From 8093d5442e4c365bfebc07abcf5fb653bd7a1d57 Mon Sep 17 00:00:00 2001 From: solenberg Date: Thu, 12 Nov 2015 06:02:30 -0800 Subject: [PATCH] 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} --- talk/media/webrtc/webrtcvoiceengine.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h index 33524dbf53..e480c13bef 100644 --- a/talk/media/webrtc/webrtcvoiceengine.h +++ b/talk/media/webrtc/webrtcvoiceengine.h @@ -315,9 +315,10 @@ 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 + uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; class WebRtcAudioSendStream; std::map send_streams_;