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

talk/media/webrtc/webrtcvoiceengine.h

Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 33524dbf53a9cdc7bcf71913204894e29ed84608..dd7c46c6be2f52e261d0085adfca24750f2ccf39 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -315,9 +315,11 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   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
+  // This should really be fixed in the RTP/RTCP module.

[pbos-webrtc, 2015/11/12 12:48:12]

Why in the RTP/RTCP module?

+  uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;