Detect from asterisk an rtp drop with an endpoint


I was wondering if there is any way to detect an RTP leak in communications with a client and if so, be able to launch a SIP renegotiation with the endoint.

In my case I have a pbx pool, some sbc and rtpengine, the idea is to renegotiate with the client a new candidate for rtp, obviously without dropping the call

Thank you very much in advance.
All the best

In what cases would you want to change the voice path, during the call?

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