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Commit 005c28b9 authored by jehan's avatar jehan
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remove symmetric ftp handling for stun packets because it’s now managed by rtp

parent a4ad76a7
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Showing with 132 additions and 71 deletions
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 = {
......
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