Skip to content

GitLab

Commit 005c28b9 authored by jehan's avatar jehan
Browse files

remove symmetric ftp handling for stun packets because it’s now managed by rtp

parent a4ad76a7
Hide whitespace changes
Inline Side-by-side
src/crypto/dtls_srtp.c
View file @ 005c28b9
......@@ -259,7 +259,6 @@ static bool_t ms_dtls_srtp_process_dtls_packet(mblk_t *msg, MSDtlsSrtpContext *c
DtlsRawPacket *incoming_dtls_packet;
RtpSession *rtp_session = ctx->stream_sessions->rtp_session;
OrtpStream *ortp_stream = is_rtp?&rtp_session->rtp.gs:&rtp_session->rtcp.gs;
incoming_dtls_packet = (DtlsRawPacket *)ms_malloc0(sizeof(DtlsRawPacket));
//DtlsRawPacket *incoming_dtls_packet = (DtlsRawPacket *)ms_malloc0(sizeof(DtlsRawPacket));
incoming_dtls_packet->next=NULL;
......@@ -268,17 +267,7 @@ static bool_t ms_dtls_srtp_process_dtls_packet(mblk_t *msg, MSDtlsSrtpContext *c
memcpy(incoming_dtls_packet->data, msg->b_rptr, msgLength);
/*required by webrtc in server case when ice is not completed yet*/
if (!rtp_session->use_connect){
struct sockaddr *addr = NULL;
socklen_t addrlen;
addr = (struct sockaddr *)&msg->net_addr;
addrlen = msg->net_addrlen;
if (ortp_stream->socket>0 && rtp_session->symmetric_rtp){
/* store the sender rtp address to do symmetric DTLS */
memcpy(&ortp_stream->rem_addr,addr,addrlen);
ortp_stream->rem_addrlen=addrlen;
}
}
rtp_session_update_remote_sock_addr(rtp_session, msg,is_rtp,FALSE);
ms_message("DTLS Receive %s packet len %d sessions: %p rtp session %p ssl state is %x", is_rtp==TRUE?"RTP":"RTCP", (int)msgLength, ctx->stream_sessions, ctx->stream_sessions->rtp_session, ssl->state);
......
src/otherfilters/msrtp.c
View file @ 005c28b9
......@@ -35,7 +35,8 @@ struct SenderData {
uint32_t tsoff;
uint32_t last_ts;
int64_t last_sent_time;
int64_t last_stun_sent_time;
int64_t last_rtp_stun_sent_time;
int64_t last_rtcp_stun_sent_time;
uint32_t skip_until;
int rate;
int dtmf_duration;
......@@ -55,7 +56,7 @@ struct SenderData {
typedef struct SenderData SenderData;
/* Send dummy STUN packet to open NAT ports ASAP. */
static void send_stun_packet(RtpSession *s)
static void send_stun_packet(RtpSession *s, bool_t enable_rtp, bool_t enable_rtcp)
{
StunMessage msg;
mblk_t *mp;
......@@ -69,11 +70,20 @@ static void send_stun_packet(RtpSession *s)
stunBuildReqSimple(&msg, NULL, FALSE, FALSE, 1);
len = stunEncodeMessage(&msg, buf, len, NULL);
if (len > 0) {
mp = allocb(len, BPRI_MED);
memcpy(mp->b_wptr, buf, len);
mp->b_wptr += len;
ms_message("Stun packet sent for session [%p]",s);
rtp_session_sendm_with_ts(s, mp, 0);
if (enable_rtp) {
mp = allocb(len, BPRI_MED);
memcpy(mp->b_wptr, buf, len);
mp->b_wptr += len;
ms_message("Stun packet sent for session [%p]",s);
rtp_session_sendm_with_ts(s, mp, 0);
}
if (enable_rtcp) {
mp = allocb(len, BPRI_MED);
memcpy(mp->b_wptr, buf, len);
mp->b_wptr += len;
ms_message("Stun packet sent on rtcp for session [%p]",s);
rtp_session_rtcp_sendm_raw(s,mp);
}
}
}
......@@ -97,7 +107,8 @@ static void sender_init(MSFilter * f)
d->relay_session_id_size=0;
d->last_rsi_time=0;
d->last_sent_time=-1;
d->last_stun_sent_time = -1;
d->last_rtp_stun_sent_time = -1;
d->last_rtcp_stun_sent_time = -1;
d->last_ts=0;
d->use_task= tmp ? (!!atoi(tmp)) : FALSE;
if (d->use_task) ms_message("MSRtpSend will use tasks to send out packet at the beginning of ticks.");
......@@ -146,7 +157,7 @@ static int sender_set_session(MSFilter * f, void *arg)
d->rate = pt->clock_rate;
d->dtmf_duration=(default_dtmf_duration_ms*d->rate)/1000;
d->dtmf_ts_step=(20*d->rate)/1000;
send_stun_packet(s);
send_stun_packet(s,TRUE,TRUE);
} else {
ms_warning("Sending undefined payload type ?");
}
......@@ -353,11 +364,17 @@ static int send_dtmf(MSFilter * f, uint32_t timestamp_start)
static void check_stun_sending(MSFilter *f) {
SenderData *d = (SenderData *) f->data;
RtpSession *s = d->session;
if ((d->last_stun_sent_time == -1) || ((f->ticker->time- d->last_sent_time>2000) /*no need to send stun packets if media sent during last 2s*/
&& (f->ticker->time - d->last_stun_sent_time) >= 500)) {
d->last_stun_sent_time = f->ticker->time;
send_stun_packet(s);
if ((d->last_rtp_stun_sent_time == -1) || ((f->ticker->time- d->last_sent_time>2000) /*no need to send stun packets if media sent during last 2s*/
&& (f->ticker->time - d->last_rtp_stun_sent_time) >= 500)) {
d->last_rtp_stun_sent_time = f->ticker->time;
send_stun_packet(s,TRUE,FALSE);
}
if ( rtp_session_rtcp_enabled(s) && (d->last_rtcp_stun_sent_time == -1
|| (rtp_session_get_stats(s)->recv_rtcp_packets == 0 /*no need to send stun packets if rtcp packet already received*/
&& (f->ticker->time - d->last_rtcp_stun_sent_time) >= 500))) {
d->last_rtcp_stun_sent_time = f->ticker->time;
send_stun_packet(s,FALSE,TRUE);
}
}
......
src/voip/mediastream.c
View file @ 005c28b9
......@@ -257,16 +257,7 @@ bool_t ms_is_ipv6(const char *remote) {
}
bool_t ms_is_multicast_addr(const struct sockaddr *addr) {
switch (addr->sa_family) {
case AF_INET:
return IN_MULTICAST(ntohl(((struct sockaddr_in *) addr)->sin_addr.s_addr));
case AF_INET6:
return IN6_IS_ADDR_MULTICAST(&(((struct sockaddr_in6 *) addr)->sin6_addr));
default:
return FALSE;
}
return ortp_is_multicast_addr(addr);
}
bool_t ms_is_multicast(const char *address) {
......@@ -331,18 +322,8 @@ void media_stream_iterate(MediaStream *stream){
ms_message("%s_stream_iterate[%p], local statistics available:"
"\n\tLocal current jitter buffer size: %5.1fms",
media_stream_type_str(stream), stream, rtp_session_get_jitter_stats(stream->sessions.rtp_session)->jitter_buffer_size_ms);
} else if (evt==ORTP_EVENT_STUN_PACKET_RECEIVED){
if (stream->ice_check_list) {
ice_handle_stun_packet(stream->ice_check_list,stream->sessions.rtp_session,ortp_event_get_data(ev));
} else if (rtp_session_get_symmetric_rtp(stream->sessions.rtp_session)){
/*try to know if we can trust stun packets for symetric rtp*/
rtp_stats_t stats;
media_stream_get_local_rtp_stats(stream, &stats);
if (stats.packet_recv == 0 && !ms_is_multicast_addr((const struct sockaddr *)&stream->sessions.rtp_session->rtp.gs.rem_addr)) {
memcpy(&stream->sessions.rtp_session->rtp.gs.rem_addr,&ortp_event_get_data(ev)->source_addr,ortp_event_get_data(ev)->source_addrlen);
ms_message("stun packet received but no rtp yet for stream [%p], switching rtp destination address",stream);
}
}
} else if (evt==ORTP_EVENT_STUN_PACKET_RECEIVED && stream->ice_check_list){
ice_handle_stun_packet(stream->ice_check_list,stream->sessions.rtp_session,ortp_event_get_data(ev));
} else if ((evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED) || (evt == ORTP_EVENT_DTLS_ENCRYPTION_CHANGED)) {
ms_message("%s_stream_iterate[%p]: is %s ",media_stream_type_str(stream) , stream, media_stream_secured(stream) ? "encrypted" : "not encrypted");
}
......
src/voip/qosanalyzer.c
View file @ 005c28b9
......@@ -172,7 +172,7 @@ static bool_t simple_analyzer_process_rtcp(MSQosAnalyzer *objbase, mblk_t *rtcp)
if (pt!=NULL) obj->clockrate=pt->clock_rate;
else return FALSE;
}
if (ortp_loss_rate_estimator_process_report_block(objbase->lre,&obj->session->rtp,rb)){
if (ortp_loss_rate_estimator_process_report_block(objbase->lre,obj->session,rb)){
cur->lost_percentage=ortp_loss_rate_estimator_get_value(objbase->lre);
cur->int_jitter=1000.0f*(float)report_block_get_interarrival_jitter(rb)/(float)obj->clockrate;
cur->rt_prop=rtp_session_get_round_trip_propagation(obj->session);
......@@ -350,7 +350,7 @@ static bool_t stateful_analyzer_process_rtcp(MSQosAnalyzer *objbase, mblk_t *rtc
if (rb && report_block_get_ssrc(rb)==rtp_session_get_send_ssrc(obj->session)){
if (ortp_loss_rate_estimator_process_report_block(objbase->lre,&obj->session->rtp,rb)){
if (ortp_loss_rate_estimator_process_report_block(objbase->lre,obj->session,rb)){
int i;
float loss_rate = ortp_loss_rate_estimator_get_value(objbase->lre);
float up_bw = stateful_qos_analyzer_upload_bandwidth(obj,report_block_get_high_ext_seq(rb));
......
src/voip/qualityindicator.c
View file @ 005c28b9
......@@ -139,7 +139,7 @@ void ms_quality_indicator_update_from_feedback(MSQualityIndicator *qi, mblk_t *r
float rt_prop=rtp_session_get_round_trip_propagation(qi->session);
bool_t new_value;
new_value=ortp_loss_rate_estimator_process_report_block(qi->lr_estimator,&qi->session->rtp,rb);
new_value=ortp_loss_rate_estimator_process_report_block(qi->lr_estimator,qi->session,rb);
loss_rate=ortp_loss_rate_estimator_get_value(qi->lr_estimator);
qi->remote_rating=compute_rating(loss_rate/100.0f,inter_jitter,0,rt_prop);
qi->remote_lq_rating=compute_lq_rating(loss_rate/100.0f,inter_jitter,0);
......
tester/mediastreamer2_adaptive_tester.c
View file @ 005c28b9
......@@ -305,7 +305,7 @@ static void event_queue_cb(MediaStream *ms, void *user_pointer) {
rb=rtcp_RR_get_report_block(evd->packet,0);
}
if (rb&&ortp_loss_rate_estimator_process_report_block(ctx->estimator,&ms->sessions.rtp_session->rtp,rb)){
if (rb&&ortp_loss_rate_estimator_process_report_block(ctx->estimator,ms->sessions.rtp_session,rb)){
float diff = fabs(ortp_loss_rate_estimator_get_value(ctx->estimator) - ctx->loss_rate);
BC_ASSERT_TRUE(diff >= 0);
BC_ASSERT_TRUE(diff <= 10);
......
tester/mediastreamer2_audio_stream_tester.c
View file @ 005c28b9
......@@ -117,12 +117,19 @@ static void event_queue_cb(MediaStream *ms, void *user_pointer) {
}
}
static void basic_audio_stream_base( const char* marielle_local_ip
, int marielle_local_rtp_port
, int marielle_local_rtcp_port
, const char* margaux_local_ip
, int margaux_local_rtp_port
, int margaux_local_rtcp_port) {
static void basic_audio_stream_base_2( const char* marielle_local_ip
, const char* marielle_remote_ip
, int marielle_local_rtp_port
, int marielle_remote_rtp_port
, int marielle_local_rtcp_port
, int marielle_remote_rtcp_port
, const char* margaux_local_ip
, const char* margaux_remote_ip
, int margaux_local_rtp_port
, int margaux_remote_rtp_port
, int margaux_local_rtcp_port
, int margaux_remote_rtcp_port
, int lost_percentage) {
AudioStream * marielle = audio_stream_new2 (marielle_local_ip, marielle_local_rtp_port, marielle_local_rtcp_port);
stats_t marielle_stats;
AudioStream * margaux = audio_stream_new2 (margaux_local_ip, margaux_local_rtp_port,margaux_local_rtcp_port);
......@@ -131,9 +138,13 @@ static void basic_audio_stream_base( const char* marielle_local_ip
char* hello_file = bc_tester_res(HELLO_8K_1S_FILE);
char* recorded_file = bc_tester_file(RECORDED_8K_1S_FILE);
int marielle_rtp_sent=0;
const jitter_stats_t *jitter_stats;
rtp_session_set_multicast_loopback(marielle->ms.sessions.rtp_session,TRUE);
rtp_session_set_multicast_loopback(margaux->ms.sessions.rtp_session,TRUE);
rtp_session_set_rtcp_report_interval(marielle->ms.sessions.rtp_session, 1000);
rtp_session_set_rtcp_report_interval(margaux->ms.sessions.rtp_session, 1000);
reset_stats(&marielle_stats);
reset_stats(&margaux_stats);
......@@ -142,10 +153,10 @@ static void basic_audio_stream_base( const char* marielle_local_ip
BC_ASSERT_EQUAL(audio_stream_start_full(margaux
, profile
, ms_is_multicast(margaux_local_ip)?margaux_local_ip:marielle_local_ip
, ms_is_multicast(margaux_local_ip)?margaux_local_rtp_port:marielle_local_rtp_port
, marielle_local_ip
, marielle_local_rtcp_port
, ms_is_multicast(margaux_local_ip)?margaux_local_ip:margaux_remote_ip
, ms_is_multicast(margaux_local_ip)?margaux_local_rtp_port:margaux_remote_rtp_port
, margaux_remote_ip
, margaux_remote_rtcp_port
, 0
, 50
, NULL
......@@ -157,10 +168,10 @@ static void basic_audio_stream_base( const char* marielle_local_ip
BC_ASSERT_EQUAL(audio_stream_start_full(marielle
, profile
, margaux_local_ip
, margaux_local_rtp_port
, margaux_local_ip
, margaux_local_rtcp_port
, marielle_remote_ip
, marielle_remote_rtp_port
, marielle_remote_ip
, marielle_remote_rtcp_port
, 0
, 50
, hello_file
......@@ -177,10 +188,17 @@ static void basic_audio_stream_base( const char* marielle_local_ip
audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);
marielle_rtp_sent = marielle_stats.rtp.sent;
jitter_stats = rtp_session_get_jitter_stats(margaux->ms.sessions.rtp_session);
if (rtp_session_rtcp_enabled(marielle->ms.sessions.rtp_session) && rtp_session_rtcp_enabled(margaux->ms.sessions.rtp_session)) {
BC_ASSERT_TRUE(rtp_session_get_round_trip_propagation(marielle->ms.sessions.rtp_session)>0);
BC_ASSERT_TRUE(rtp_session_get_stats(marielle->ms.sessions.rtp_session)->recv_rtcp_packets >0);
}
audio_stream_stop(marielle);
/* No packet loss is assumed */
wait_for_until(&margaux->ms,NULL,(int*)&margaux_stats.rtp.hw_recv,marielle_rtp_sent,2500);
BC_ASSERT_TRUE(wait_for_until(&margaux->ms,NULL,(int*)&margaux_stats.rtp.hw_recv,marielle_rtp_sent*(100-lost_percentage)/100,2500));
audio_stream_stop(margaux);
......@@ -189,7 +207,27 @@ static void basic_audio_stream_base( const char* marielle_local_ip
free(hello_file);
rtp_profile_destroy(profile);
}
static void basic_audio_stream_base( const char* marielle_local_ip
, int marielle_local_rtp_port
, int marielle_local_rtcp_port
, const char* margaux_local_ip
, int margaux_local_rtp_port
, int margaux_local_rtcp_port) {
basic_audio_stream_base_2( marielle_local_ip
, margaux_local_ip
, marielle_local_rtp_port
, margaux_local_rtp_port
, marielle_local_rtcp_port
, margaux_local_rtcp_port
, margaux_local_ip
, marielle_local_ip
, margaux_local_rtp_port
, marielle_local_rtp_port
, margaux_local_rtcp_port
, marielle_local_rtcp_port
, 0);
}
static void basic_audio_stream(void) {
basic_audio_stream_base(MARIELLE_IP,MARIELLE_RTP_PORT,MARIELLE_RTCP_PORT
,MARGAUX_IP, MARGAUX_RTP_PORT, MARGAUX_RTCP_PORT);
......@@ -551,6 +589,40 @@ static void audio_stream_dtmf(int codec_payload, int initial_bitrate,int target_
}
#endif
static void symetric_rtp_with_wrong_addr(void) {
basic_audio_stream_base_2( MARIELLE_IP
, "10.10.10.10" /*dummy ip*/
, MARIELLE_RTP_PORT
, MARGAUX_RTP_PORT
, MARIELLE_RTCP_PORT
, MARGAUX_RTCP_PORT
, MARGAUX_IP
, MARIELLE_IP
, MARGAUX_RTP_PORT
, MARIELLE_RTP_PORT
, MARGAUX_RTCP_PORT
, MARIELLE_RTCP_PORT
,5);
}
static void symetric_rtp_with_wrong_rtcp_port(void) {
basic_audio_stream_base_2( MARIELLE_IP
, MARGAUX_IP
, MARIELLE_RTP_PORT
, MARGAUX_RTP_PORT
, MARIELLE_RTCP_PORT
, MARGAUX_RTCP_PORT
, MARGAUX_IP
, MARIELLE_IP
, MARGAUX_RTP_PORT
, MARIELLE_RTP_PORT
, MARGAUX_RTCP_PORT
, MARIELLE_RTCP_PORT +10 /*dummy port*/
,5);
}
static test_t tests[] = {
{ "Basic audio stream", basic_audio_stream },
......@@ -563,7 +635,9 @@ static test_t tests[] = {
{ "Encrypted audio stream, encryption mandatory", encrypted_audio_stream_encryption_mandatory },
{ "Encrypted audio stream with key change + encryption mandatory", encrypted_audio_stream_with_key_change_encryption_mandatory},
{ "Codec change for audio stream", codec_change_for_audio_stream },
{ "TMMBR feedback for audio stream", tmmbr_feedback_for_audio_stream }
{ "TMMBR feedback for audio stream", tmmbr_feedback_for_audio_stream },
{ "Symetric rtp with wrong address", symetric_rtp_with_wrong_addr },
{ "Symetric rtp with wrong rtcp port", symetric_rtp_with_wrong_rtcp_port },
};
test_suite_t audio_stream_test_suite = {
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment