Implement RTP keepalive in native stack.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 759bc9c980a8e85204e20f860df583ededf194f3..b862c29de22eaeb0f3f623edb1c68cffd92d30fb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1289,4 +1289,23 @@ void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
+int64_t RTPSender::LastTimestampTimeMs() const {
+  rtc::CritScope lock(&send_critsect_);
+  return last_timestamp_time_ms_;
+}
+
+void RTPSender::SendKeepAlive(uint8_t payload_type) {
+  rtc::CritScope lock(&send_critsect_);
+  if (!ssrc_)
+    return;
+
+  std::unique_ptr<RtpPacketToSend> packet = AllocatePacket();
+  packet->SetMarker(true);
+  packet->SetPayloadType(payload_type);
+  packet->SetTimestamp(last_rtp_timestamp_);

[åsapersson, 2017/06/30 15:09:13]

Could last_rtp_timestamp_ be 0 here and if so ok?

[sprang_webrtc, 2017/06/30 16:21:00]

Hm, if the stream is started with media being sent, then yes it can be 0.
This can potentially lead to a problem since it seems that the bandwidth
estimator
is triggered for the packet, even though it get's discarded.

However the alternative is to not keep the connection open, which is a
problem...
Maybe we should use the timestamp offset logic from SendPadData() though, for
temporary video mute?

Should we also try to ignore these packets in the bwe?

+  packet->set_capture_time_ms(capture_time_ms_);

[åsapersson, 2017/06/30 15:09:13]

is locked needed for SendToNetwork?

And AllocatePacket?

[sprang_webrtc, 2017/06/30 16:21:00]

No, in fact it probably shouldn't. Missed cleaning this up.
Only needed for reading timestamps.

+  SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
+                RtpPacketSender::Priority::kLowPriority);
+}
+
 }  // namespace webrtc