Do not regenerate ssrc on SetSendingStatus

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 50885f457f2d9838a02682c653ae42c943cdc028..8445a256fe8ed20384e91965b3a3a3145e6c2f92 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -355,19 +355,6 @@ int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
     if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
       LOG(LS_WARNING) << "Failed to send RTCP BYE";
     }
-
-    collision_detected_ = false;
-
-    // Generate a new SSRC for the next "call" if false
-    rtp_sender_.SetSendingStatus(sending);
-
-    // Make sure that RTCP objects are aware of our SSRC (it could have changed
-    // Due to collision)
-    uint32_t SSRC = rtp_sender_.SSRC();
-    rtcp_sender_.SetSSRC(SSRC);
-    SetRtcpReceiverSsrcs(SSRC);
-
-    return 0;
   }
   return 0;
 }