Wire up transport sequence number / send time callbacks to webrtc via libjingle.

webrtc/video/call.cc

Index: webrtc/video/call.cc
diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index 2b2d5968559c1c28c8ad91a7585f80418ee4c424..e542c5040ebd6a609672ac105784b5dab13f5cb6 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -80,6 +80,8 @@ class Call : public webrtc::Call, public PacketReceiver {
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
   void SignalNetworkState(NetworkState state) override;
 
+  void OnSentPacket(const SentPacket& packet_sent) override;
+
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
@@ -422,6 +424,10 @@ void Call::SignalNetworkState(NetworkState state) {
   }
 }
 
+void Call::OnSentPacket(const SentPacket& packet_sent) {
+  channel_group_->OnSentPacket(packet_sent);
+}
+
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (config_.voice_engine == nullptr || sync_group.empty())