SSRC_ Streaming Media protocol in RTP

In the RTP protocol, the source of the ssrc,synchronization source is defined as the RTP packet stream, and the ssrc identifier of the 32-bit value in the RTP header is identified so that it does not depend on the network address. Usually the change of microphone, audio interface, camera, video interface will lead to SSRC changes.
In Opal and OpenH323, when the ssrc of the RTP stream in the same session is changed, the RTP packet is discarded, which causes the media stream of the audio or video to be interrupted and cannot be restored, and the default value should be changed to allow the ssrc of the RTP stream to change. The code is in Rtp.cxx, as follows:

     if (ignoreothersources     && frame. Getsyncsource ()!= Syncsourcein) {
      ptrace (2, "Rtp\tpacket from ssrc=" << frame. Getsyncsource ()
             << "ignored, expecting ssrc=" << Syncsourcein);
        return e_ignorepacket; Non fatal error, just ignore 
    }

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You need to set the ignoreothersources to False.