Hi all, is it a problem, when a RTP-stream during a call is replaced by another RTP-stream?
Let's assume, stream with SSRC1 is arriving at a UAC (100.11.12.13:12345). Now, the stream stops (without any SIP-signalling like Re-INVITEs, etc.) and a stream with SSRC2 is now arriving at 100.11.12.13:12345 instead of SSRC1. Is there a regulation defining to block a changing SSRC? Or is it satisfying that the stream is arriving at the same socket (defined in the SDP of the INVITE) like the one before? Thx in advance and br, Michael _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
