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SIP service does not redirect RTP to new remote port after re-INVITE changes m= line #661

@ArdentTeammate

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@ArdentTeammate

Carrier sent a re-INVITE mid-call that changed the remote RTP port from 25540 to 25446 (codec offer changed PCMU → G.722). LiveKit sent a 200 OK with its own SDP unchanged, but continued transmitting RTP to the old port for the remainder of the call. Per RFC 3264 §8, the answerer must send media to the new port in the updated offer.

SCL_VGXD6nQxzwks_sip_trace.txt

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