Dear community
I am running Asterisk 1.4.21.2 and have the following problem:
I have 2 SIP clients in a call with RTP not going through Asterisk due
to canreinvite=yes.
Everything is fine, until one of the SIP clients chooses to change RTP endpoint, and sends a new INVITE with new SDP info to Asterisk
Asterisk responds with a 200 OK containing the SDP info of the other client, so far so good.
But Asterisk never informs the other SIP client of the SDP change, and hence this device keeps sending RTP to the “old” RTP IP:port of the other device. Result is a loss of audio, of course.
This problem is also present in 1.4.13, and hence to be of general nature.
If RTP is going through Asterisk, everything is fine of course, as Asterisk terminates the RTP from both clients.
Any suggestions ??
Thanks,
Peter Mariager, dCAP
RTX Telecom A/S