capture)
+ dir = MediaStreamSendOnly;
+ else if ((vstream->dir == SalStreamRecvOnly) && getCCore()->video_conf.display)
+ dir = MediaStreamRecvOnly;
+ else if (vstream->dir == SalStreamSendRecv) {
+ if (getCCore()->video_conf.display && getCCore()->video_conf.capture)
+ dir = MediaStreamSendRecv;
+ else if (getCCore()->video_conf.display)
+ dir = MediaStreamRecvOnly;
+ else
+ dir = MediaStreamSendOnly;
+ }else {
+ lWarning() << "Video stream is inactive";
+ /* Either inactive or incompatible with local capabilities */
+ stop();
+ return;
+ }
+ if (vstream->multicast_role == SalMulticastReceiver){
+ dir = MediaStreamRecvOnly;
+ }else if (vstream->multicast_role == SalMulticastSender){
+ dir = MediaStreamSendOnly;
+ }
+
+ MSWebCam *cam = getVideoDevice(targetState);
+
+ getMediaSession().getLog()->video_enabled = true;
+ video_stream_set_direction(mStream, dir);
+ lInfo() << "Device rotation =" << getCCore()->device_rotation;
+ video_stream_set_device_rotation(mStream, getCCore()->device_rotation);
+ video_stream_set_freeze_on_error(mStream, !!lp_config_get_int(linphone_core_get_config(getCCore()), "video", "freeze_on_error", 1));
+ video_stream_use_video_preset(mStream, lp_config_get_string(linphone_core_get_config(getCCore()), "video", "preset", nullptr));
+ if (getCCore()->video_conf.reuse_preview_source && source) {
+ lInfo() << "video_stream_start_with_source kept: " << source;
+ video_stream_start_with_source(mStream, videoProfile, dest.rtpAddr.c_str(), dest.rtpPort, dest.rtcpAddr.c_str(),
+ dest.rtcpPort,
+ usedPt, -1, cam, source);
+ reusedPreview = true;
+ } else {
+ bool ok = true;
+ MSMediaStreamIO io = MS_MEDIA_STREAM_IO_INITIALIZER;
+ if (linphone_config_get_bool(linphone_core_get_config(getCCore()), "video", "rtp_io", FALSE)) {
+ io.input.type = io.output.type = MSResourceRtp;
+ io.input.session = io.output.session = createRtpIoSession();
+ if (!io.input.session) {
+ ok = false;
+ lWarning() << "Cannot create video RTP IO session";
+ }
+ } else {
+ io.input.type = MSResourceCamera;
+ io.input.camera = cam;
+ io.output.type = MSResourceDefault;
+ }
+ if (ok) {
+ AudioStream *as = getPeerAudioStream();
+ if (as) audio_stream_link_video(as, mStream);
+ video_stream_start_from_io(mStream, videoProfile, dest.rtpAddr.c_str(), dest.rtpPort, dest.rtcpAddr.c_str(), dest.rtcpPort,
+ usedPt, &io);
+ }
+ }
+ mStartCount++;
+
+ if (listener)
+ listener->onResetFirstVideoFrameDecoded(getMediaSession().getSharedFromThis());
+ /* Start ZRTP engine if needed : set here or remote have a zrtp-hash attribute */
+ const SalStreamDescription *remoteStream = ctx.remoteStreamDescription;
+ if ((getMediaSessionPrivate().getParams()->getMediaEncryption() == LinphoneMediaEncryptionZRTP) || (remoteStream->haveZrtpHash == 1)) {
+ Stream *audioStream = getGroup().lookupMainStream(SalAudio);
+ /* Audio stream is already encrypted and video stream is active */
+ if (audioStream && audioStream->isEncrypted()) {
+ activateZrtp();
+ if (remoteStream->haveZrtpHash == 1) {
+ int retval = ms_zrtp_setPeerHelloHash(mSessions.zrtp_context, (uint8_t *)remoteStream->zrtphash, strlen((const char *)(remoteStream->zrtphash)));
+ if (retval != 0)
+ lError() << "Video stream ZRTP hash mismatch 0x" << hex << retval;
+ }
+ }
+ }
+
+ if (linphone_core_retransmission_on_nack_enabled(getCCore())) {
+ video_stream_enable_retransmission_on_nack(mStream, TRUE);
+ }
+
+ if (!reusedPreview && source) {
+ /* Destroy not-reused source filter */
+ lWarning() << "Video preview (" << source << ") not reused: destroying it";
+ ms_filter_destroy(source);
+ }
+
+}
+
+void MS2VideoStream::stop(){
+ MS2Stream::stop();
+ AudioStream *as = getPeerAudioStream();
+ if (as) audio_stream_unlink_video(as, mStream);
+ video_stream_stop(mStream);
+ /* In mediastreamer2, stop actually stops and destroys. We immediately need to recreate the stream object for later use, keeping the
+ * sessions (for RTP, SRTP, ZRTP etc) that were setup at the beginning. */
+ mStream = video_stream_new_with_sessions(getCCore()->factory, &mSessions);
+ getMediaSessionPrivate().getCurrentParams()->getPrivate()->setUsedVideoCodec(nullptr);
+}
+
+void MS2VideoStream::handleEvent(const OrtpEvent *ev){
+ OrtpEventType evt = ortp_event_get_type(ev);
+ OrtpEventData *evd = ortp_event_get_data(const_cast<OrtpEvent*>(ev));
+
+ if (evt == ORTP_EVENT_NEW_VIDEO_BANDWIDTH_ESTIMATION_AVAILABLE) {
+ lInfo() << "Video bandwidth estimation is " << (int)(evd->info.video_bandwidth_available / 1000.) << " kbit/s";
+ if (isMain())
+ linphone_call_stats_set_estimated_download_bandwidth(mStats, (float)(evd->info.video_bandwidth_available*1e-3));
+ }
+}
+
+void MS2VideoStream::zrtpStarted(Stream *mainZrtpStream){
+ if (getState() == Running){
+ lInfo() << "Trying to start ZRTP encryption on video stream";
+ activateZrtp();
+ if (getMediaSessionPrivate().isEncryptionMandatory()) {
+ /* Nothing could have been sent yet so generating key frame */
+ video_stream_send_vfu(mStream);
+ }
+ }
+}
+
+void MS2VideoStream::tryEarlyMediaForking(const OfferAnswerContext &ctx){
+ MS2Stream::tryEarlyMediaForking(ctx);
+ sendVfu();
+}
+
+void MS2VideoStream::oglRender(){
+ if (mStream->output && (ms_filter_get_id(mStream->output) == MS_OGL_ID))
+ ms_filter_call_method(mStream->output, MS_OGL_RENDER, nullptr);
+}
+
+AudioStream *MS2VideoStream::getPeerAudioStream(){
+ MS2AudioStream *as = getGroup().lookupMainStreamInterface<MS2AudioStream>(SalAudio);
+ return as ? (AudioStream*)as->getMediaStream() : nullptr;
+}
+
+void MS2VideoStream::requestNotifyNextVideoFrameDecoded () {
+ if (mStream->ms.decoder)
+ ms_filter_call_method_noarg(mStream->ms.decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
+}
+
+
+void MS2VideoStream::snapshotTakenCb(void *userdata, struct _MSFilter *f, unsigned int id, void *arg) {
+ if (id == MS_JPEG_WRITER_SNAPSHOT_TAKEN) {
+ CallSessionListener *listener = getMediaSessionPrivate().getCallSessionListener();
+ const char *filepath = (const char *) arg;
+ listener->onSnapshotTaken(getMediaSession().getSharedFromThis(), filepath);
+ }
+}
+
+void MS2VideoStream::sSnapshotTakenCb(void *userdata, struct _MSFilter *f, unsigned int id, void *arg) {
+ MS2VideoStream *d = (MS2VideoStream *)userdata;
+ d->snapshotTakenCb(userdata, f, id, arg);
+}
+
+int MS2VideoStream::takePreviewSnapshot (const string& file) {
+ if (mStream && mStream->local_jpegwriter) {
+ ms_filter_clear_notify_callback(mStream->jpegwriter);
+ const char *filepath = file.empty() ? nullptr : file.c_str();
+ ms_filter_add_notify_callback(mStream->local_jpegwriter, sSnapshotTakenCb, this, TRUE);
+ return ms_filter_call_method(mStream->local_jpegwriter, MS_JPEG_WRITER_TAKE_SNAPSHOT, (void *)filepath);
+ }
+ lWarning() << "Cannot take local snapshot: no currently running video stream on this call";
+ return -1;
+}
+
+int MS2VideoStream::takeVideoSnapshot (const string& file) {
+ if (mStream && mStream->jpegwriter) {
+ ms_filter_clear_notify_callback(mStream->jpegwriter);
+ const char *filepath = file.empty() ? nullptr : file.c_str();
+ ms_filter_add_notify_callback(mStream->jpegwriter, sSnapshotTakenCb, this, TRUE);
+ return ms_filter_call_method(mStream->jpegwriter, MS_JPEG_WRITER_TAKE_SNAPSHOT, (void *)filepath);
+ }
+ lWarning() << "Cannot take snapshot: no currently running video stream on this call";
+ return -1;
+}
+
+bool MS2VideoStream::cameraEnabled() const{
+ return mCameraEnabled;
+}
+
+void MS2VideoStream::getRecvStats(VideoStats *s) const{
+ if (mStream){
+ s->fps = video_stream_get_received_framerate(mStream);
+ MSVideoSize vsize = video_stream_get_received_video_size(mStream);
+ s->width = vsize.width;
+ s->height = vsize.height;
+ }else{
+ s->fps = 0.0;
+ s->width = s->height = 0;
+ }
+}
+
+void MS2VideoStream::getSendStats(VideoStats *s) const{
+ if (mStream){
+ s->fps = video_stream_get_sent_framerate(mStream);
+ MSVideoSize vsize = video_stream_get_sent_video_size(mStream);
+ s->width = vsize.width;
+ s->height = vsize.height;
+ }else{
+ s->fps = 0.0;
+ s->width = s->height = 0;
+ }
+}
+
+void MS2VideoStream::finish(){
+ if (mStream) {
+ video_stream_stop(mStream);
+ mStream = nullptr;
+ }
+ MS2Stream::finish();
+}
+
+MS2VideoStream::~MS2VideoStream(){
+ if (mStream) video_stream_stop(mStream);
+}
+
+LINPHONE_END_NAMESPACE
+
+#endif
+
+
diff --git a/src/core/core-call.cpp b/src/core/core-call.cpp
index 65f705456468be0512248416941e40ae00a9792a..53447eb540fe3e51b4e07eee4fbc7f162b0625bf 100644
--- a/src/core/core-call.cpp
+++ b/src/core/core-call.cpp
@@ -23,6 +23,7 @@
#include "core-p.h"
#include "call/call-p.h"
#include "conference/session/call-session-p.h"
+#include "conference/session/media-session.h"
#include "logger/logger.h"
// TODO: Remove me later.
@@ -110,120 +111,21 @@ int CorePrivate::removeCall (const shared_ptr<Call> &call) {
return 0;
}
-void CorePrivate::unsetVideoWindowId (bool preview, void *id) {
+void CorePrivate::setVideoWindowId (bool preview, void *id) {
#ifdef VIDEO_ENABLED
for (const auto &call : calls) {
- VideoStream *vstream = reinterpret_cast<VideoStream *>(call->getPrivate()->getMediaStream(LinphoneStreamTypeVideo));
- if (vstream) {
- if (preview)
- video_stream_set_native_preview_window_id(vstream, id);
- else
- video_stream_set_native_window_id(vstream, id);
- }
- }
-#endif
-}
-
-// -----------------------------------------------------------------------------
-
-void CorePrivate::parameterizeEqualizer (AudioStream *stream) {
- L_Q();
- LinphoneConfig *config = linphone_core_get_config(q->getCCore());
- const char *eqActive = lp_config_get_string(config, "sound", "eq_active", nullptr);
- if (eqActive)
- lWarning() << "'eq_active' linphonerc parameter has no effect anymore. Please use 'mic_eq_active' or 'spk_eq_active' instead";
- const char *eqGains = lp_config_get_string(config, "sound", "eq_gains", nullptr);
- if(eqGains)
- lWarning() << "'eq_gains' linphonerc parameter has no effect anymore. Please use 'mic_eq_gains' or 'spk_eq_gains' instead";
- if (stream->mic_equalizer) {
- MSFilter *f = stream->mic_equalizer;
- bool enabled = !!lp_config_get_int(config, "sound", "mic_eq_active", 0);
- ms_filter_call_method(f, MS_EQUALIZER_SET_ACTIVE, &enabled);
- const char *gains = lp_config_get_string(config, "sound", "mic_eq_gains", nullptr);
- if (enabled && gains) {
- bctbx_list_t *gainsList = ms_parse_equalizer_string(gains);
- for (bctbx_list_t *it = gainsList; it; it = bctbx_list_next(it)) {
- MSEqualizerGain *g = reinterpret_cast<MSEqualizerGain *>(bctbx_list_get_data(it));
- lInfo() << "Read microphone equalizer gains: " << g->frequency << "(~" << g->width << ") --> " << g->gain;
- ms_filter_call_method(f, MS_EQUALIZER_SET_GAIN, g);
+ shared_ptr<MediaSession> ms = dynamic_pointer_cast<MediaSession>(call->getPrivate()->getActiveSession());
+ if (ms){
+ if (preview){
+ ms->setNativePreviewWindowId(id);
+ }else{
+ ms->setNativeVideoWindowId(id);
}
- if (gainsList)
- bctbx_list_free_with_data(gainsList, ms_free);
}
}
- if (stream->spk_equalizer) {
- MSFilter *f = stream->spk_equalizer;
- bool enabled = !!lp_config_get_int(config, "sound", "spk_eq_active", 0);
- ms_filter_call_method(f, MS_EQUALIZER_SET_ACTIVE, &enabled);
- const char *gains = lp_config_get_string(config, "sound", "spk_eq_gains", nullptr);
- if (enabled && gains) {
- bctbx_list_t *gainsList = ms_parse_equalizer_string(gains);
- for (bctbx_list_t *it = gainsList; it; it = bctbx_list_next(it)) {
- MSEqualizerGain *g = reinterpret_cast<MSEqualizerGain *>(bctbx_list_get_data(it));
- lInfo() << "Read speaker equalizer gains: " << g->frequency << "(~" << g->width << ") --> " << g->gain;
- ms_filter_call_method(f, MS_EQUALIZER_SET_GAIN, g);
- }
- if (gainsList)
- bctbx_list_free_with_data(gainsList, ms_free);
- }
- }
-}
-
-void CorePrivate::postConfigureAudioStream (AudioStream *stream, bool muted) {
- L_Q();
- float micGain = q->getCCore()->sound_conf.soft_mic_lev;
- if (muted)
- audio_stream_set_mic_gain(stream, 0);
- else
- audio_stream_set_mic_gain_db(stream, micGain);
- float recvGain = q->getCCore()->sound_conf.soft_play_lev;
- if (static_cast<int>(recvGain))
- setPlaybackGainDb(stream, recvGain);
- LinphoneConfig *config = linphone_core_get_config(q->getCCore());
- float ngThres = lp_config_get_float(config, "sound", "ng_thres", 0.05f);
- float ngFloorGain = lp_config_get_float(config, "sound", "ng_floorgain", 0);
- if (stream->volsend) {
- int dcRemoval = lp_config_get_int(config, "sound", "dc_removal", 0);
- ms_filter_call_method(stream->volsend, MS_VOLUME_REMOVE_DC, &dcRemoval);
- float speed = lp_config_get_float(config, "sound", "el_speed", -1);
- float thres = lp_config_get_float(config, "sound", "el_thres", -1);
- float force = lp_config_get_float(config, "sound", "el_force", -1);
- int sustain = lp_config_get_int(config, "sound", "el_sustain", -1);
- float transmitThres = lp_config_get_float(config, "sound", "el_transmit_thres", -1);
- if (static_cast<int>(speed) == -1)
- speed = 0.03f;
- if (static_cast<int>(force) == -1)
- force = 25;
- MSFilter *f = stream->volsend;
- ms_filter_call_method(f, MS_VOLUME_SET_EA_SPEED, &speed);
- ms_filter_call_method(f, MS_VOLUME_SET_EA_FORCE, &force);
- if (static_cast<int>(thres) != -1)
- ms_filter_call_method(f, MS_VOLUME_SET_EA_THRESHOLD, &thres);
- if (static_cast<int>(sustain) != -1)
- ms_filter_call_method(f, MS_VOLUME_SET_EA_SUSTAIN, &sustain);
- if (static_cast<int>(transmitThres) != -1)
- ms_filter_call_method(f, MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD, &transmitThres);
- ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_THRESHOLD, &ngThres);
- ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_FLOORGAIN, &ngFloorGain);
- }
- if (stream->volrecv) {
- /* Parameters for a limited noise-gate effect, using echo limiter threshold */
- float floorGain = (float)(1 / pow(10, micGain / 10));
- int spkAgc = lp_config_get_int(config, "sound", "speaker_agc_enabled", 0);
- MSFilter *f = stream->volrecv;
- ms_filter_call_method(f, MS_VOLUME_ENABLE_AGC, &spkAgc);
- ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_THRESHOLD, &ngThres);
- ms_filter_call_method(f, MS_VOLUME_SET_NOISE_GATE_FLOORGAIN, &floorGain);
- }
- parameterizeEqualizer(stream);
+#endif
}
-void CorePrivate::setPlaybackGainDb (AudioStream *stream, float gain) {
- if (stream->volrecv)
- ms_filter_call_method(stream->volrecv, MS_VOLUME_SET_DB_GAIN, &gain);
- else
- lWarning() << "Could not apply playback gain: gain control wasn't activated";
-}
// =============================================================================
diff --git a/src/core/core-p.h b/src/core/core-p.h
index e544c863d34d904d26b1a86b6bccfa6363b891e5..3e8bd67b3cff99204faba8c61940b6ca3f091b2b 100644
--- a/src/core/core-p.h
+++ b/src/core/core-p.h
@@ -73,11 +73,7 @@ public:
void notifySoundcardUsage (bool used);
int removeCall (const std::shared_ptr<Call> &call);
void setCurrentCall (const std::shared_ptr<Call> &call) { currentCall = call; }
- void unsetVideoWindowId (bool preview, void *id);
-
- void parameterizeEqualizer (AudioStream *stream);
- void postConfigureAudioStream (AudioStream *stream, bool muted);
- void setPlaybackGainDb (AudioStream *stream, float gain);
+ void setVideoWindowId (bool preview, void *id);
void loadChatRooms ();
void handleEphemeralMessages (time_t currentTime);
diff --git a/src/core/core.cpp b/src/core/core.cpp
index 29c5f97bd849cc9663a60b1cbb957c4d86c7e673..e5b056f0eaf868a3c20daf9adab8509a433cc49d 100644
--- a/src/core/core.cpp
+++ b/src/core/core.cpp
@@ -656,9 +656,10 @@ void Core::doLater(const std::function<void ()> &something){
getPrivate()->doLater(something);
}
-belle_sip_source_t *Core::createTimer(const std::function<bool ()> &something, unsigned int milliseconds){
- return belle_sip_main_loop_create_cpp_timeout_2(getPrivate()->getMainLoop(), something, milliseconds, "");
+belle_sip_source_t *Core::createTimer(const std::function<bool ()> &something, int milliseconds, const string &name){
+ return belle_sip_main_loop_create_cpp_timeout_2(getPrivate()->getMainLoop(), something, (unsigned)milliseconds, name.c_str());
}
+
/* Stop and destroy a timer created by createTimer()*/
void Core::destroyTimer(belle_sip_source_t *timer){
belle_sip_main_loop_remove_source(getPrivate()->getMainLoop(), timer);
diff --git a/src/core/core.h b/src/core/core.h
index 4274e305f7d88d8c202dddf02bec65f7bf16e966..9ec89dd8165d072852ff5e946fee7c52c425690c 100644
--- a/src/core/core.h
+++ b/src/core/core.h
@@ -192,10 +192,12 @@ public:
/*
* Run supplied std::function as a timer. It should return true if repeated, false otherwise.
- * It may be unrefed with (with belle_sip_object_unref()) before expiration, if this timer never needs to be cancelled.
+ * The returned belle_sip_source_t must be unrefed (with belle_sip_object_unref() ).
+ * It may be unrefed before expiration, if this timer never needs to be cancelled.
*/
- belle_sip_source_t *createTimer(const std::function<bool ()> &something, unsigned int milliseconds);
- /* Stop (ie cancel) and destroy a timer created by createTimer() */
+ belle_sip_source_t *createTimer(const std::function<bool ()> &something, int millisecond, const std::string &name);
+ /* Stop (ie cancel) and destroy a timer created by createTimer()*/
+
void destroyTimer(belle_sip_source_t *timer);
private:
Core ();
diff --git a/src/nat/ice-agent.cpp b/src/nat/ice-agent.cpp
deleted file mode 100644
index 9b69310a832304f291d5d2be80ccddd9d012a32c..0000000000000000000000000000000000000000
--- a/src/nat/ice-agent.cpp
+++ /dev/null
@@ -1,768 +0,0 @@
-/*
- * Copyright (c) 2010-2019 Belledonne Communications SARL.
- *
- * This file is part of Liblinphone.
- *
- * This program is free software: you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include "linphone/core.h"
-
-#include "private.h"
-
-#include "conference/session/media-session-p.h"
-#include "core/core.h"
-#include "logger/logger.h"
-
-#include "ice-agent.h"
-
-// =============================================================================
-
-using namespace std;
-
-LINPHONE_BEGIN_NAMESPACE
-
-bool IceAgent::candidatesGathered () const {
- if (!iceSession)
- return false;
- return !!ice_session_candidates_gathered(iceSession);
-}
-
-void IceAgent::checkSession (IceRole role, bool isReinvite) {
- // Already created.
- if (iceSession)
- return;
-
- LinphoneConfig *config = linphone_core_get_config(mediaSession.getCore()->getCCore());
-
- if (lp_config_get_int(config, "net", "force_ice_disablement", 0)){
- lWarning()<<"ICE is disabled in this version";
- return;
- }
-
- if (isReinvite && (lp_config_get_int(config, "net", "allow_late_ice", 0) == 0))
- return;
-
- iceSession = ice_session_new();
-
- // For backward compatibility purposes, shall be enabled by default in the future.
- ice_session_enable_message_integrity_check(
- iceSession,
- !!lp_config_get_int(config, "net", "ice_session_enable_message_integrity_check", 1)
- );
- if (lp_config_get_int(config, "net", "dont_default_to_stun_candidates", 0)) {
- IceCandidateType types[ICT_CandidateTypeMax];
- types[0] = ICT_HostCandidate;
- types[1] = ICT_RelayedCandidate;
- types[2] = ICT_CandidateInvalid;
- ice_session_set_default_candidates_types(iceSession, types);
- }
- ice_session_set_role(iceSession, role);
-}
-
-void IceAgent::deleteSession () {
- if (!iceSession)
- return;
-
- ice_session_destroy(iceSession);
- iceSession = nullptr;
- mediaSession.getPrivate()->deactivateIce();
-}
-
-void IceAgent::gatheringFinished () {
- const SalMediaDescription *rmd = mediaSession.getPrivate()->getOp()->getRemoteMediaDescription();
- if (rmd)
- clearUnusedIceCandidates(mediaSession.getPrivate()->getLocalDesc(), rmd);
- if (!iceSession)
- return;
-
- ice_session_compute_candidates_foundations(iceSession);
- ice_session_eliminate_redundant_candidates(iceSession);
- ice_session_choose_default_candidates(iceSession);
-
- int pingTime = ice_session_average_gathering_round_trip_time(iceSession);
- if (pingTime >= 0) {
- mediaSession.getPrivate()->setPingTime(pingTime);
- }
-}
-
-int IceAgent::getNbLosingPairs () const {
- if (!iceSession)
- return 0;
- return ice_session_nb_losing_pairs(iceSession);
-}
-
-bool IceAgent::hasCompleted () const {
- if (!iceSession)
- return true;
- return ice_session_state(iceSession) == IS_Completed;
-}
-
-bool IceAgent::hasCompletedCheckList () const {
- if (!iceSession)
- return false;
- switch (ice_session_state(iceSession)) {
- case IS_Completed:
- case IS_Failed:
- return !!ice_session_has_completed_check_list(iceSession);
- default:
- return false;
- }
-}
-
-bool IceAgent::isControlling () const {
- if (!iceSession)
- return false;
- return ice_session_role(iceSession) == IR_Controlling;
-}
-
-bool IceAgent::prepare (const SalMediaDescription *localDesc, bool incomingOffer, bool allowGathering) {
- if (!iceSession)
- return false;
-
- SalMediaDescription *remoteDesc = nullptr;
- bool hasVideo = false;
- if (incomingOffer) {
- remoteDesc = mediaSession.getPrivate()->getOp()->getRemoteMediaDescription();
- hasVideo = linphone_core_video_enabled(mediaSession.getCore()->getCCore()) &&
- linphone_core_media_description_contains_video_stream(remoteDesc);
- } else
- hasVideo = mediaSession.getMediaParams()->videoEnabled();
-
- prepareIceForStream(mediaSession.getPrivate()->getMediaStream(LinphoneStreamTypeAudio), true);
- if (hasVideo)
- prepareIceForStream(mediaSession.getPrivate()->getMediaStream(LinphoneStreamTypeVideo), true);
- if (mediaSession.getMediaParams()->realtimeTextEnabled())
- prepareIceForStream(mediaSession.getPrivate()->getMediaStream(LinphoneStreamTypeText), true);
-
- // Start ICE gathering.
- if (incomingOffer){
- // This may delete the ice session.
- updateFromRemoteMediaDescription(localDesc, remoteDesc, true);
- }
- if (iceSession && allowGathering && !ice_session_candidates_gathered(iceSession)) {
- mediaSession.getPrivate()->prepareStreamsForIceGathering(hasVideo);
- int err = gatherIceCandidates();
- if (err == 0) {
- // Ice candidates gathering wasn't started, but we can proceed with the call anyway.
- mediaSession.getPrivate()->stopStreamsForIceGathering();
- return false;
- } else if (err == -1) {
- mediaSession.getPrivate()->stopStreamsForIceGathering();
- deleteSession();
- return false;
- }
- return true;
- }
- return false;
-}
-
-void IceAgent::prepareIceForStream (MediaStream *ms, bool createChecklist) {
- if (!iceSession)
- return;
-
- int streamIndex = mediaSession.getPrivate()->getStreamIndex(ms);
- rtp_session_set_pktinfo(ms->sessions.rtp_session, true);
- IceCheckList *cl = ice_session_check_list(iceSession, streamIndex);
- if (!cl && createChecklist) {
- cl = ice_check_list_new();
- ice_session_add_check_list(iceSession, cl, static_cast<unsigned int>(streamIndex));
- lInfo() << "Created new ICE check list for stream [" << streamIndex << "]";
- }
- if (cl)
- media_stream_set_ice_check_list(ms, cl);
-}
-
-void IceAgent::resetSession (IceRole role) {
- if (!iceSession)
- return;
- ice_session_reset(iceSession, role);
-}
-
-void IceAgent::restartSession (IceRole role) {
- if (!iceSession)
- return;
- ice_session_restart(iceSession, role);
-}
-
-void IceAgent::startConnectivityChecks () {
- if (!iceSession)
- return;
- ice_session_start_connectivity_checks(iceSession);
-}
-
-void IceAgent::stopIceForInactiveStreams (SalMediaDescription *desc) {
- if (!iceSession)
- return;
- if (ice_session_state(iceSession) == IS_Completed)
- return;
- for (int i = 0; i < desc->nb_streams; i++) {
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (!sal_stream_description_active(&desc->streams[i]) && cl) {
- ice_session_remove_check_list(iceSession, cl);
- mediaSession.getPrivate()->clearIceCheckList(cl);
- }
- }
- updateIceStateInCallStats();
-}
-
-void IceAgent::updateFromRemoteMediaDescription (
- const SalMediaDescription *localDesc,
- const SalMediaDescription *remoteDesc,
- bool isOffer
-) {
- if (!iceSession)
- return;
-
- if (!iceParamsFoundInRemoteMediaDescription(remoteDesc)) {
- // Response from remote does not contain mandatory ICE attributes, delete the session.
- deleteSession();
- mediaSession.getPrivate()->enableSymmetricRtp(!!linphone_core_symmetric_rtp_enabled(mediaSession.getCore()->getCCore()));
- return;
- }
-
- // Check for ICE restart and set remote credentials.
- bool iceRestarted = checkForIceRestartAndSetRemoteCredentials(remoteDesc, isOffer);
-
- // Create ICE check lists if needed and parse ICE attributes.
- createIceCheckListsAndParseIceAttributes(remoteDesc, iceRestarted);
- for (int i = 0; i < remoteDesc->nb_streams; i++) {
- const SalStreamDescription *stream = &remoteDesc->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (!cl) continue;
- if (!sal_stream_description_active(stream)) {
- ice_session_remove_check_list_from_idx(iceSession, static_cast<unsigned int>(i));
- mediaSession.getPrivate()->clearIceCheckList(cl);
- }
- }
- clearUnusedIceCandidates(localDesc, remoteDesc);
- ice_session_check_mismatch(iceSession);
-
- if (ice_session_nb_check_lists(iceSession) == 0) {
- deleteSession();
- mediaSession.getPrivate()->enableSymmetricRtp(!!linphone_core_symmetric_rtp_enabled(mediaSession.getCore()->getCCore()));
- }
-}
-
-void IceAgent::updateIceStateInCallStats () {
- if (!iceSession)
- return;
- IceCheckList *audioCheckList = ice_session_check_list(iceSession, mediaSession.getPrivate()->getStreamIndex(LinphoneStreamTypeAudio));
- IceCheckList *videoCheckList = ice_session_check_list(iceSession, mediaSession.getPrivate()->getStreamIndex(LinphoneStreamTypeVideo));
- IceCheckList *textCheckList = ice_session_check_list(iceSession, mediaSession.getPrivate()->getStreamIndex(LinphoneStreamTypeText));
- if (!audioCheckList && !videoCheckList && !textCheckList)
- return;
-
- LinphoneCallStats *audioStats = mediaSession.getPrivate()->getStats(LinphoneStreamTypeAudio);
- LinphoneCallStats *videoStats = mediaSession.getPrivate()->getStats(LinphoneStreamTypeVideo);
- LinphoneCallStats *textStats = mediaSession.getPrivate()->getStats(LinphoneStreamTypeText);
- IceSessionState sessionState = ice_session_state(iceSession);
- if ((sessionState == IS_Completed) || ((sessionState == IS_Failed) && ice_session_has_completed_check_list(iceSession))) {
- _linphone_call_stats_set_ice_state(audioStats, LinphoneIceStateNotActivated);
- if (audioCheckList && mediaSession.getMediaParams()->audioEnabled())
- updateIceStateInCallStatsForStream(audioStats, audioCheckList);
-
- _linphone_call_stats_set_ice_state(videoStats, LinphoneIceStateNotActivated);
- if (videoCheckList && mediaSession.getMediaParams()->videoEnabled())
- updateIceStateInCallStatsForStream(videoStats, videoCheckList);
-
- _linphone_call_stats_set_ice_state(textStats, LinphoneIceStateNotActivated);
- if (textCheckList && mediaSession.getMediaParams()->realtimeTextEnabled())
- updateIceStateInCallStatsForStream(textStats, textCheckList);
- } else if (sessionState == IS_Running) {
- if (audioCheckList && mediaSession.getMediaParams()->audioEnabled())
- _linphone_call_stats_set_ice_state(audioStats, LinphoneIceStateInProgress);
- if (videoCheckList && mediaSession.getMediaParams()->videoEnabled())
- _linphone_call_stats_set_ice_state(videoStats, LinphoneIceStateInProgress);
- if (textCheckList && mediaSession.getMediaParams()->realtimeTextEnabled())
- _linphone_call_stats_set_ice_state(textStats, LinphoneIceStateInProgress);
- } else {
- if (audioCheckList && mediaSession.getMediaParams()->audioEnabled())
- _linphone_call_stats_set_ice_state(audioStats, LinphoneIceStateFailed);
- if (videoCheckList && mediaSession.getMediaParams()->videoEnabled())
- _linphone_call_stats_set_ice_state(videoStats, LinphoneIceStateFailed);
- if (textCheckList && mediaSession.getMediaParams()->realtimeTextEnabled())
- _linphone_call_stats_set_ice_state(textStats, LinphoneIceStateFailed);
- }
- lInfo() << "CallSession [" << &mediaSession << "] New ICE state: audio: [" << linphone_ice_state_to_string(linphone_call_stats_get_ice_state(audioStats)) <<
- "] video: [" << linphone_ice_state_to_string(linphone_call_stats_get_ice_state(videoStats)) <<
- "] text: [" << linphone_ice_state_to_string(linphone_call_stats_get_ice_state(textStats)) << "]";
-}
-
-void IceAgent::updateLocalMediaDescriptionFromIce (SalMediaDescription *desc) {
- if (!iceSession)
- return;
- IceCandidate *rtpCandidate = nullptr;
- IceCandidate *rtcpCandidate = nullptr;
- bool result = false;
- IceSessionState sessionState = ice_session_state(iceSession);
- if (sessionState == IS_Completed) {
- IceCheckList *firstCl = nullptr;
- for (int i = 0; i < desc->nb_streams; i++) {
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (cl) {
- firstCl = cl;
- break;
- }
- }
- if (firstCl)
- result = !!ice_check_list_selected_valid_local_candidate(firstCl, &rtpCandidate, nullptr);
- if (result) {
- strncpy(desc->addr, rtpCandidate->taddr.ip, sizeof(desc->addr));
- } else {
- lWarning() << "If ICE has completed successfully, rtp_candidate should be set!";
- ice_dump_valid_list(firstCl);
- }
- }
-
- strncpy(desc->ice_pwd, ice_session_local_pwd(iceSession), sizeof(desc->ice_pwd));
- desc->ice_pwd[sizeof(desc->ice_pwd) - 1] = '\0';
- strncpy(desc->ice_ufrag, ice_session_local_ufrag(iceSession), sizeof(desc->ice_ufrag));
- desc->ice_ufrag[sizeof(desc->ice_ufrag) - 1] = '\0';
-
- for (int i = 0; i < desc->nb_streams; i++) {
- SalStreamDescription *stream = &desc->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- rtpCandidate = rtcpCandidate = nullptr;
- if (!sal_stream_description_active(stream) || !cl)
- continue;
- if (ice_check_list_state(cl) == ICL_Completed) {
- LinphoneConfig *config = linphone_core_get_config(mediaSession.getCore()->getCCore());
- // TODO: Remove `ice_uses_nortpproxy` option, let's say in December 2018.
- bool useNoRtpProxy = !!lp_config_get_int(config, "sip", "ice_uses_nortpproxy", false);
- if (useNoRtpProxy)
- stream->set_nortpproxy = true;
- result = !!ice_check_list_selected_valid_local_candidate(ice_session_check_list(iceSession, i), &rtpCandidate, &rtcpCandidate);
- } else {
- stream->set_nortpproxy = false;
- result = !!ice_check_list_default_local_candidate(ice_session_check_list(iceSession, i), &rtpCandidate, &rtcpCandidate);
- }
- if (result) {
- strncpy(stream->rtp_addr, rtpCandidate->taddr.ip, sizeof(stream->rtp_addr));
- strncpy(stream->rtcp_addr, rtcpCandidate->taddr.ip, sizeof(stream->rtcp_addr));
- stream->rtp_port = rtpCandidate->taddr.port;
- stream->rtcp_port = rtcpCandidate->taddr.port;
- } else {
- memset(stream->rtp_addr, 0, sizeof(stream->rtp_addr));
- memset(stream->rtcp_addr, 0, sizeof(stream->rtcp_addr));
- }
- if ((strlen(ice_check_list_local_pwd(cl)) != strlen(desc->ice_pwd)) || (strcmp(ice_check_list_local_pwd(cl), desc->ice_pwd)))
- strncpy(stream->ice_pwd, ice_check_list_local_pwd(cl), sizeof(stream->ice_pwd));
- else
- memset(stream->ice_pwd, 0, sizeof(stream->ice_pwd));
- if ((strlen(ice_check_list_local_ufrag(cl)) != strlen(desc->ice_ufrag)) || (strcmp(ice_check_list_local_ufrag(cl), desc->ice_ufrag)))
- strncpy(stream->ice_ufrag, ice_check_list_local_ufrag(cl), sizeof(stream->ice_ufrag));
- else
- memset(stream->ice_pwd, 0, sizeof(stream->ice_pwd));
- stream->ice_mismatch = ice_check_list_is_mismatch(cl);
- if ((ice_check_list_state(cl) == ICL_Running) || (ice_check_list_state(cl) == ICL_Completed)) {
- memset(stream->ice_candidates, 0, sizeof(stream->ice_candidates));
- int nbCandidates = 0;
- for (int j = 0; j < MIN((int)bctbx_list_size(cl->local_candidates), SAL_MEDIA_DESCRIPTION_MAX_ICE_CANDIDATES); j++) {
- SalIceCandidate *salCandidate = &stream->ice_candidates[nbCandidates];
- IceCandidate *iceCandidate = reinterpret_cast<IceCandidate *>(bctbx_list_nth_data(cl->local_candidates, j));
- const char *defaultAddr = nullptr;
- int defaultPort = 0;
- if (iceCandidate->componentID == 1) {
- defaultAddr = stream->rtp_addr;
- defaultPort = stream->rtp_port;
- } else if (iceCandidate->componentID == 2) {
- defaultAddr = stream->rtcp_addr;
- defaultPort = stream->rtcp_port;
- } else
- continue;
- if (defaultAddr[0] == '\0')
- defaultAddr = desc->addr;
- // Only include the candidates matching the default destination for each component of the stream if the state is Completed as specified in RFC5245 section 9.1.2.2.
- if (
- ice_check_list_state(cl) == ICL_Completed &&
- !((iceCandidate->taddr.port == defaultPort) && (strlen(iceCandidate->taddr.ip) == strlen(defaultAddr)) && (strcmp(iceCandidate->taddr.ip, defaultAddr) == 0))
- )
- continue;
- strncpy(salCandidate->foundation, iceCandidate->foundation, sizeof(salCandidate->foundation));
- salCandidate->componentID = iceCandidate->componentID;
- salCandidate->priority = iceCandidate->priority;
- strncpy(salCandidate->type, ice_candidate_type(iceCandidate), sizeof(salCandidate->type));
- strncpy(salCandidate->addr, iceCandidate->taddr.ip, sizeof(salCandidate->addr));
- salCandidate->port = iceCandidate->taddr.port;
- if (iceCandidate->base && (iceCandidate->base != iceCandidate)) {
- strncpy(salCandidate->raddr, iceCandidate->base->taddr.ip, sizeof(salCandidate->raddr));
- salCandidate->rport = iceCandidate->base->taddr.port;
- }
- nbCandidates++;
- }
- }
- if ((ice_check_list_state(cl) == ICL_Completed) && (ice_session_role(iceSession) == IR_Controlling)) {
- memset(stream->ice_remote_candidates, 0, sizeof(stream->ice_remote_candidates));
- if (ice_check_list_selected_valid_remote_candidate(cl, &rtpCandidate, &rtcpCandidate)) {
- strncpy(stream->ice_remote_candidates[0].addr, rtpCandidate->taddr.ip, sizeof(stream->ice_remote_candidates[0].addr));
- stream->ice_remote_candidates[0].port = rtpCandidate->taddr.port;
- strncpy(stream->ice_remote_candidates[1].addr, rtcpCandidate->taddr.ip, sizeof(stream->ice_remote_candidates[1].addr));
- stream->ice_remote_candidates[1].port = rtcpCandidate->taddr.port;
- } else
- lError() << "ice: Selected valid remote candidates should be present if the check list is in the Completed state";
- } else {
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES; j++) {
- stream->ice_remote_candidates[j].addr[0] = '\0';
- stream->ice_remote_candidates[j].port = 0;
- }
- }
- }
-}
-
-// -----------------------------------------------------------------------------
-
-void IceAgent::addLocalIceCandidates (int family, const char *addr, IceCheckList *audioCl, IceCheckList *videoCl, IceCheckList *textCl) {
- if ((ice_check_list_state(audioCl) != ICL_Completed) && !ice_check_list_candidates_gathered(audioCl)) {
- int rtpPort = mediaSession.getPrivate()->getRtpPort(LinphoneStreamTypeAudio);
- int rtcpPort = mediaSession.getPrivate()->getRtcpPort(LinphoneStreamTypeAudio);
- ice_add_local_candidate(audioCl, "host", family, addr, rtpPort, 1, nullptr);
- ice_add_local_candidate(audioCl, "host", family, addr, rtcpPort, 2, nullptr);
- LinphoneCallStats *audioStats = mediaSession.getPrivate()->getStats(LinphoneStreamTypeAudio);
- _linphone_call_stats_set_ice_state(audioStats, LinphoneIceStateInProgress);
- }
- LinphoneCore *core = mediaSession.getCore()->getCCore();
- if (linphone_core_video_enabled(core) && videoCl && (ice_check_list_state(videoCl) != ICL_Completed) && !ice_check_list_candidates_gathered(videoCl)) {
- int rtpPort = mediaSession.getPrivate()->getRtpPort(LinphoneStreamTypeVideo);
- int rtcpPort = mediaSession.getPrivate()->getRtcpPort(LinphoneStreamTypeVideo);
- ice_add_local_candidate(videoCl, "host", family, addr, rtpPort, 1, nullptr);
- ice_add_local_candidate(videoCl, "host", family, addr, rtcpPort, 2, nullptr);
- LinphoneCallStats *videoStats = mediaSession.getPrivate()->getStats(LinphoneStreamTypeVideo);
- _linphone_call_stats_set_ice_state(videoStats, LinphoneIceStateInProgress);
- }
- if (mediaSession.getMediaParams()->realtimeTextEnabled() && textCl && (ice_check_list_state(textCl) != ICL_Completed) && !ice_check_list_candidates_gathered(textCl)) {
- int rtpPort = mediaSession.getPrivate()->getRtpPort(LinphoneStreamTypeText);
- int rtcpPort = mediaSession.getPrivate()->getRtcpPort(LinphoneStreamTypeText);
- ice_add_local_candidate(textCl, "host", family, addr, rtpPort, 1, nullptr);
- ice_add_local_candidate(textCl, "host", family, addr, rtcpPort, 2, nullptr);
- LinphoneCallStats *textStats = mediaSession.getPrivate()->getStats(LinphoneStreamTypeText);
- _linphone_call_stats_set_ice_state(textStats, LinphoneIceStateInProgress);
- }
-}
-
-bool IceAgent::checkForIceRestartAndSetRemoteCredentials (const SalMediaDescription *md, bool isOffer) {
- bool iceRestarted = false;
- string addr = md->addr;
- if ((addr == "0.0.0.0") || (addr == "::0")) {
- ice_session_restart(iceSession, isOffer ? IR_Controlled : IR_Controlling);
- iceRestarted = true;
- } else {
- for (int i = 0; i < md->nb_streams; i++) {
- const SalStreamDescription *stream = &md->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- string rtpAddr = stream->rtp_addr;
- if (cl && (rtpAddr == "0.0.0.0")) {
- ice_session_restart(iceSession, isOffer ? IR_Controlled : IR_Controlling);
- iceRestarted = true;
- break;
- }
- }
- }
- if (!ice_session_remote_ufrag(iceSession) && !ice_session_remote_pwd(iceSession)) {
- ice_session_set_remote_credentials(iceSession, md->ice_ufrag, md->ice_pwd);
- } else if (ice_session_remote_credentials_changed(iceSession, md->ice_ufrag, md->ice_pwd)) {
- if (!iceRestarted) {
- ice_session_restart(iceSession, isOffer ? IR_Controlled : IR_Controlling);
- iceRestarted = true;
- }
- ice_session_set_remote_credentials(iceSession, md->ice_ufrag, md->ice_pwd);
- }
- for (int i = 0; i < md->nb_streams; i++) {
- const SalStreamDescription *stream = &md->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (cl && (stream->ice_pwd[0] != '\0') && (stream->ice_ufrag[0] != '\0')) {
- if (ice_check_list_remote_credentials_changed(cl, stream->ice_ufrag, stream->ice_pwd)) {
- if (!iceRestarted && ice_check_list_get_remote_ufrag(cl) && ice_check_list_get_remote_pwd(cl)) {
- // Restart only if remote ufrag/paswd was already set.
- ice_session_restart(iceSession, isOffer ? IR_Controlled : IR_Controlling);
- iceRestarted = true;
- }
- ice_check_list_set_remote_credentials(cl, stream->ice_ufrag, stream->ice_pwd);
- }
- }
- }
- return iceRestarted;
-}
-
-void IceAgent::clearUnusedIceCandidates (const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc) {
- if (!localDesc)
- return;
- for (int i = 0; i < remoteDesc->nb_streams; i++) {
- const SalStreamDescription *localStream = &localDesc->streams[i];
- const SalStreamDescription *stream = &remoteDesc->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (!cl || !localStream)
- continue;
- if (stream->rtcp_mux && localStream->rtcp_mux) {
- ice_check_list_remove_rtcp_candidates(cl);
- }
- }
-}
-
-void IceAgent::createIceCheckListsAndParseIceAttributes (const SalMediaDescription *md, bool iceRestarted) {
- for (int i = 0; i < md->nb_streams; i++) {
- const SalStreamDescription *stream = &md->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (!cl)
- continue;
- if (stream->ice_mismatch) {
- ice_check_list_set_state(cl, ICL_Failed);
- continue;
- }
- if (stream->rtp_port == 0) {
- ice_session_remove_check_list(iceSession, cl);
- mediaSession.getPrivate()->clearIceCheckList(cl);
- continue;
- }
- if ((stream->ice_pwd[0] != '\0') && (stream->ice_ufrag[0] != '\0'))
- ice_check_list_set_remote_credentials(cl, stream->ice_ufrag, stream->ice_pwd);
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_CANDIDATES; j++) {
- bool defaultCandidate = false;
- const SalIceCandidate *candidate = &stream->ice_candidates[j];
- if (candidate->addr[0] == '\0')
- break;
- if ((candidate->componentID == 0) || (candidate->componentID > 2))
- continue;
- const char *addr = nullptr;
- int port = 0;
- getIceDefaultAddrAndPort(static_cast<uint16_t>(candidate->componentID), md, stream, &addr, &port);
- if (addr && (candidate->port == port) && (strlen(candidate->addr) == strlen(addr)) && (strcmp(candidate->addr, addr) == 0))
- defaultCandidate = true;
- int family = AF_INET;
- if (strchr(candidate->addr, ':'))
- family = AF_INET6;
- ice_add_remote_candidate(
- cl, candidate->type, family, candidate->addr, candidate->port,
- static_cast<uint16_t>(candidate->componentID),
- candidate->priority, candidate->foundation, defaultCandidate
- );
- }
- if (!iceRestarted) {
- bool losingPairsAdded = false;
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES; j++) {
- const SalIceRemoteCandidate *remoteCandidate = &stream->ice_remote_candidates[j];
- const char *addr = nullptr;
- int port = 0;
- int componentID = j + 1;
- if (remoteCandidate->addr[0] == '\0') break;
- getIceDefaultAddrAndPort(static_cast<uint16_t>(componentID), md, stream, &addr, &port);
-
- // If we receive a re-invite with remote-candidates, supply these pairs to the ice check list.
- // They might be valid pairs already selected, or losing pairs.
-
- int remoteFamily = AF_INET;
- if (strchr(remoteCandidate->addr, ':'))
- remoteFamily = AF_INET6;
- int family = AF_INET;
- if (strchr(addr, ':'))
- family = AF_INET6;
- ice_add_losing_pair(cl, static_cast<uint16_t>(j + 1), remoteFamily, remoteCandidate->addr, remoteCandidate->port, family, addr, port);
- losingPairsAdded = true;
- }
- if (losingPairsAdded)
- ice_check_list_check_completed(cl);
- }
- }
-}
-
-/** Return values:
- * 1: STUN gathering is started
- * 0: no STUN gathering is started, but it's ok to proceed with ICE anyway (with local candidates only or because STUN gathering was already done before)
- * -1: no gathering started and something went wrong with local candidates. There is no way to start the ICE session.
- */
-int IceAgent::gatherIceCandidates () {
- if (!iceSession)
- return -1;
- IceCheckList *audioCl = ice_session_check_list(iceSession, mediaSession.getPrivate()->getStreamIndex(LinphoneStreamTypeAudio));
- IceCheckList *videoCl = ice_session_check_list(iceSession, mediaSession.getPrivate()->getStreamIndex(LinphoneStreamTypeVideo));
- IceCheckList *textCl = ice_session_check_list(iceSession, mediaSession.getPrivate()->getStreamIndex(LinphoneStreamTypeText));
- if (!audioCl && !videoCl && !textCl)
- return -1;
-
- const struct addrinfo *ai = nullptr;
- LinphoneNatPolicy *natPolicy = mediaSession.getPrivate()->getNatPolicy();
- if (natPolicy && linphone_nat_policy_stun_server_activated(natPolicy)) {
- ai = linphone_nat_policy_get_stun_server_addrinfo(natPolicy);
- if (ai)
- ai = getIcePreferredStunServerAddrinfo(ai);
- else
- lWarning() << "Failed to resolve STUN server for ICE gathering, continuing without STUN";
- } else
- lWarning() << "ICE is used without STUN server";
- LinphoneCore *core = mediaSession.getCore()->getCCore();
- ice_session_enable_forced_relay(iceSession, core->forced_ice_relay);
- ice_session_enable_short_turn_refresh(iceSession, core->short_turn_refresh);
-
- // Gather local host candidates.
- char localAddr[LINPHONE_IPADDR_SIZE];
- if (mediaSession.getPrivate()->getAf() == AF_INET6) {
- if (linphone_core_get_local_ip_for(AF_INET6, nullptr, localAddr) < 0) {
- lError() << "Fail to get local IPv6";
- } else
- addLocalIceCandidates(AF_INET6, localAddr, audioCl, videoCl, textCl);
- }
- if (linphone_core_get_local_ip_for(AF_INET, nullptr, localAddr) < 0) {
- if (mediaSession.getPrivate()->getAf() != AF_INET6) {
- lError() << "Fail to get local IPv4";
- return -1;
- }
- } else
- addLocalIceCandidates(AF_INET, localAddr, audioCl, videoCl, textCl);
- if (ai && natPolicy && linphone_nat_policy_stun_server_activated(natPolicy)) {
- string server = linphone_nat_policy_get_stun_server(natPolicy);
- lInfo() << "ICE: gathering candidates from [" << server << "] using " << (linphone_nat_policy_turn_enabled(natPolicy) ? "TURN" : "STUN");
- // Gather local srflx candidates.
- ice_session_enable_turn(iceSession, linphone_nat_policy_turn_enabled(natPolicy));
- ice_session_set_stun_auth_requested_cb(iceSession, MediaSessionPrivate::stunAuthRequestedCb, mediaSession.getPrivate());
- return ice_session_gather_candidates(iceSession, ai->ai_addr, (socklen_t)ai->ai_addrlen) ? 1 : 0;
- } else {
- lInfo() << "ICE: bypass candidates gathering";
- ice_session_compute_candidates_foundations(iceSession);
- ice_session_eliminate_redundant_candidates(iceSession);
- ice_session_choose_default_candidates(iceSession);
- }
- return 0;
-}
-
-void IceAgent::getIceDefaultAddrAndPort (
- uint16_t componentID,
- const SalMediaDescription *md,
- const SalStreamDescription *stream,
- const char **addr,
- int *port
-) {
- if (componentID == 1) {
- *addr = stream->rtp_addr;
- *port = stream->rtp_port;
- } else if (componentID == 2) {
- *addr = stream->rtcp_addr;
- *port = stream->rtcp_port;
- } else
- return;
- if ((*addr)[0] == '\0') *addr = md->addr;
-}
-
-/**
- * Choose the preferred IP address to use to contact the STUN server from the list of IP addresses
- * the DNS resolution returned. If a NAT64 address is present, use it, otherwise if an IPv4 address
- * is present, use it, otherwise use an IPv6 address if it is present.
- */
-const struct addrinfo *IceAgent::getIcePreferredStunServerAddrinfo (const struct addrinfo *ai) {
- // Search for NAT64 addrinfo.
- const struct addrinfo *it = ai;
- while (it) {
- if (it->ai_family == AF_INET6) {
- struct sockaddr_storage ss;
- socklen_t sslen = sizeof(ss);
- memset(&ss, 0, sizeof(ss));
- bctbx_sockaddr_remove_nat64_mapping(it->ai_addr, (struct sockaddr *)&ss, &sslen);
- if (ss.ss_family == AF_INET) break;
- }
- it = it->ai_next;
- }
- const struct addrinfo *preferredAi = it;
- if (!preferredAi) {
- // Search for IPv4 addrinfo.
- it = ai;
- while (it) {
- if (it->ai_family == AF_INET)
- break;
- if ((it->ai_family == AF_INET6) && (it->ai_flags & AI_V4MAPPED))
- break;
- it = it->ai_next;
- }
- preferredAi = it;
- }
- if (!preferredAi) {
- // Search for IPv6 addrinfo.
- it = ai;
- while (it) {
- if (it->ai_family == AF_INET6)
- break;
- it = it->ai_next;
- }
- preferredAi = it;
- }
- return preferredAi;
-}
-
-bool IceAgent::iceParamsFoundInRemoteMediaDescription (const SalMediaDescription *md) {
- if ((md->ice_pwd[0] != '\0') && (md->ice_ufrag[0] != '\0'))
- return true;
- bool found = false;
- for (int i = 0; i < md->nb_streams; i++) {
- const SalStreamDescription *stream = &md->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (cl) {
- if ((stream->ice_pwd[0] != '\0') && (stream->ice_ufrag[0] != '\0'))
- found = true;
- else {
- found = false;
- break;
- }
- }
- }
- return found;
-}
-
-void IceAgent::updateIceStateInCallStatsForStream (LinphoneCallStats *stats, IceCheckList *cl) {
- if (ice_check_list_state(cl) != ICL_Completed) {
- _linphone_call_stats_set_ice_state(stats, LinphoneIceStateFailed);
- return;
- }
-
- switch (ice_check_list_selected_valid_candidate_type(cl)) {
- case ICT_HostCandidate:
- _linphone_call_stats_set_ice_state(stats, LinphoneIceStateHostConnection);
- break;
- case ICT_ServerReflexiveCandidate:
- case ICT_PeerReflexiveCandidate:
- _linphone_call_stats_set_ice_state(stats, LinphoneIceStateReflexiveConnection);
- break;
- case ICT_RelayedCandidate:
- _linphone_call_stats_set_ice_state(stats, LinphoneIceStateRelayConnection);
- break;
- case ICT_CandidateInvalid:
- case ICT_CandidateTypeMax:
- // Shall not happen.
- L_ASSERT(false);
- break;
- }
-}
-
-bool IceAgent::checkIceReinviteNeedsDeferedResponse(SalMediaDescription *md) {
- if (!iceSession || (ice_session_state(iceSession) != IS_Running))
- return false;
-
- for (int i = 0; i < md->nb_streams; i++) {
- SalStreamDescription *stream = &md->streams[i];
- IceCheckList *cl = ice_session_check_list(iceSession, i);
- if (!cl)
- continue;
-
- if (stream->ice_mismatch)
- return false;
- if ((stream->rtp_port == 0) || (ice_check_list_state(cl) != ICL_Running))
- continue;
-
- for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES; j++) {
- const SalIceRemoteCandidate *remote_candidate = &stream->ice_remote_candidates[j];
- if (remote_candidate->addr[0] != '\0')
- return true;
- }
- }
- return false;
-}
-
-LINPHONE_END_NAMESPACE
diff --git a/src/nat/ice-agent.h b/src/nat/ice-agent.h
deleted file mode 100644
index baa763a2cd424e933340fb5295ec752688f5f0ec..0000000000000000000000000000000000000000
--- a/src/nat/ice-agent.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2010-2019 Belledonne Communications SARL.
- *
- * This file is part of Liblinphone.
- *
- * This program is free software: you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef _L_ICE_AGENT_H_
-#define _L_ICE_AGENT_H_
-
-#include <mediastreamer2/ice.h>
-#include <ortp/event.h>
-
-#include "linphone/utils/general.h"
-
-// =============================================================================
-
-L_DECL_C_STRUCT_PREFIX_LESS(SalMediaDescription);
-L_DECL_C_STRUCT_PREFIX_LESS(SalStreamDescription);
-L_DECL_C_STRUCT(LinphoneCallStats);
-L_DECL_C_STRUCT(LinphoneCore);
-L_DECL_C_STRUCT(MediaStream);
-
-class MediaSession;
-
-LINPHONE_BEGIN_NAMESPACE
-
-class IceAgent {
-public:
- explicit IceAgent (MediaSession &mediaSession) : mediaSession(mediaSession) {}
-
- bool candidatesGathered () const;
- void checkSession (IceRole role, bool isReinvite);
- void deleteSession ();
- void gatheringFinished ();
- int getNbLosingPairs () const;
- IceSession *getIceSession () const {
- return iceSession;
- }
-
- bool hasCompleted () const;
- bool hasCompletedCheckList () const;
- bool hasSession () const {
- return !!iceSession;
- }
-
- bool isControlling () const;
- bool prepare (const SalMediaDescription *localDesc, bool incomingOffer, bool allowGathering = true);
- void prepareIceForStream (MediaStream *ms, bool createChecklist);
- void resetSession (IceRole role);
- void restartSession (IceRole role);
- void startConnectivityChecks ();
- void stopIceForInactiveStreams (SalMediaDescription *desc);
- void updateFromRemoteMediaDescription (const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc, bool isOffer);
- void updateIceStateInCallStats ();
- void updateLocalMediaDescriptionFromIce (SalMediaDescription *desc);
- /*
- * Checks if an incoming offer with ICE needs a delayed answer, because the ice session hasn't completed yet with
- * connecvity checks.
- */
- bool checkIceReinviteNeedsDeferedResponse (SalMediaDescription *md);
-
-private:
- void addLocalIceCandidates (int family, const char *addr, IceCheckList *audioCl, IceCheckList *videoCl, IceCheckList *textCl);
- bool checkForIceRestartAndSetRemoteCredentials (const SalMediaDescription *md, bool isOffer);
- void clearUnusedIceCandidates (const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc);
- void createIceCheckListsAndParseIceAttributes (const SalMediaDescription *md, bool iceRestarted);
- int gatherIceCandidates ();
- void getIceDefaultAddrAndPort (uint16_t componentID, const SalMediaDescription *md, const SalStreamDescription *stream, const char **addr, int *port);
- const struct addrinfo *getIcePreferredStunServerAddrinfo (const struct addrinfo *ai);
- bool iceParamsFoundInRemoteMediaDescription (const SalMediaDescription *md);
- void updateIceStateInCallStatsForStream (LinphoneCallStats *stats, IceCheckList *cl);
-
-private:
- MediaSession &mediaSession;
- IceSession *iceSession = nullptr;
-};
-
-LINPHONE_END_NAMESPACE
-
-#endif // ifndef _L_ICE_AGENT_H_
diff --git a/src/nat/ice-service.cpp b/src/nat/ice-service.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..0638788446dc790eea41c6a632ea4cd8f6c001a9
--- /dev/null
+++ b/src/nat/ice-service.cpp
@@ -0,0 +1,719 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "private.h"
+
+#include "ice-service.h"
+#include "conference/session/streams.h"
+#include "conference/session/media-session-p.h"
+#include "utils/if-addrs.h"
+
+
+using namespace::std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+IceService::IceService(StreamsGroup & sg) : mStreamsGroup(sg){
+}
+
+IceService::~IceService(){
+ deleteSession();
+}
+
+bool IceService::isActive() const{
+ return mIceSession != nullptr;
+}
+
+bool IceService::hasCompleted() const{
+ if (!isActive()) return true; // Completed because nothing to do.
+ return ice_session_state(mIceSession) == IS_Completed;
+}
+
+MediaSessionPrivate &IceService::getMediaSessionPrivate() const{
+ return mStreamsGroup.getMediaSessionPrivate();
+}
+
+bool IceService::iceFoundInMediaDescription (const SalMediaDescription *md) {
+ if ((md->ice_pwd[0] != '\0') && (md->ice_ufrag[0] != '\0'))
+ return true;
+ for (int i = 0; i < md->nb_streams; i++) {
+ const SalStreamDescription *stream = &md->streams[i];
+ if ((stream->ice_pwd[0] != '\0') && (stream->ice_ufrag[0] != '\0')){
+ return true;
+ }
+
+ }
+ return false;
+}
+
+void IceService::checkSession (IceRole role) {
+ LinphoneNatPolicy *natPolicy = getMediaSessionPrivate().getNatPolicy();
+ if (!natPolicy || !linphone_nat_policy_ice_enabled(natPolicy)){
+ return;
+ }
+
+ // Already created.
+ if (mIceSession)
+ return;
+
+ LinphoneConfig *config = linphone_core_get_config(getCCore());
+
+ if (lp_config_get_int(config, "net", "force_ice_disablement", 0)){
+ lWarning()<<"ICE is disabled in this version";
+ return;
+ }
+
+ mIceSession = ice_session_new();
+
+ // For backward compatibility purposes, shall be enabled by default in the future.
+ ice_session_enable_message_integrity_check(
+ mIceSession,
+ !!lp_config_get_int(config, "net", "ice_session_enable_message_integrity_check", 1)
+ );
+ if (lp_config_get_int(config, "net", "dont_default_to_stun_candidates", 0)) {
+ IceCandidateType types[ICT_CandidateTypeMax];
+ types[0] = ICT_HostCandidate;
+ types[1] = ICT_RelayedCandidate;
+ types[2] = ICT_CandidateInvalid;
+ ice_session_set_default_candidates_types(mIceSession, types);
+ }
+ ice_session_set_role(mIceSession, role);
+}
+
+void IceService::fillLocalMediaDescription(OfferAnswerContext & ctx){
+ if (!mIceSession) return;
+
+ if (mGatheringFinished){
+ if (ctx.remoteMediaDescription)
+ clearUnusedIceCandidates(ctx.localMediaDescription, ctx.remoteMediaDescription, ctx.localIsOfferer);
+
+ ice_session_compute_candidates_foundations(mIceSession);
+ ice_session_eliminate_redundant_candidates(mIceSession);
+ ice_session_choose_default_candidates(mIceSession);
+ mGatheringFinished = false;
+ }
+ updateLocalMediaDescriptionFromIce(ctx.localMediaDescription);
+}
+
+void IceService::createStreams(const OfferAnswerContext ¶ms){
+ checkSession(params.localIsOfferer ? IR_Controlling : IR_Controlled);
+
+ if (!mIceSession) return;
+
+ const auto & streams = mStreamsGroup.getStreams();
+ for (auto & stream : streams){
+ size_t index = stream->getIndex();
+ params.scopeStreamToIndex(index);
+ bool streamActive = sal_stream_description_enabled(params.localStreamDescription);
+
+ if (!params.localIsOfferer){
+ int bundleOwnerIndex = sal_media_description_get_index_of_transport_owner(params.remoteMediaDescription, params.remoteStreamDescription);
+ if (bundleOwnerIndex != -1 && bundleOwnerIndex != (int) index){
+ lInfo() << *stream << " is part of a bundle as secondary stream, ICE not needed.";
+ streamActive = false;
+ }
+ }else{
+ RtpInterface *i = dynamic_cast<RtpInterface*>(stream.get());
+ if (i && !i->isTransportOwner()){
+ lInfo() << *stream << " is currently part of a bundle as secondary stream, ICE not needed.";
+ streamActive = false;
+ }
+ }
+ IceCheckList *cl = ice_session_check_list(mIceSession, (int)index);
+ if (!cl && streamActive) {
+ cl = ice_check_list_new();
+ ice_session_add_check_list(mIceSession, cl, static_cast<unsigned int>(index));
+ lInfo() << "Created new ICE check list " << cl << " for stream #" << index;
+ } else if (cl && !streamActive){
+ ice_session_remove_check_list_from_idx(mIceSession, static_cast<unsigned int>(index));
+ cl = nullptr;
+ }
+ stream->setIceCheckList(cl);
+ stream->iceStateChanged();
+ }
+
+ if (!params.localIsOfferer){
+ if (params.remoteMediaDescription){
+ // This may delete the ice session.
+ updateFromRemoteMediaDescription(params.localMediaDescription, params.remoteMediaDescription, true);
+ }
+ }
+}
+
+bool IceService::prepare(){
+ if (!mIceSession) return false;
+
+ // Start ICE gathering if needed.
+ if (!ice_session_candidates_gathered(mIceSession)) {
+ int err = gatherIceCandidates();
+ if (err == 0) {
+ // Ice candidates gathering wasn't started, but we can proceed with the call anyway.
+ return false;
+ } else if (err == -1) {
+ deleteSession();
+ return false;
+ }
+ return true;
+ }
+ return false;
+}
+
+LinphoneCore *IceService::getCCore()const{
+ return mStreamsGroup.getCCore();
+}
+
+void IceService::gatherLocalCandidates(){
+ list<string> localAddrs = IfAddrs::fetchLocalAddresses();
+ bool ipv6Allowed = linphone_core_ipv6_enabled(getCCore());
+
+ const auto & streams = mStreamsGroup.getStreams();
+ for (auto & stream : streams){
+ size_t index = stream->getIndex();
+ IceCheckList *cl = ice_session_check_list(mIceSession, (int)index);
+ if (cl) {
+ if ((ice_check_list_state(cl) != ICL_Completed) && !ice_check_list_candidates_gathered(cl)) {
+ for (const string & addr : localAddrs){
+ int family = addr.find(':') != string::npos ? AF_INET6 : AF_INET;
+ if (family == AF_INET6 && !ipv6Allowed) continue;
+ ice_add_local_candidate(cl, "host", family, addr.c_str(), stream->getPortConfig().rtpPort, 1, nullptr);
+ ice_add_local_candidate(cl, "host", family, addr.c_str(), stream->getPortConfig().rtcpPort, 2, nullptr);
+ }
+ }
+ }
+ }
+}
+
+/** Return values:
+ * 1: STUN gathering is started
+ * 0: no STUN gathering is started, but it's ok to proceed with ICE anyway (with local candidates only or because STUN gathering was already done before)
+ * -1: no gathering started and something went wrong with local candidates. There is no way to start the ICE session.
+ */
+int IceService::gatherIceCandidates () {
+ const struct addrinfo *ai = nullptr;
+ int err = 0;
+
+ LinphoneNatPolicy *natPolicy = getMediaSessionPrivate().getNatPolicy();
+ if (natPolicy && linphone_nat_policy_stun_server_activated(natPolicy)) {
+ ai = linphone_nat_policy_get_stun_server_addrinfo(natPolicy);
+ if (ai)
+ ai = getIcePreferredStunServerAddrinfo(ai);
+ else
+ lWarning() << "Failed to resolve STUN server for ICE gathering, continuing without STUN";
+ } else
+ lWarning() << "ICE is used without STUN server";
+ LinphoneCore *core = getCCore();
+ ice_session_enable_forced_relay(mIceSession, core->forced_ice_relay);
+ ice_session_enable_short_turn_refresh(mIceSession, core->short_turn_refresh);
+
+ // Gather local host candidates.
+ gatherLocalCandidates();
+
+ if (ai && natPolicy && linphone_nat_policy_stun_server_activated(natPolicy)) {
+ string server = linphone_nat_policy_get_stun_server(natPolicy);
+ lInfo() << "ICE: gathering candidates from [" << server << "] using " << (linphone_nat_policy_turn_enabled(natPolicy) ? "TURN" : "STUN");
+ // Gather local srflx candidates.
+ ice_session_enable_turn(mIceSession, linphone_nat_policy_turn_enabled(natPolicy));
+ ice_session_set_stun_auth_requested_cb(mIceSession, MediaSessionPrivate::stunAuthRequestedCb, &getMediaSessionPrivate());
+ err = ice_session_gather_candidates(mIceSession, ai->ai_addr, (socklen_t)ai->ai_addrlen) ? 1 : 0;
+ } else {
+ lInfo() << "ICE: bypass candidates gathering";
+ }
+ if (err == 0) gatheringFinished();
+ return err;
+}
+
+bool IceService::checkForIceRestartAndSetRemoteCredentials (const SalMediaDescription *md, bool isOffer) {
+ bool iceRestarted = false;
+ string addr = md->addr;
+ if ((addr == "0.0.0.0") || (addr == "::0")) {
+ ice_session_restart(mIceSession, isOffer ? IR_Controlled : IR_Controlling);
+ iceRestarted = true;
+ } else {
+ for (int i = 0; i < md->nb_streams; i++) {
+ const SalStreamDescription *stream = &md->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ string rtpAddr = stream->rtp_addr;
+ if (cl && (rtpAddr == "0.0.0.0")) {
+ ice_session_restart(mIceSession, isOffer ? IR_Controlled : IR_Controlling);
+ iceRestarted = true;
+ break;
+ }
+ }
+ }
+ if (!ice_session_remote_ufrag(mIceSession) && !ice_session_remote_pwd(mIceSession)) {
+ ice_session_set_remote_credentials(mIceSession, md->ice_ufrag, md->ice_pwd);
+ } else if (ice_session_remote_credentials_changed(mIceSession, md->ice_ufrag, md->ice_pwd)) {
+ if (!iceRestarted) {
+ ice_session_restart(mIceSession, isOffer ? IR_Controlled : IR_Controlling);
+ iceRestarted = true;
+ }
+ ice_session_set_remote_credentials(mIceSession, md->ice_ufrag, md->ice_pwd);
+ }
+ for (int i = 0; i < md->nb_streams; i++) {
+ const SalStreamDescription *stream = &md->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ if (cl && (stream->ice_pwd[0] != '\0') && (stream->ice_ufrag[0] != '\0')) {
+ if (ice_check_list_remote_credentials_changed(cl, stream->ice_ufrag, stream->ice_pwd)) {
+ if (!iceRestarted && ice_check_list_get_remote_ufrag(cl) && ice_check_list_get_remote_pwd(cl)) {
+ // Restart only if remote ufrag/paswd was already set.
+ ice_session_restart(mIceSession, isOffer ? IR_Controlled : IR_Controlling);
+ iceRestarted = true;
+ }
+ ice_check_list_set_remote_credentials(cl, stream->ice_ufrag, stream->ice_pwd);
+ }
+ }
+ }
+ return iceRestarted;
+}
+
+void IceService::getIceDefaultAddrAndPort (
+ uint16_t componentID,
+ const SalMediaDescription *md,
+ const SalStreamDescription *stream,
+ const char **addr,
+ int *port
+) {
+ if (componentID == 1) {
+ *addr = stream->rtp_addr;
+ *port = stream->rtp_port;
+ } else if (componentID == 2) {
+ *addr = stream->rtcp_addr;
+ *port = stream->rtcp_port;
+ } else
+ return;
+ if ((*addr)[0] == '\0') *addr = md->addr;
+}
+
+void IceService::createIceCheckListsAndParseIceAttributes (const SalMediaDescription *md, bool iceRestarted) {
+ for (int i = 0; i < md->nb_streams; i++) {
+ const SalStreamDescription *stream = &md->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ if (!cl)
+ continue;
+ if (stream->ice_mismatch) {
+ ice_check_list_set_state(cl, ICL_Failed);
+ continue;
+ }
+ if (stream->rtp_port == 0) {
+ ice_session_remove_check_list(mIceSession, cl);
+ mStreamsGroup.getStream((size_t)i)->setIceCheckList(nullptr);
+ continue;
+ }
+ if ((stream->ice_pwd[0] != '\0') && (stream->ice_ufrag[0] != '\0'))
+ ice_check_list_set_remote_credentials(cl, stream->ice_ufrag, stream->ice_pwd);
+ for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_CANDIDATES; j++) {
+ bool defaultCandidate = false;
+ const SalIceCandidate *candidate = &stream->ice_candidates[j];
+ if (candidate->addr[0] == '\0')
+ break;
+ if ((candidate->componentID == 0) || (candidate->componentID > 2))
+ continue;
+ const char *addr = nullptr;
+ int port = 0;
+ getIceDefaultAddrAndPort(static_cast<uint16_t>(candidate->componentID), md, stream, &addr, &port);
+ if (addr && (candidate->port == port) && (strlen(candidate->addr) == strlen(addr)) && (strcmp(candidate->addr, addr) == 0))
+ defaultCandidate = true;
+ int family = AF_INET;
+ if (strchr(candidate->addr, ':'))
+ family = AF_INET6;
+ ice_add_remote_candidate(
+ cl, candidate->type, family, candidate->addr, candidate->port,
+ static_cast<uint16_t>(candidate->componentID),
+ candidate->priority, candidate->foundation, defaultCandidate
+ );
+ }
+ if (!iceRestarted) {
+ bool losingPairsAdded = false;
+ for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES; j++) {
+ const SalIceRemoteCandidate *remoteCandidate = &stream->ice_remote_candidates[j];
+ const char *addr = nullptr;
+ int port = 0;
+ int componentID = j + 1;
+ if (remoteCandidate->addr[0] == '\0') break;
+ getIceDefaultAddrAndPort(static_cast<uint16_t>(componentID), md, stream, &addr, &port);
+
+ // If we receive a re-invite with remote-candidates, supply these pairs to the ice check list.
+ // They might be valid pairs already selected, or losing pairs.
+
+ int remoteFamily = AF_INET;
+ if (strchr(remoteCandidate->addr, ':'))
+ remoteFamily = AF_INET6;
+ int family = AF_INET;
+ if (strchr(addr, ':'))
+ family = AF_INET6;
+ ice_add_losing_pair(cl, static_cast<uint16_t>(j + 1), remoteFamily, remoteCandidate->addr, remoteCandidate->port, family, addr, port);
+ losingPairsAdded = true;
+ }
+ if (losingPairsAdded)
+ ice_check_list_check_completed(cl);
+ }
+ }
+}
+
+void IceService::clearUnusedIceCandidates (const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc, bool localIsOfferer) {
+ for (int i = 0; i < remoteDesc->nb_streams; i++) {
+ const SalStreamDescription *localStream = &localDesc->streams[i];
+ const SalStreamDescription *stream = &remoteDesc->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ if (!cl || !localStream)
+ continue;
+ if ((localIsOfferer && stream->rtcp_mux && localStream->rtcp_mux)
+ || (!localIsOfferer && stream->rtcp_mux)) {
+ ice_check_list_remove_rtcp_candidates(cl);
+ }
+ }
+}
+
+void IceService::updateFromRemoteMediaDescription(const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc, bool isOffer) {
+ if (!mIceSession)
+ return;
+
+ if (!iceFoundInMediaDescription(remoteDesc)) {
+ // Response from remote does not contain mandatory ICE attributes, delete the session.
+ deleteSession();
+ return;
+ }
+
+ // Check for ICE restart and set remote credentials.
+ bool iceRestarted = checkForIceRestartAndSetRemoteCredentials(remoteDesc, isOffer);
+
+ // Create ICE check lists if needed and parse ICE attributes.
+ createIceCheckListsAndParseIceAttributes(remoteDesc, iceRestarted);
+ for (int i = 0; i < remoteDesc->nb_streams; i++) {
+ const SalStreamDescription *stream = &remoteDesc->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ if (!cl) continue;
+ if (!sal_stream_description_enabled(stream) || stream->rtp_port == 0) {
+ /*
+ * rtp_port == 0 is true when it is a secondary stream part of bundle.
+ */
+ ice_session_remove_check_list_from_idx(mIceSession, static_cast<unsigned int>(i));
+ auto stream = mStreamsGroup.getStream(i);
+ stream->setIceCheckList(nullptr);
+ stream->iceStateChanged();
+
+ }
+ }
+ clearUnusedIceCandidates(localDesc, remoteDesc, !isOffer);
+ ice_session_check_mismatch(mIceSession);
+
+ if (ice_session_nb_check_lists(mIceSession) == 0) {
+ deleteSession();
+ }
+}
+
+
+void IceService::updateLocalMediaDescriptionFromIce (SalMediaDescription *desc) {
+ if (!mIceSession)
+ return;
+ IceCandidate *rtpCandidate = nullptr;
+ IceCandidate *rtcpCandidate = nullptr;
+ bool result = false;
+ IceSessionState sessionState = ice_session_state(mIceSession);
+ if (sessionState == IS_Completed) {
+ IceCheckList *firstCl = nullptr;
+ for (int i = 0; i < desc->nb_streams; i++) {
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ if (cl) {
+ firstCl = cl;
+ break;
+ }
+ }
+ if (firstCl)
+ result = !!ice_check_list_selected_valid_local_candidate(firstCl, &rtpCandidate, nullptr);
+ if (result) {
+ strncpy(desc->addr, rtpCandidate->taddr.ip, sizeof(desc->addr));
+ } else {
+ lWarning() << "If ICE has completed successfully, rtp_candidate should be set!";
+ ice_dump_valid_list(firstCl);
+ }
+ }
+
+ strncpy(desc->ice_pwd, ice_session_local_pwd(mIceSession), sizeof(desc->ice_pwd)-1);
+ strncpy(desc->ice_ufrag, ice_session_local_ufrag(mIceSession), sizeof(desc->ice_ufrag)-1);
+
+ for (int i = 0; i < desc->nb_streams; i++) {
+ SalStreamDescription *stream = &desc->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ rtpCandidate = rtcpCandidate = nullptr;
+ if (!sal_stream_description_enabled(stream) || !cl || stream->rtp_port == 0)
+ continue;
+ if (ice_check_list_state(cl) == ICL_Completed) {
+ result = !!ice_check_list_selected_valid_local_candidate(ice_session_check_list(mIceSession, i), &rtpCandidate, &rtcpCandidate);
+ } else {
+ result = !!ice_check_list_default_local_candidate(ice_session_check_list(mIceSession, i), &rtpCandidate, &rtcpCandidate);
+ }
+ if (result) {
+ strncpy(stream->rtp_addr, rtpCandidate->taddr.ip, sizeof(stream->rtp_addr));
+ strncpy(stream->rtcp_addr, rtcpCandidate->taddr.ip, sizeof(stream->rtcp_addr));
+ stream->rtp_port = rtpCandidate->taddr.port;
+ stream->rtcp_port = rtcpCandidate->taddr.port;
+ } else {
+ memset(stream->rtp_addr, 0, sizeof(stream->rtp_addr));
+ memset(stream->rtcp_addr, 0, sizeof(stream->rtcp_addr));
+ }
+ if ((strlen(ice_check_list_local_pwd(cl)) != strlen(desc->ice_pwd)) || (strcmp(ice_check_list_local_pwd(cl), desc->ice_pwd)))
+ strncpy(stream->ice_pwd, ice_check_list_local_pwd(cl), sizeof(stream->ice_pwd) - 1);
+ else
+ memset(stream->ice_pwd, 0, sizeof(stream->ice_pwd));
+ if ((strlen(ice_check_list_local_ufrag(cl)) != strlen(desc->ice_ufrag)) || (strcmp(ice_check_list_local_ufrag(cl), desc->ice_ufrag)))
+ strncpy(stream->ice_ufrag, ice_check_list_local_ufrag(cl), sizeof(stream->ice_ufrag) -1 );
+ else
+ memset(stream->ice_pwd, 0, sizeof(stream->ice_pwd));
+ stream->ice_mismatch = ice_check_list_is_mismatch(cl);
+ if ((ice_check_list_state(cl) == ICL_Running) || (ice_check_list_state(cl) == ICL_Completed)) {
+ memset(stream->ice_candidates, 0, sizeof(stream->ice_candidates));
+ int nbCandidates = 0;
+ for (int j = 0; j < MIN((int)bctbx_list_size(cl->local_candidates), SAL_MEDIA_DESCRIPTION_MAX_ICE_CANDIDATES); j++) {
+ SalIceCandidate *salCandidate = &stream->ice_candidates[nbCandidates];
+ IceCandidate *iceCandidate = reinterpret_cast<IceCandidate *>(bctbx_list_nth_data(cl->local_candidates, j));
+ const char *defaultAddr = nullptr;
+ int defaultPort = 0;
+ if (iceCandidate->componentID == 1) {
+ defaultAddr = stream->rtp_addr;
+ defaultPort = stream->rtp_port;
+ } else if (iceCandidate->componentID == 2) {
+ defaultAddr = stream->rtcp_addr;
+ defaultPort = stream->rtcp_port;
+ } else
+ continue;
+ if (defaultAddr[0] == '\0')
+ defaultAddr = desc->addr;
+ // Only include the candidates matching the default destination for each component of the stream if the state is Completed as specified in RFC5245 section 9.1.2.2.
+ if (
+ ice_check_list_state(cl) == ICL_Completed &&
+ !((iceCandidate->taddr.port == defaultPort) && (strlen(iceCandidate->taddr.ip) == strlen(defaultAddr)) && (strcmp(iceCandidate->taddr.ip, defaultAddr) == 0))
+ )
+ continue;
+ strncpy(salCandidate->foundation, iceCandidate->foundation, sizeof(salCandidate->foundation));
+ salCandidate->componentID = iceCandidate->componentID;
+ salCandidate->priority = iceCandidate->priority;
+ strncpy(salCandidate->type, ice_candidate_type(iceCandidate), sizeof(salCandidate->type) - 1);
+ strncpy(salCandidate->addr, iceCandidate->taddr.ip, sizeof(salCandidate->addr));
+ salCandidate->port = iceCandidate->taddr.port;
+ if (iceCandidate->base && (iceCandidate->base != iceCandidate)) {
+ strncpy(salCandidate->raddr, iceCandidate->base->taddr.ip, sizeof(salCandidate->raddr));
+ salCandidate->rport = iceCandidate->base->taddr.port;
+ }
+ nbCandidates++;
+ }
+ }
+ if ((ice_check_list_state(cl) == ICL_Completed) && (ice_session_role(mIceSession) == IR_Controlling)) {
+ memset(stream->ice_remote_candidates, 0, sizeof(stream->ice_remote_candidates) -1);
+ if (ice_check_list_selected_valid_remote_candidate(cl, &rtpCandidate, &rtcpCandidate)) {
+ strncpy(stream->ice_remote_candidates[0].addr, rtpCandidate->taddr.ip, sizeof(stream->ice_remote_candidates[0].addr));
+ stream->ice_remote_candidates[0].port = rtpCandidate->taddr.port;
+ if (rtcpCandidate){
+ strncpy(stream->ice_remote_candidates[1].addr, rtcpCandidate->taddr.ip, sizeof(stream->ice_remote_candidates[1].addr));
+ stream->ice_remote_candidates[1].port = rtcpCandidate->taddr.port;
+ }
+ } else
+ lError() << "ice: Selected valid remote candidates should be present if the check list is in the Completed state";
+ } else {
+ for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES; j++) {
+ stream->ice_remote_candidates[j].addr[0] = '\0';
+ stream->ice_remote_candidates[j].port = 0;
+ }
+ }
+ }
+}
+
+void IceService::gatheringFinished () {
+ if (!mIceSession)
+ return;
+
+ int pingTime = ice_session_average_gathering_round_trip_time(mIceSession);
+ if (pingTime >= 0) {
+ /* FIXME: is ping time still useful for the MediaSession ? */
+ getMediaSessionPrivate().setPingTime(pingTime);
+ }
+ mGatheringFinished = true;
+}
+
+
+/**
+ * Choose the preferred IP address to use to contact the STUN server from the list of IP addresses
+ * the DNS resolution returned. If a NAT64 address is present, use it, otherwise if an IPv4 address
+ * is present, use it, otherwise use an IPv6 address if it is present.
+ */
+const struct addrinfo *IceService::getIcePreferredStunServerAddrinfo (const struct addrinfo *ai) {
+ // Search for NAT64 addrinfo.
+ const struct addrinfo *it = ai;
+ while (it) {
+ if (it->ai_family == AF_INET6) {
+ struct sockaddr_storage ss;
+ socklen_t sslen = sizeof(ss);
+ memset(&ss, 0, sizeof(ss));
+ bctbx_sockaddr_remove_nat64_mapping(it->ai_addr, (struct sockaddr *)&ss, &sslen);
+ if (ss.ss_family == AF_INET) break;
+ }
+ it = it->ai_next;
+ }
+ const struct addrinfo *preferredAi = it;
+ if (!preferredAi) {
+ // Search for IPv4 addrinfo.
+ it = ai;
+ while (it) {
+ if (it->ai_family == AF_INET)
+ break;
+ if ((it->ai_family == AF_INET6) && (it->ai_flags & AI_V4MAPPED))
+ break;
+ it = it->ai_next;
+ }
+ preferredAi = it;
+ }
+ if (!preferredAi) {
+ // Search for IPv6 addrinfo.
+ it = ai;
+ while (it) {
+ if (it->ai_family == AF_INET6)
+ break;
+ it = it->ai_next;
+ }
+ preferredAi = it;
+ }
+ return preferredAi;
+}
+
+void IceService::finishPrepare(){
+ if (!mIceSession) return;
+ gatheringFinished();
+}
+
+void IceService::render(const OfferAnswerContext & ctx, CallSession::State targetState){
+ if (!mIceSession) return;
+
+ updateFromRemoteMediaDescription(ctx.localMediaDescription, ctx.remoteMediaDescription, !ctx.localIsOfferer);
+ if (mIceSession && ice_session_state(mIceSession) != IS_Completed)
+ ice_session_start_connectivity_checks(mIceSession);
+}
+
+void IceService::sessionConfirmed(const OfferAnswerContext &ctx){
+}
+
+void IceService::stop(){
+ //Nothing to do. The ice session can survive.
+}
+
+void IceService::finish(){
+ deleteSession();
+}
+
+void IceService::deleteSession () {
+ if (!mIceSession)
+ return;
+ /* clear all check lists */
+ for (auto & stream : mStreamsGroup.getStreams())
+ stream->setIceCheckList(nullptr);
+ ice_session_destroy(mIceSession);
+ mIceSession = nullptr;
+}
+
+void IceService::setListener(IceServiceListener *listener){
+ mListener = listener;
+}
+
+void IceService::restartSession (IceRole role) {
+ if (!mIceSession)
+ return;
+ ice_session_restart(mIceSession, role);
+}
+
+void IceService::resetSession() {
+ if (!mIceSession)
+ return;
+ ice_session_reset(mIceSession, IR_Controlling);
+}
+
+bool IceService::hasCompletedCheckList () const {
+ if (!mIceSession)
+ return false;
+ switch (ice_session_state(mIceSession)) {
+ case IS_Completed:
+ case IS_Failed:
+ return !!ice_session_has_completed_check_list(mIceSession);
+ default:
+ return false;
+ }
+}
+
+void IceService::handleIceEvent(const OrtpEvent *ev){
+ OrtpEventType evt = ortp_event_get_type(ev);
+ const OrtpEventData *evd = ortp_event_get_data(const_cast<OrtpEvent*>(ev));
+
+ switch (evt){
+ case ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED:
+ if (hasCompletedCheckList()) {
+ if (mListener) mListener->onIceCompleted(*this);
+ }
+ break;
+ case ORTP_EVENT_ICE_GATHERING_FINISHED:
+ if (!evd->info.ice_processing_successful)
+ lWarning() << "No STUN answer from [" << linphone_nat_policy_get_stun_server(getMediaSessionPrivate().getNatPolicy()) << "], continuing without STUN";
+ mStreamsGroup.finishPrepare();
+ if (mListener) mListener->onGatheringFinished(*this);
+ break;
+ case ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED:
+ if (mListener) mListener->onLosingPairsCompleted(*this);
+ break;
+ case ORTP_EVENT_ICE_RESTART_NEEDED:
+ if (mListener) mListener->onIceRestartNeeded(*this);
+ break;
+ default:
+ lError() << "IceService::handleIceEvent() is passed with a non-ICE event.";
+ break;
+ }
+ /* Notify all the streams of the ICE state change, so that they can update their stats and so on. */
+ for(auto & stream : mStreamsGroup.getStreams()){
+ stream->iceStateChanged();
+ }
+}
+
+bool IceService::isControlling () const {
+ if (!mIceSession)
+ return false;
+ return ice_session_role(mIceSession) == IR_Controlling;
+}
+
+bool IceService::reinviteNeedsDeferedResponse(SalMediaDescription *remoteMd){
+ if (!mIceSession || (ice_session_state(mIceSession) != IS_Running))
+ return false;
+
+ for (int i = 0; i < remoteMd->nb_streams; i++) {
+ SalStreamDescription *stream = &remoteMd->streams[i];
+ IceCheckList *cl = ice_session_check_list(mIceSession, i);
+ if (!cl)
+ continue;
+
+ if (stream->ice_mismatch)
+ return false;
+ if ((stream->rtp_port == 0) || (ice_check_list_state(cl) != ICL_Running))
+ continue;
+
+ for (int j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_ICE_REMOTE_CANDIDATES; j++) {
+ const SalIceRemoteCandidate *remote_candidate = &stream->ice_remote_candidates[j];
+ if (remote_candidate->addr[0] != '\0')
+ return true;
+ }
+ }
+ return false;
+}
+
+
+LINPHONE_END_NAMESPACE
diff --git a/src/nat/ice-service.h b/src/nat/ice-service.h
new file mode 100644
index 0000000000000000000000000000000000000000..f4db58dbec256a70f09ecee7dcddea5f193651c9
--- /dev/null
+++ b/src/nat/ice-service.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef ice_service_h
+#define ice_service_h
+
+#include <memory>
+
+#include "conference/session/call-session.h"
+#include "conference/session/media-description-renderer.h"
+
+LINPHONE_BEGIN_NAMESPACE
+
+class StreamsGroup;
+class MediaSessionPrivate;
+class IceServiceListener;
+
+class IceService : public MediaDescriptionRenderer{
+public:
+ IceService(StreamsGroup & sg);
+ virtual ~IceService();
+
+ bool isActive() const;
+
+ /* Returns true if ICE has completed succesfully. */
+ bool hasCompleted() const;
+
+ /* Returns true if ICE has finished with the check lists processing, even if it has failed for some of the check list.*/
+ bool hasCompletedCheckList()const;
+
+ bool isControlling () const;
+
+ /* The ICE restart procedure as in RFC */
+ void restartSession(IceRole role);
+
+ /* Called after a network connectivity change, to restart ICE from the beginning.*/
+ void resetSession();
+
+ /* Returns true if the incoming offer requires a defered response, due to check-list(s) not yet completed.*/
+ bool reinviteNeedsDeferedResponse(SalMediaDescription *remoteMd);
+
+ void createStreams(const OfferAnswerContext ¶ms);
+ /**
+ * Called by the StreamsGroup when the local media description must be filled with ICE parameters.
+ *
+ */
+ virtual void fillLocalMediaDescription(OfferAnswerContext & ctx) override;
+ /*
+ * Prepare to run.
+ */
+ virtual bool prepare() override;
+ /*
+ * Prepare stage is finishing.
+ * Called by the StreamsGroup (who receives mediastreamer2 events) when the ICE gathering is finished.
+ *
+ */
+ virtual void finishPrepare() override;
+ /*
+ * Render the streams according to offer answer context.
+ */
+ virtual void render(const OfferAnswerContext & ctx, CallSession::State targetState) override;
+ /*
+ * Called to notify that the session is confirmed (corresponding to SIP ACK).
+ */
+ virtual void sessionConfirmed(const OfferAnswerContext &ctx) override;
+ /*
+ * Stop rendering streams.
+ */
+ virtual void stop() override;
+ /*
+ * Release engine's resource, pending object destruction.
+ */
+ virtual void finish() override;
+
+ /*
+ * Set the listener to get notified of major ICE events. Used by the MediaSession to perform required signaling operations.
+ */
+ void setListener(IceServiceListener *listener);
+
+ /*
+ * Called by streams (who receive oRTP events) to notify ICE related events to the IceService.
+ * Ideally the IceService should place its own listener to these ortp events, but well oRTP is C and has to be simple.
+ */
+ void handleIceEvent(const OrtpEvent *ev);
+
+ /**
+ * used by non-regression tests only.
+ */
+ IceSession *getSession()const{
+ return mIceSession;
+ }
+private:
+ MediaSessionPrivate &getMediaSessionPrivate()const;
+ LinphoneCore *getCCore()const;
+ bool iceFoundInMediaDescription (const SalMediaDescription *md);
+ const struct addrinfo *getIcePreferredStunServerAddrinfo (const struct addrinfo *ai);
+ void updateLocalMediaDescriptionFromIce(SalMediaDescription *desc);
+ void getIceDefaultAddrAndPort(uint16_t componentID, const SalMediaDescription *md, const SalStreamDescription *stream, const char **addr, int *port);
+ void clearUnusedIceCandidates (const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc, bool localIsOfferer);
+ bool checkForIceRestartAndSetRemoteCredentials (const SalMediaDescription *md, bool isOffer);
+ void createIceCheckListsAndParseIceAttributes (const SalMediaDescription *md, bool iceRestarted);
+ void updateFromRemoteMediaDescription (const SalMediaDescription *localDesc, const SalMediaDescription *remoteDesc, bool isOffer);
+ void gatheringFinished();
+ void deleteSession();
+ void checkSession (IceRole role);
+ int gatherIceCandidates ();
+ void gatherLocalCandidates();
+ StreamsGroup & mStreamsGroup;
+ IceSession * mIceSession = nullptr;
+ IceServiceListener *mListener = nullptr;
+ bool mGatheringFinished = false;
+
+};
+
+class IceServiceListener{
+public:
+ virtual void onGatheringFinished(IceService &service) = 0;
+ virtual void onIceCompleted(IceService &service) = 0;
+ virtual void onLosingPairsCompleted(IceService &service) = 0;
+ virtual void onIceRestartNeeded(IceService & service) = 0;
+ virtual ~IceServiceListener() = default;
+};
+
+LINPHONE_END_NAMESPACE
+
+#endif
+
diff --git a/src/nat/stun-client.cpp b/src/nat/stun-client.cpp
index 60f5f20be53785a4d22329a43827342743235466..07ed9b257a7178c5d72f6c4cc49e0690775096d0 100644
--- a/src/nat/stun-client.cpp
+++ b/src/nat/stun-client.cpp
@@ -147,7 +147,7 @@ int StunClient::run (int audioPort, int videoPort, int textPort) {
void StunClient::updateMediaDescription (SalMediaDescription *md) const {
if (!stunDiscoveryDone) return;
for (int i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
- if (!sal_stream_description_active(&md->streams[i]))
+ if (!sal_stream_description_enabled(&md->streams[i]))
continue;
if (md->streams[i].type == SalAudio && audioCandidate.port != 0) {
strncpy(md->streams[i].rtp_addr, audioCandidate.address.c_str(), sizeof(md->streams[i].rtp_addr));
diff --git a/src/sal/call-op.cpp b/src/sal/call-op.cpp
index 380f7ed5dab2bbb0226391d367b64f775e389359..08733c3f670c75370303a03972e5ca4e179f599d 100644
--- a/src/sal/call-op.cpp
+++ b/src/sal/call-op.cpp
@@ -488,13 +488,13 @@ void SalCallOp::processResponseCb (void *userCtx, const belle_sip_response_event
}
// Ref the ack request so that it is not destroyed when the call_ack_being_sent callbacks is called
belle_sip_object_ref(ack);
+ belle_sip_message_add_header(BELLE_SIP_MESSAGE(ack), BELLE_SIP_HEADER(op->mRoot->mUserAgentHeader));
+ op->mRoot->mCallbacks.call_accepted(op); // INVITE
if (op->mSdpAnswer) {
setSdp(BELLE_SIP_MESSAGE(ack), op->mSdpAnswer);
belle_sip_object_unref(op->mSdpAnswer);
op->mSdpAnswer = nullptr;
}
- belle_sip_message_add_header(BELLE_SIP_MESSAGE(ack), BELLE_SIP_HEADER(op->mRoot->mUserAgentHeader));
- op->mRoot->mCallbacks.call_accepted(op); // INVITE
op->mRoot->mCallbacks.call_ack_being_sent(op, reinterpret_cast<SalCustomHeader *>(ack));
belle_sip_dialog_send_ack(op->mDialog, ack);
belle_sip_object_unref(ack);
diff --git a/src/utils/if-addrs.cpp b/src/utils/if-addrs.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..c04d38683f283725ea932cb5d4c5e22be996f3d2
--- /dev/null
+++ b/src/utils/if-addrs.cpp
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2010-2020 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "private.h"
+#include "tester_utils.h"
+#include "c-wrapper/internal/c-tools.h"
+
+#ifdef HAVE_GETIFADDRS
+#include <sys/types.h>
+#include <ifaddrs.h>
+#include <net/if.h>
+#endif
+
+#include "if-addrs.h"
+
+using namespace std;
+
+LINPHONE_BEGIN_NAMESPACE
+
+#ifdef HAVE_GETIFADDRS
+list<string> IfAddrs::fetchWithGetIfAddrs(){
+ list<string> ret;
+ struct ifaddrs *ifap = nullptr;
+
+ lInfo() << "Fetching current local IP addresses using getifaddrs().";
+
+ if (getifaddrs(&ifap) == 0){
+ struct ifaddrs *ifaddr;
+ for (ifaddr = ifap; ifaddr != nullptr; ifaddr = ifaddr->ifa_next){
+ if (ifaddr->ifa_flags & IFF_LOOPBACK) continue;
+ if (ifaddr->ifa_flags & IFF_UP){
+ struct sockaddr *saddr = ifaddr->ifa_addr;
+ char addr[INET6_ADDRSTRLEN] = { 0 };
+ if (!saddr){
+ lError() << "NULL sockaddr returned by getifaddrs().";
+ continue;
+ }
+ switch (saddr->sa_family){
+ case AF_INET:
+ if (inet_ntop(AF_INET, &((struct sockaddr_in*)saddr)->sin_addr, addr, sizeof(addr)) != nullptr){
+ ret.push_back(addr);
+ }else{
+ lError() << "inet_ntop() failed with AF_INET: " << strerror(errno);
+ }
+ break;
+ case AF_INET6:
+ if (inet_ntop(AF_INET6, &((struct sockaddr_in6*)saddr)->sin6_addr, addr, sizeof(addr)) != nullptr){
+ ret.push_back(addr);
+ }else{
+ lError() << "inet_ntop() failed with AF_INET6: " << strerror(errno);
+ }
+ break;
+ default:
+ // ignored.
+ break;
+ }
+ }
+ }
+ freeifaddrs(ifap);
+ }else{
+ lError() << "getifaddrs(): " << strerror(errno);
+ }
+ return ret;
+}
+#endif
+
+list<string> IfAddrs::fetchLocalAddresses(){
+ list<string> ret;
+
+#ifdef HAVE_GETIFADDRS
+ ret = fetchWithGetIfAddrs();
+#endif
+ /*
+ * FIXME: implement here code for WIN32 that fetches all addresses of all interfaces.
+ */
+
+ /*
+ * Finally if none of the above methods worked, fallback with linphone_core_get_local_ip() that uses the socket/connect/getsockname method
+ * to get the local ip address that has the route to public internet.
+ */
+ if (ret.empty()){
+ lInfo() << "Fetching local ip addresses using the connect() method.";
+ char localAddr[LINPHONE_IPADDR_SIZE];
+
+ if (linphone_core_get_local_ip_for(AF_INET6, nullptr, localAddr) == 0) {
+ ret.push_back(localAddr);
+ }else{
+ lInfo() << "IceService::fetchLocalAddresses(): Fail to get default IPv6";
+ }
+
+ if (linphone_core_get_local_ip_for(AF_INET, nullptr, localAddr) == 0){
+ ret.push_back(localAddr);
+ }else{
+ lInfo() << "IceService::fetchLocalAddresses(): Fail to get default IPv4";
+ }
+ }
+ return ret;
+}
+
+LINPHONE_END_NAMESPACE
+
+bctbx_list_t *linphone_fetch_local_addresses(void){
+ return LinphonePrivate::Wrapper::getCListFromCppList(LinphonePrivate::IfAddrs::fetchLocalAddresses());
+}
+
diff --git a/src/utils/if-addrs.h b/src/utils/if-addrs.h
new file mode 100644
index 0000000000000000000000000000000000000000..2320b55e0598d99f00f37e724c851c54d26d0fb5
--- /dev/null
+++ b/src/utils/if-addrs.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2010-2020 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef linphone_if_addrs_h
+#define linphone_if_addrs_h
+
+#include <list>
+#include <string>
+
+LINPHONE_BEGIN_NAMESPACE
+
+class IfAddrs{
+public:
+ static std::list<std::string> fetchLocalAddresses();
+private:
+ static std::list<std::string> fetchWithGetIfAddrs();
+};
+
+LINPHONE_END_NAMESPACE
+
+#endif
diff --git a/tester/CMakeLists.txt b/tester/CMakeLists.txt
index ec196fda3484c09e1316968b3e5592c430987a81..31699ee9c6776c20b698a67200deb3c498ef92e9 100644
--- a/tester/CMakeLists.txt
+++ b/tester/CMakeLists.txt
@@ -223,6 +223,7 @@ set(SOURCE_FILES_C
tunnel_tester.c
vcard_tester.c
video_tester.c
+ call_with_rtp_bundle.c
)
set(SOURCE_FILES_CXX
diff --git a/tester/call_ice.c b/tester/call_ice.c
index 51cb776f99f46f1da56fd7c13f3a24df09041373..c48e061ae52df926af1a7b7384c6640ed45538f0 100644
--- a/tester/call_ice.c
+++ b/tester/call_ice.c
@@ -223,9 +223,12 @@ static void call_with_ice_and_rtcp_mux_without_reinvite(void){
_call_with_rtcp_mux(TRUE, TRUE, TRUE,FALSE);
}
-static bool_t is_matching_local_v4_or_v6(const char *ip, const char *localv4, const char *localv6){
+static bool_t is_matching_a_local_address(const char *ip, const bctbx_list_t *addresses){
if (strlen(ip)==0) return FALSE;
- return strcmp(ip, localv4) == 0 || strcmp(ip, localv6) == 0;
+ for (; addresses != NULL; addresses = addresses->next){
+ if (strcmp(ip, (const char*)addresses->data) == 0) return TRUE;
+ }
+ return FALSE;
}
/*
@@ -237,6 +240,7 @@ static void call_with_ice_with_default_candidate_not_stun(void){
char localip[LINPHONE_IPADDR_SIZE]={0};
char localip6[LINPHONE_IPADDR_SIZE]={0};
bool_t call_ok;
+ bctbx_list_t *local_addresses = linphone_fetch_local_addresses();
lp_config_set_int(linphone_core_get_config(marie->lc), "net", "dont_default_to_stun_candidates", 1);
linphone_core_set_firewall_policy(marie->lc, LinphonePolicyUseIce);
@@ -246,14 +250,15 @@ static void call_with_ice_with_default_candidate_not_stun(void){
call_ok = call(marie, pauline);
if (call_ok){
check_ice(marie, pauline, LinphoneIceStateHostConnection);
- BC_ASSERT_TRUE(is_matching_local_v4_or_v6(_linphone_call_get_local_desc(linphone_core_get_current_call(marie->lc))->addr, localip, localip6));
- BC_ASSERT_TRUE(is_matching_local_v4_or_v6(_linphone_call_get_local_desc(linphone_core_get_current_call(marie->lc))->streams[0].rtp_addr, localip, localip6));
- BC_ASSERT_TRUE(is_matching_local_v4_or_v6(_linphone_call_get_local_desc(linphone_core_get_current_call(marie->lc))->streams[0].rtp_addr, localip, localip6));
- BC_ASSERT_TRUE(is_matching_local_v4_or_v6(_linphone_call_get_result_desc(linphone_core_get_current_call(pauline->lc))->streams[0].rtp_addr, localip, localip6)
- || is_matching_local_v4_or_v6(_linphone_call_get_result_desc(linphone_core_get_current_call(pauline->lc))->addr, localip, localip6)
+ BC_ASSERT_TRUE(is_matching_a_local_address(_linphone_call_get_local_desc(linphone_core_get_current_call(marie->lc))->addr, local_addresses));
+ BC_ASSERT_TRUE(is_matching_a_local_address(_linphone_call_get_local_desc(linphone_core_get_current_call(marie->lc))->streams[0].rtp_addr, local_addresses));
+ BC_ASSERT_TRUE(is_matching_a_local_address(_linphone_call_get_local_desc(linphone_core_get_current_call(marie->lc))->streams[0].rtp_addr, local_addresses));
+ BC_ASSERT_TRUE(is_matching_a_local_address(_linphone_call_get_result_desc(linphone_core_get_current_call(pauline->lc))->streams[0].rtp_addr, local_addresses)
+ || is_matching_a_local_address(_linphone_call_get_result_desc(linphone_core_get_current_call(pauline->lc))->addr, local_addresses)
);
}
end_call(marie, pauline);
+ bctbx_list_free_with_data(local_addresses, bctbx_free);
linphone_core_manager_destroy(marie);
linphone_core_manager_destroy(pauline);
}
@@ -330,7 +335,9 @@ static void call_with_ice_ipv6(void) {
}
}
-/*ICE is not expected to work in this case, however this should not crash*/
+/*ICE is not expected to work in this case, however this should not crash.
+ Updated 08/01/2020: now ICE works also in the case of an INVITE without SDP.
+ */
static void call_with_ice_no_sdp(void){
LinphoneCoreManager* marie = linphone_core_manager_new("marie_rc");
LinphoneCoreManager* pauline = linphone_core_manager_new(transport_supported(LinphoneTransportTls) ? "pauline_rc" : "pauline_tcp_rc");
diff --git a/tester/call_single_tester.c b/tester/call_single_tester.c
index 9ebca886b555eac129d6b25e9dd92cf55fea38e5..2494159ab7f8ceff334c8e7e9ba757ee6d905da7 100644
--- a/tester/call_single_tester.c
+++ b/tester/call_single_tester.c
@@ -2857,7 +2857,7 @@ static void early_media_call_with_update_base(bool_t media_change){
BC_ASSERT_EQUAL(linphone_core_get_tone_manager_stats(pauline->lc)->number_of_stopRingtone, ringWithEarlyMedia ? 0 : 1, int, "%d");
BC_ASSERT_EQUAL(linphone_core_get_tone_manager_stats(marie->lc)->number_of_stopRingbackTone, 1, int, "%d");
- pauline_params = linphone_call_params_copy(linphone_call_get_current_params(pauline_call));
+ pauline_params = linphone_core_create_call_params(pauline->lc, pauline_call);
if (media_change) {
disable_all_audio_codecs_except_one(marie->lc,"pcma",-1);
@@ -3304,7 +3304,11 @@ void check_media_direction(LinphoneCoreManager* mgr, LinphoneCall *call, bctbx_l
}
switch (video_dir) {
case LinphoneMediaDirectionInactive:
- BC_ASSERT_LOWER((int)linphone_call_stats_get_upload_bandwidth(stats), 5, int, "%i");
+ if (stats){
+ BC_ASSERT_LOWER((int)linphone_call_stats_get_upload_bandwidth(stats), 5, int, "%i");
+ }else{
+ /* it is expected that there is no stats for an inactive stream.*/
+ }
break;
case LinphoneMediaDirectionSendOnly:
expected_recv_iframe = 0;
@@ -3319,7 +3323,7 @@ void check_media_direction(LinphoneCoreManager* mgr, LinphoneCall *call, bctbx_l
default:
break;
}
- linphone_call_stats_unref(stats);
+ if (stats) linphone_call_stats_unref(stats);
BC_ASSERT_TRUE(wait_for_list(lcs, &mgr->stat.number_of_IframeDecoded,current_recv_iframe + expected_recv_iframe,10000));
}
#endif
diff --git a/tester/call_video_tester.c b/tester/call_video_tester.c
index 442bf400be0ac934bfcbbac7fad17697578641cf..3a27ebdd77e9ed1062cd871b6497ac7c460d321b 100644
--- a/tester/call_video_tester.c
+++ b/tester/call_video_tester.c
@@ -998,7 +998,7 @@ static void _call_with_ice_video(LinphoneVideoPolicy caller_policy, LinphoneVide
bool_t video_added_by_caller, bool_t video_added_by_callee, bool_t video_removed_by_caller, bool_t video_removed_by_callee, bool_t video_only) {
LinphoneCoreManager *marie = linphone_core_manager_new("marie_rc");
LinphoneCoreManager* pauline = linphone_core_manager_new(transport_supported(LinphoneTransportTls) ? "pauline_rc" : "pauline_tcp_rc");
- unsigned int nb_audio_starts = 1, nb_video_starts = 1;
+ unsigned int nb_audio_starts = 1, nb_video_starts = 0;
const LinphoneCallParams *marie_remote_params;
const LinphoneCallParams *pauline_current_params;
@@ -1018,6 +1018,7 @@ static void _call_with_ice_video(LinphoneVideoPolicy caller_policy, LinphoneVide
if (video_only) {
linphone_core_enable_payload_type(marie->lc, linphone_core_find_payload_type(marie->lc, "PCMU", 8000, 1), FALSE); /* Disable PCMU */
linphone_core_enable_payload_type(marie->lc, linphone_core_find_payload_type(marie->lc, "PCMA", 8000, 1), TRUE); /* Enable PCMA */
+ nb_audio_starts = 0;
}
linphone_core_manager_wait_for_stun_resolution(marie);
@@ -1052,6 +1053,8 @@ static void _call_with_ice_video(LinphoneVideoPolicy caller_policy, LinphoneVide
if (pauline_current_params){
BC_ASSERT_TRUE(linphone_call_params_video_enabled(pauline_current_params) ==
(caller_policy.automatically_initiate && callee_policy.automatically_accept));
+ if (linphone_call_params_video_enabled(pauline_current_params))
+ nb_video_starts++;
}
/* Wait for ICE reINVITEs to complete. */
@@ -1064,6 +1067,7 @@ static void _call_with_ice_video(LinphoneVideoPolicy caller_policy, LinphoneVide
}
BC_ASSERT_TRUE(check_ice(pauline, marie, LinphoneIceStateHostConnection));
BC_ASSERT_TRUE(check_nb_media_starts(AUDIO_START, pauline, marie, nb_audio_starts, nb_audio_starts));
+ BC_ASSERT_TRUE(check_nb_media_starts(VIDEO_START, pauline, marie, nb_video_starts, nb_video_starts));
if (caller_policy.automatically_initiate && callee_policy.automatically_accept && (video_added_by_caller || video_added_by_callee)){
@@ -1094,7 +1098,6 @@ static void _call_with_ice_video(LinphoneVideoPolicy caller_policy, LinphoneVide
}
if (video_removed_by_caller || video_removed_by_callee) {
BC_ASSERT_TRUE(check_ice(pauline, marie, LinphoneIceStateHostConnection));
- nb_video_starts++;
BC_ASSERT_TRUE(check_nb_media_starts(VIDEO_START, pauline, marie, nb_video_starts, nb_video_starts));
}
diff --git a/tester/call_with_rtp_bundle.c b/tester/call_with_rtp_bundle.c
new file mode 100644
index 0000000000000000000000000000000000000000..8625491d6a78d587ec85c91ca95821c46b15126a
--- /dev/null
+++ b/tester/call_with_rtp_bundle.c
@@ -0,0 +1,212 @@
+/*
+ * Copyright (c) 2010-2019 Belledonne Communications SARL.
+ *
+ * This file is part of Liblinphone.
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include "linphone/core.h"
+#include "linphone/lpconfig.h"
+#include "liblinphone_tester.h"
+#include "tester_utils.h"
+#include "mediastreamer2/msutils.h"
+#include "belle-sip/sipstack.h"
+#include <bctoolbox/defs.h>
+
+static void check_rtp_bundle(LinphoneCall *call, bool_t should_be_active){
+ const LinphoneCallParams *remote_params = linphone_call_get_remote_params(call);
+ const LinphoneCallParams *current_params = linphone_call_get_current_params(call);
+ if (should_be_active){
+ BC_ASSERT_TRUE(linphone_call_params_rtp_bundle_enabled(remote_params));
+ BC_ASSERT_TRUE(linphone_call_params_rtp_bundle_enabled(current_params));
+ }else{
+ BC_ASSERT_FALSE(linphone_call_params_rtp_bundle_enabled(remote_params));
+ BC_ASSERT_FALSE(linphone_call_params_rtp_bundle_enabled(current_params));
+ }
+}
+
+static bool_t setup_dtls_srtp(LinphoneCoreManager *marie, LinphoneCoreManager *pauline){
+ if (!linphone_core_media_encryption_supported(marie->lc,LinphoneMediaEncryptionDTLS)){
+ BC_FAIL("SRTP-DTLS not supported.");
+ return FALSE;
+ }
+ linphone_core_set_media_encryption(marie->lc, LinphoneMediaEncryptionDTLS);
+ linphone_core_set_media_encryption(pauline->lc, LinphoneMediaEncryptionDTLS);
+ char *path = bc_tester_file("certificates-marie");
+ linphone_core_set_user_certificates_path(marie->lc, path);
+ bc_free(path);
+ path = bc_tester_file("certificates-pauline");
+ linphone_core_set_user_certificates_path(pauline->lc, path);
+ bc_free(path);
+ belle_sip_mkdir(linphone_core_get_user_certificates_path(marie->lc));
+ belle_sip_mkdir(linphone_core_get_user_certificates_path(pauline->lc));
+ return TRUE;
+}
+
+static void _simple_audio_call(bool_t with_dtls_srtp) {
+ LinphoneCoreManager* marie;
+ LinphoneCoreManager* pauline;
+ LinphoneCall *pauline_call, *marie_call;
+
+ marie = linphone_core_manager_new( "marie_rc");
+ pauline = linphone_core_manager_new(transport_supported(LinphoneTransportTls) ? "pauline_rc" : "pauline_tcp_rc");
+
+ linphone_core_enable_rtp_bundle(marie->lc, TRUE);
+
+ if (with_dtls_srtp){
+ setup_dtls_srtp(marie, pauline);
+ }
+
+ BC_ASSERT_TRUE(call(marie,pauline));
+ pauline_call=linphone_core_get_current_call(pauline->lc);
+ marie_call = linphone_core_get_current_call(marie->lc);
+
+ if (BC_ASSERT_PTR_NOT_NULL(pauline_call))
+ check_rtp_bundle(pauline_call, TRUE);
+
+ if (BC_ASSERT_PTR_NOT_NULL(marie_call))
+ check_rtp_bundle(marie_call, TRUE);
+
+ liblinphone_tester_check_rtcp(marie,pauline);
+ end_call(marie,pauline);
+ linphone_core_manager_destroy(pauline);
+ linphone_core_manager_destroy(marie);
+}
+
+static void simple_audio_call(void){
+ _simple_audio_call(FALSE);
+}
+
+static void simple_audio_call_with_srtp_dtls(void){
+ _simple_audio_call(TRUE);
+}
+
+typedef struct params{
+ bool_t with_ice;
+ bool_t with_dtls_srtp;
+} params_t;
+
+static void audio_video_call(const params_t *params) {
+ LinphoneCoreManager* marie;
+ LinphoneCoreManager* pauline;
+ LinphoneCall *pauline_call, *marie_call;
+ LinphoneVideoActivationPolicy *vpol = linphone_factory_create_video_activation_policy(linphone_factory_get());
+
+ marie = linphone_core_manager_new( "marie_rc");
+ pauline = linphone_core_manager_new(transport_supported(LinphoneTransportTls) ? "pauline_rc" : "pauline_tcp_rc");
+
+ linphone_core_enable_rtp_bundle(marie->lc, TRUE);
+
+ linphone_video_activation_policy_set_automatically_initiate(vpol, TRUE);
+ linphone_video_activation_policy_set_automatically_accept(vpol, TRUE);
+
+ linphone_core_enable_video_capture(marie->lc, TRUE);
+ linphone_core_enable_video_display(marie->lc, TRUE);
+ linphone_core_enable_video_capture(pauline->lc, TRUE);
+ linphone_core_enable_video_display(pauline->lc, TRUE);
+
+ linphone_core_set_preferred_video_size_by_name(marie->lc, "QVGA");
+ linphone_core_set_preferred_video_size_by_name(pauline->lc, "QVGA");
+
+ linphone_core_set_video_device(marie->lc, "Mire: Mire (synthetic moving picture)");
+ linphone_core_set_video_device(pauline->lc, "Mire: Mire (synthetic moving picture)");
+
+ if (params->with_ice){
+ /*enable ICE on both ends*/
+ LinphoneNatPolicy *pol;
+ pol = linphone_core_get_nat_policy(marie->lc);
+ linphone_nat_policy_enable_ice(pol, TRUE);
+ linphone_nat_policy_enable_stun(pol, TRUE);
+ linphone_core_set_nat_policy(marie->lc, pol);
+ pol = linphone_core_get_nat_policy(pauline->lc);
+ linphone_nat_policy_enable_ice(pol, TRUE);
+ linphone_nat_policy_enable_stun(pol, TRUE);
+ linphone_core_set_nat_policy(pauline->lc, pol);
+ }
+
+ if (params->with_dtls_srtp){
+ setup_dtls_srtp(marie, pauline);
+ }
+
+ linphone_core_set_video_activation_policy(marie->lc, vpol);
+ linphone_core_set_video_activation_policy(pauline->lc, vpol);
+ linphone_video_activation_policy_unref(vpol);
+
+ if (!BC_ASSERT_TRUE(call(marie,pauline))) goto end;
+ pauline_call=linphone_core_get_current_call(pauline->lc);
+ marie_call = linphone_core_get_current_call(marie->lc);
+
+ check_rtp_bundle(pauline_call, TRUE);
+ check_rtp_bundle(marie_call, TRUE);
+
+ BC_ASSERT_TRUE(linphone_call_params_video_enabled(linphone_call_get_current_params(pauline_call)));
+ BC_ASSERT_TRUE(linphone_call_params_video_enabled(linphone_call_get_current_params(marie_call)));
+
+ if (params->with_ice){
+ BC_ASSERT_TRUE(check_ice(marie, pauline, LinphoneIceStateHostConnection));
+ }
+
+ liblinphone_tester_check_rtcp(marie,pauline);
+ liblinphone_tester_set_next_video_frame_decoded_cb(pauline_call);
+ BC_ASSERT_TRUE(wait_for(marie->lc,pauline->lc,&pauline->stat.number_of_IframeDecoded,1));
+ liblinphone_tester_set_next_video_frame_decoded_cb(marie_call);
+ BC_ASSERT_TRUE(wait_for(marie->lc,pauline->lc,&marie->stat.number_of_IframeDecoded,1));
+
+ if (params->with_dtls_srtp){
+ BC_ASSERT_TRUE(linphone_call_params_get_media_encryption(linphone_call_get_current_params(pauline_call)) == LinphoneMediaEncryptionDTLS);
+ BC_ASSERT_TRUE(linphone_call_params_get_media_encryption(linphone_call_get_current_params(marie_call)) == LinphoneMediaEncryptionDTLS);
+ }
+
+
+ end_call(marie,pauline);
+
+end:
+ linphone_core_manager_destroy(pauline);
+ linphone_core_manager_destroy(marie);
+}
+
+static void simple_audio_video_call(void) {
+ params_t params = {0};
+ audio_video_call(¶ms);
+}
+
+static void audio_video_call_with_ice(void) {
+ params_t params = {0};
+ params.with_ice = TRUE;
+ audio_video_call(¶ms);
+}
+
+static void audio_video_call_with_ice_and_dtls_srtp(void) {
+ params_t params = {0};
+ params.with_ice = TRUE;
+ params.with_dtls_srtp = TRUE;
+ audio_video_call(¶ms);
+}
+
+static test_t call_with_rtp_bundle_tests[] = {
+ TEST_NO_TAG("Simple audio call", simple_audio_call),
+ TEST_NO_TAG("Simple audio call with DTLS-SRTP", simple_audio_call_with_srtp_dtls),
+ TEST_NO_TAG("Simple audio-video call", simple_audio_video_call),
+ TEST_NO_TAG("Audio-video call with ICE", audio_video_call_with_ice),
+ TEST_NO_TAG("Audio-video call with ICE and DTLS-SRTP", audio_video_call_with_ice_and_dtls_srtp)
+};
+
+test_suite_t call_with_rtp_bundle_test_suite = {"Call with RTP bundle", NULL, NULL, liblinphone_tester_before_each, liblinphone_tester_after_each,
+ sizeof(call_with_rtp_bundle_tests) / sizeof(call_with_rtp_bundle_tests[0]), call_with_rtp_bundle_tests};
+
+
diff --git a/tester/certificates/cn/cafile.pem b/tester/certificates/cn/cafile.pem
index 9f031906a56e59a5d7e3e239d3b0a1daa6d2921e..6bc2237bb092328e3005b1a91b69c2ec27840970 100644
--- a/tester/certificates/cn/cafile.pem
+++ b/tester/certificates/cn/cafile.pem
@@ -19,44 +19,44 @@ FUWGJhPnkrnklmBdVB0l7qXYjR5uf766HDkoDxuLhNifow3IYvsS+L2Y6puRQb9w
HLMDE29mBDl0WyoX3h0yR0EiAO15V9A7I10=
-----END CERTIFICATE-----
-USERTrust RSA Certification Authority (used for *.linphone.org certificates)
-=====================================
+Usertrust (2020) for *.linphone.org
-----BEGIN CERTIFICATE-----
-MIIF3jCCA8agAwIBAgIQAf1tMPyjylGoG7xkDjUDLTANBgkqhkiG9w0BAQwFADCB
+MIIF6TCCA9GgAwIBAgIQBeTcO5Q4qzuFl8umoZhQ4zANBgkqhkiG9w0BAQwFADCB
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+1hMwt6CaDixVBgBwAc/qOR2A24j3pSC4W/0xJmmPLQphgzpHphNULB7j7UTKvGof
+KA5R2d4On3XNDgOVyvnFqSot/kGkoUeuDcL5OWYzSlvhhChZbH2UF3bkRYKtcCD9
+0m9jqNf6oDP6N8v3smWe2lBvP+Sn845dWDKXcCMu5/3EFZucJ48y7RetWIExKREa
+m9T8bJUox04FB6b9HbwZ4ui3uRGKLXASUoWNjDNKD/yZkuBjcNqllEdjB+dYxzFf
+BT02Vf6Dsuimrdfp5gJ0iHRc2jTbkNJtUQoj1iM=
-----END CERTIFICATE-----
-AddTrust External Root (used for *.linphone.org certificates generated before 2020)
+
+AddTrust External Root used for *.linphone.org
======================
-----BEGIN CERTIFICATE-----
MIIENjCCAx6gAwIBAgIBATANBgkqhkiG9w0BAQUFADBvMQswCQYDVQQGEwJTRTEUMBIGA1UEChML
@@ -79,3 +79,5 @@ j7DYd7usQWxHYINRsPkyPef89iYTx4AWpb9a/IfPeHmJIZriTAcKhjW88t5RxNKWt9x+Tu5w/Rw5
e6cJDUCrat2PisP29owaQgVR1EX1n6diIWgVIEM8med8vSTYqZEXc4g/VhsxOBi0cQ+azcgOno4u
G+GMmIPLHzHxREzGBHNJdmAPx/i9F4BrLunMTA5amnkPIAou1Z5jJh5VkpTYghdae9C8x49OhgQ=
-----END CERTIFICATE-----
+
+
diff --git a/tester/group_chat_tester.c b/tester/group_chat_tester.c
index 6fb58024e48fb01cf6f4cece007a2a2c36ec23ff..d8d683f5f1d7447c0d40e61a4acc3a4d34401fb3 100644
--- a/tester/group_chat_tester.c
+++ b/tester/group_chat_tester.c
@@ -5368,11 +5368,8 @@ static void group_chat_room_join_one_to_one_chat_room_with_a_new_device_not_noti
coresList = bctbx_list_concat(coresList, tmpCoresList);
linphone_core_manager_start(pauline, TRUE);
- //wait for first notify to be received by pauline
- wait_for_list(coresList, NULL, 0, 1000);
-
// Marie2 gets the one-to-one chat room with Pauline
- paulineCr = check_has_chat_room_client_side(coresList, pauline, &initialPaulineStats, confAddr, initialSubject, 1, FALSE);
+ paulineCr = check_creation_chat_room_client_side(coresList, pauline, &initialPaulineStats, confAddr, initialSubject, 1, FALSE);
LinphoneAddress *marieAddress = linphone_address_new(linphone_core_get_identity(marie2->lc));
LinphoneParticipant *marieParticipant = linphone_chat_room_find_participant(paulineCr, marieAddress);
BC_ASSERT_EQUAL(bctbx_list_size(linphone_participant_get_devices (marieParticipant)), 1, int, "%i");
@@ -5385,14 +5382,11 @@ static void group_chat_room_join_one_to_one_chat_room_with_a_new_device_not_noti
tmpCoresList = init_core_for_conference(tmpCoresManagerList);
bctbx_list_free(tmpCoresManagerList);
coresList = bctbx_list_concat(coresList, tmpCoresList);
+ initialPaulineStats = pauline->stat;
linphone_core_manager_start(pauline, TRUE);
-
- //wait for first notify to be received by pauline
- wait_for_list(coresList, NULL, 0, 1000);
-
// Marie2 gets the one-to-one chat room with Pauline
- paulineCr = check_has_chat_room_client_side(coresList, pauline, &initialPaulineStats, confAddr, initialSubject, 1, FALSE);
+ paulineCr = check_creation_chat_room_client_side(coresList, pauline, &initialPaulineStats, confAddr, initialSubject, 1, FALSE);
marieParticipant = linphone_chat_room_find_participant(paulineCr, marieAddress);
BC_ASSERT_EQUAL(bctbx_list_size(linphone_participant_get_devices (marieParticipant)), 1, int, "%i");
BC_ASSERT_EQUAL(linphone_chat_room_get_history_events_size(paulineCr), initialPaulineEvent, int, "%i");
@@ -5487,7 +5481,12 @@ static void subscribe_test_after_set_chat_database_path(void) {
linphone_core_cbs_set_chat_room_state_changed(cbs, core_chat_room_state_changed);
configure_core_for_callbacks(pauline, cbs);
linphone_core_cbs_unref(cbs);
+ initialPaulineStats = pauline->stat;
linphone_core_manager_start(pauline, TRUE);
+
+ /* Since pauline has unregistered (in linphone_core_manager_reinit(), the conference server will INVITE it again in the chatroom.
+ * Wait for this event before doing next steps, otherwise the subject changed notification could be masked. */
+ paulineCr = check_creation_chat_room_client_side(coresList, pauline, &initialPaulineStats, confAddr, initialSubject, 2, FALSE);
LinphoneAddress *paulineAddress = linphone_address_clone(linphone_proxy_config_get_contact(linphone_core_get_default_proxy_config(pauline->lc)));
paulineCr = linphone_core_find_chat_room(pauline->lc, confAddr, paulineAddress);
diff --git a/tester/liblinphone_tester.c b/tester/liblinphone_tester.c
index b70b96ea00d70e31a49afc94ad333402881c7b13..654717834778ef4e9a6cdceecc11a002a7ec6ea8 100644
--- a/tester/liblinphone_tester.c
+++ b/tester/liblinphone_tester.c
@@ -368,6 +368,7 @@ void liblinphone_tester_add_suites() {
bc_tester_add_suite(&vcard_test_suite);
#endif
bc_tester_add_suite(&utils_test_suite);
+ bc_tester_add_suite(&call_with_rtp_bundle_test_suite);
}
void liblinphone_tester_init(void(*ftester_printf)(int level, const char *fmt, va_list args)) {
diff --git a/tester/liblinphone_tester.h b/tester/liblinphone_tester.h
index ec0ec34a4595f4d3907c04b7298fe121a9197819..094fd13693176e0c073353c5aa449bceb93f1a8a 100644
--- a/tester/liblinphone_tester.h
+++ b/tester/liblinphone_tester.h
@@ -88,6 +88,7 @@ extern test_suite_t video_test_suite;
extern test_suite_t call_recovery_test_suite;
extern test_suite_t call_with_ice_test_suite;
extern test_suite_t call_secure_test_suite;
+extern test_suite_t call_with_rtp_bundle_test_suite;
#ifdef VCARD_ENABLED
extern test_suite_t vcard_test_suite;
diff --git a/tester/message_tester.c b/tester/message_tester.c
index 02f748c4bd98acbcc9a1d72bfe675e59daf739cc..7f30ecd8f8abdd8604966078d0913f01124d7285 100644
--- a/tester/message_tester.c
+++ b/tester/message_tester.c
@@ -795,14 +795,15 @@ static void file_transfer_2_messages_simultaneously(void) {
LinphoneChatMessage *recvMsg2 = marie->stat.last_received_chat_message;
BC_ASSERT_EQUAL((unsigned int)bctbx_list_size(linphone_core_get_chat_rooms(marie->lc)), 1, unsigned int, "%u");
if (bctbx_list_size(linphone_core_get_chat_rooms(marie->lc)) != 1) {
- char * buf = ms_strdup_printf("Found %d rooms instead of 1: ", bctbx_list_size(linphone_core_get_chat_rooms(marie->lc)));
+ char * buf = ms_strdup_printf("Found %d rooms instead of 1: ", (int)bctbx_list_size(linphone_core_get_chat_rooms(marie->lc)));
const bctbx_list_t *it = linphone_core_get_chat_rooms(marie->lc);
while (it) {
const LinphoneAddress * peer = linphone_chat_room_get_peer_address(it->data);
- buf = ms_strcat_printf("%s, ", linphone_address_get_username(peer));
+ buf = ms_strcat_printf(buf, "%s, ", linphone_address_get_username(peer));
it = it->next;
}
ms_error("%s", buf);
+ ms_free(buf);
}
cbs = linphone_chat_message_get_callbacks(recvMsg);
diff --git a/tester/tester.c b/tester/tester.c
index 4fd6edee5d71f66bbec9ed62e24834ce5b387b44..b8dbbe47ee1b36e7ce232ff6500317aa129e490b 100644
--- a/tester/tester.c
+++ b/tester/tester.c
@@ -1498,9 +1498,11 @@ void call_stats_updated(LinphoneCore *lc, LinphoneCall *call, const LinphoneCall
counters->number_of_rtcp_received_via_mux++;
}
rtcp_received(counters, _linphone_call_stats_get_received_rtcp(lstats));
+ BC_ASSERT_TRUE(_linphone_call_stats_has_received_rtcp(lstats));
}
if (updated & LINPHONE_CALL_STATS_SENT_RTCP_UPDATE ) {
counters->number_of_rtcp_sent++;
+ BC_ASSERT_TRUE(_linphone_call_stats_has_sent_rtcp(lstats));
}
if (updated & LINPHONE_CALL_STATS_PERIODICAL_UPDATE ) {
const int tab_size = sizeof counters->audio_download_bandwidth / sizeof(int);
@@ -1861,7 +1863,6 @@ bool_t call_with_params2(LinphoneCoreManager* caller_mgr
/*wait ice re-invite*/
if (linphone_core_get_firewall_policy(caller_mgr->lc) == LinphonePolicyUseIce
&& linphone_core_get_firewall_policy(callee_mgr->lc) == LinphonePolicyUseIce
- && !linphone_core_sdp_200_ack_enabled(caller_mgr->lc) /*ice does not work with sdp less invite*/
&& lp_config_get_int(linphone_core_get_config(callee_mgr->lc), "sip", "update_call_when_ice_completed", TRUE)
&& lp_config_get_int(linphone_core_get_config(callee_mgr->lc), "sip", "update_call_when_ice_completed", TRUE)
&& linphone_core_get_media_encryption(caller_mgr->lc) != LinphoneMediaEncryptionDTLS /*no ice-reinvite with DTLS*/) {