Change default SSRC for RTCP receiver reports to not collide with video.
BUG=chromium:547661
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1438183002
Cr-Commit-Position: refs/heads/master@{#10621}
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 33524db..e480c13 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -315,9 +315,10 @@
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;
class WebRtcAudioSendStream;
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
