Change default SSRC for RTCP receiver reports to not collide with video.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 297 matching lines @@
   bool nack_enabled_ = false;
   bool playout_ = false;
   SendFlags desired_send_ = SEND_NOTHING;
   SendFlags send_ = SEND_NOTHING;
   webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
-  // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled
+  // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
-  uint32_t receiver_reports_ssrc_ = 1;
+  // and https://code.google.com/p/chromium/issues/detail?id=547661
+  uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;

[pthatcher1, 2015/11/12 20:46:40]

Sorry for being late to the bug.

Changing the RR SSRC to something other than 1 may cause RRs to not be received
if the SSRCs aren't signalled because the BundleFilter won't know what to do
with them.  I see there is a CL for updating the BundleFilter to let in all RRs,
regardless of SSRC=1, but that's not submitted yet.  Until it has been submitted
and rolled out for a while, I think the fix here to change the SSRC is risky.

 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<RtpHeaderExtension> send_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> receive_channels_;
   std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_;
   std::map<uint32_t, StreamParams> receive_stream_params_;
   // receive_channels_ can be read from WebRtc callback thread.  Access from
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_