diff --git a/contrib/src/ffmpeg/rtp_ext_abs_send_time.patch b/contrib/src/ffmpeg/rtp_ext_abs_send_time.patch
new file mode 100644
index 0000000000000000000000000000000000000000..cf52cf972ae6ba49ac97f80455d5cba8515f23d0
--- /dev/null
+++ b/contrib/src/ffmpeg/rtp_ext_abs_send_time.patch
@@ -0,0 +1,71 @@
+diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
+index 63047beccc..d59ec3dc8c 100644
+--- a/libavformat/rtpenc.c
++++ b/libavformat/rtpenc.c
+@@ -28,6 +28,8 @@
+
+ #include "rtpenc.h"
+
++#define EXT_ABS_SEND_TIME
++
+ static const AVOption options[] = {
+ FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
+ { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
+@@ -146,7 +148,11 @@ static int rtp_write_header(AVFormatContext *s1)
+ s1->pb->max_packet_size);
+ } else
+ s1->packet_size = s1->pb->max_packet_size;
++#ifdef EXT_ABS_SEND_TIME
++ if (s1->packet_size <= 20) {
++#else
+ if (s1->packet_size <= 12) {
++#endif
+ av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
+ return AVERROR(EIO);
+ }
+@@ -154,7 +160,11 @@ static int rtp_write_header(AVFormatContext *s1)
+ if (!s->buf) {
+ return AVERROR(ENOMEM);
+ }
++#ifdef EXT_ABS_SEND_TIME
++ s->max_payload_size = s1->packet_size - 20;
++#else
+ s->max_payload_size = s1->packet_size - 12;
++#endif
+
+ if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
+ avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
+@@ -332,16 +342,34 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
+ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
+ {
+ RTPMuxContext *s = s1->priv_data;
++ uint64_t ntp64_time;
++ uint32_t absoluteSendTime;
+
+ av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
+
+ /* build the RTP header */
++#ifdef EXT_ABS_SEND_TIME
++ avio_w8(s1->pb, RTP_VERSION << 6 | 0x10); // extention bit
++#else
+ avio_w8(s1->pb, RTP_VERSION << 6);
++#endif
+ avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
+ avio_wb16(s1->pb, s->seq);
+ avio_wb32(s1->pb, s->timestamp);
+ avio_wb32(s1->pb, s->ssrc);
+
++#ifdef EXT_ABS_SEND_TIME
++ avio_wb16(s1->pb, 0xBEDE); // magic word
++ avio_wb16(s1->pb, 0x0001); // length=1
++ avio_w8(s1->pb, 0x32); // ID=3 and lenght=2
++ ntp64_time = ff_get_formatted_ntp_time(ff_ntp_time());
++ absoluteSendTime = (uint32_t)((ntp64_time>> 14) & 0x00ffffff);
++ av_log(s1, AV_LOG_TRACE, "ntp64:%lu, abs_time:%u\n", ntp64_time, absoluteSendTime);
++ avio_w8(s1->pb, (uint8_t)(absoluteSendTime >> 16));
++ avio_w8(s1->pb, (uint8_t)(absoluteSendTime >> 8 & 0xff));
++ avio_w8(s1->pb, (uint8_t)(absoluteSendTime & 0xff));
++#endif
++
+ avio_write(s1->pb, buf1, len);
+ avio_flush(s1->pb);
diff --git a/contrib/src/ffmpeg/rules.mak b/contrib/src/ffmpeg/rules.mak
index a8cd10aab269a185eb14a1af305998eb2cd7c771..32e32b1b63f021b914324922a0738d7294ff49c6 100644
--- a/contrib/src/ffmpeg/rules.mak
+++ b/contrib/src/ffmpeg/rules.mak
@@ -325,6 +325,7 @@ ffmpeg: ffmpeg-$(FFMPEG_HASH).tar.gz
(cd $@-$(FFMPEG_HASH) && tar x $(if ${BATCH_MODE},,-v) --strip-components=1 -f ../$<)
$(APPLY) $(SRC)/ffmpeg/remove-mjpeg-log.patch
$(APPLY) $(SRC)/ffmpeg/change-RTCP-ratio.patch
+ $(APPLY) $(SRC)/ffmpeg/rtp_ext_abs_send_time.patch
$(UPDATE_AUTOCONFIG)
$(MOVE)
diff --git a/src/media/socket_pair.cpp b/src/media/socket_pair.cpp
index b0773132d7fcf950c7e0208e2065462027b905cc..5cebf77e9c88d4415e10b5fab6d7ddf7631c6bf8 100644
--- a/src/media/socket_pair.cpp
+++ b/src/media/socket_pair.cpp
@@ -493,14 +493,21 @@ SocketPair::readCallback(uint8_t* buf, int buf_size)
// SRTP decrypt
if (not fromRTCP and srtpContext_ and srtpContext_->srtp_in.aes) {
- uint32_t curentSendTS = buf[4] << 24 | buf[5] << 16 | buf[6] << 8 | buf[7];
- int32_t delay = 0;
+ int32_t gradient = 0;
+ int32_t deltaT = 0;
float abs = 0.0f;
+ bool res_parse = false;
+ bool res_delay = false;
+
+ res_parse = parse_RTP_ext(buf, &abs);
bool marker = (buf[1] & 0x80) >> 7;
- bool res = getOneWayDelayGradient2(abs, marker, &delay);
- if (res)
- rtpDelayCallback_(delay);
+ if(res_parse)
+ res_delay = getOneWayDelayGradient2(abs, marker, &gradient, &deltaT);
+
+ // rtpDelayCallback_ is not set for audio
+ if (rtpDelayCallback_ && res_delay)
+ rtpDelayCallback_(gradient);
auto err = ff_srtp_decrypt(&srtpContext_->srtp_in, buf, &len);
if(packetLossCallback_ and (buf[2] << 8 | buf[3]) != lastSeqNum_+1) {
@@ -667,7 +674,7 @@ SocketPair::getOneWayDelayGradient(uint32_t sendTS)
}
bool
-SocketPair::getOneWayDelayGradient2(float sendTS, bool marker, int32_t* gradient)
+SocketPair::getOneWayDelayGradient2(float sendTS, bool marker, int32_t* gradient, int32_t* deltaT)
{
// Keep only last packet of each frame
if (not marker) {
@@ -681,7 +688,7 @@ SocketPair::getOneWayDelayGradient2(float sendTS, bool marker, int32_t* gradient
return 0;
}
- uint32_t deltaS = (sendTS - lastSendTS_) * 1000; // milliseconds
+ int32_t deltaS = (sendTS - lastSendTS_) * 1000; // milliseconds
if(deltaS < 0)
deltaS += 64000;
lastSendTS_ = sendTS;
@@ -691,6 +698,7 @@ SocketPair::getOneWayDelayGradient2(float sendTS, bool marker, int32_t* gradient
lastReceiveTS_ = arrival_TS;
*gradient = deltaR - deltaS;
+ *deltaT = deltaR;
return true;
}
@@ -739,4 +747,22 @@ SocketPair::getOneWayDelayGradient3(uint32_t sendTS, int32_t* gradient)
return true;
}
+bool
+SocketPair::parse_RTP_ext(uint8_t* buf, float* abs)
+{
+ if(not(buf[0] & 0x10))
+ return false;
+
+ uint16_t magic_word = (buf[12] << 8) + buf[13];
+ if(magic_word != 0xBEDE)
+ return false;
+
+ uint8_t sec = buf[17] >> 2;
+ uint32_t fract = ((buf[17] & 0x3) << 16 | (buf[18] << 8) | buf[19]) << 14;
+ float milli = fract / pow(2,32);
+
+ *abs = sec + (milli);
+ return true;
+}
+
} // namespace jami
diff --git a/src/media/socket_pair.h b/src/media/socket_pair.h
index 6024aa9d139bb1bfac6839d73eae2cba4ef93666..0a4d6104bcab1b3e03568f70d0722fdd9c8497ac 100644
--- a/src/media/socket_pair.h
+++ b/src/media/socket_pair.h
@@ -203,8 +203,9 @@ class SocketPair {
std::function<void(void)> packetLossCallback_;
std::function<void(int)> rtpDelayCallback_;
int32_t getOneWayDelayGradient(uint32_t sendTS);
- bool getOneWayDelayGradient2(float sendTS, bool marker, int32_t* gradient);
+ bool getOneWayDelayGradient2(float sendTS, bool marker, int32_t* gradient, int32_t* deltaR);
bool getOneWayDelayGradient3(uint32_t sendTS, int32_t* gradient);
+ bool parse_RTP_ext(uint8_t* buf, float* abs);
std::list<rtcpRRHeader> listRtcpRRHeader_;
std::list<rtcpREMBHeader> listRtcpREMBHeader_;
@@ -220,8 +221,7 @@ class SocketPair {
std::chrono::steady_clock::time_point lastRR_time;
uint16_t lastSeqNum_ {0};
- uint32_t lastSendTS_ {0};
- bool lastMarker_ {false};
+ float lastSendTS_ {0.0f};
std::chrono::steady_clock::time_point lastReceiveTS_ {};
std::chrono::steady_clock::time_point arrival_TS {};
diff --git a/src/media/video/video_rtp_session.cpp b/src/media/video/video_rtp_session.cpp
index 88a1cc6674a2eff6b4f43f3d8af694c740db4959..17440c0478860d82e6ab9a73a0b4e8d3cad038e7 100644
--- a/src/media/video/video_rtp_session.cpp
+++ b/src/media/video/video_rtp_session.cpp
@@ -151,10 +151,6 @@ void VideoRtpSession::startSender()
rtcpCheckerThread_.start();
else if (not autoQuality and rtcpCheckerThread_.isRunning())
rtcpCheckerThread_.join();
-
- socketPair_->setRtpDelayCallback([&](int delay) {
- this->delayMonitor(delay);
- });
}
}
@@ -207,6 +203,8 @@ void VideoRtpSession::start(std::unique_ptr<IceSocket> rtp_sock,
else
socketPair_.reset(new SocketPair(getRemoteRtpUri().c_str(), receive_.addr.getPort()));
+ socketPair_->setRtpDelayCallback([&](int delay) {delayMonitor(delay);});
+
if (send_.crypto and receive_.crypto) {
socketPair_->createSRTP(receive_.crypto.getCryptoSuite().c_str(),
receive_.crypto.getSrtpKeyInfo().c_str(),