Prevent RTCP SR to be sent with bogus timestamp.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 451f8bfa9398e4227b5542580e6dc9658fcfc553..0a6ea5a7ec04524f33584262706cd11aa5a98a5b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -183,8 +183,13 @@ int32_t ModuleRtpRtcpImpl::Process() {
       set_rtt_ms(rtt_stats_->LastProcessedRtt());
   }
 
-  if (rtcp_sender_.TimeToSendRTCPReport())
-    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+  // For sending streams, make sure to not send a SR before media has been sent.
+  if (rtcp_sender_.TimeToSendRTCPReport()) {
+    RTCPSender::FeedbackState state = GetFeedbackState();
+    // Prevent send streams to send SR before any media has been sent.
+    if (!rtcp_sender_.Sending() || state.packets_sent > 0)
+      rtcp_sender_.SendRTCP(state, kRtcpReport);
+  }
 
   if (UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out