RTP Timeout and SIP trunks

I’m running Asterisk 13, SIP only, with four SIP trunk connections to different DID suppliers - three incoming, one outgoing. I’ve noticed that sometimes disconnections are undetected, particularly when a mobile goes out of range. My global sip.conf settings include:


When turning on rtp debugging, I can see that RTP packets are being received across

If you are still receiving media then there’s nothing to know that the remote side is gone except the remote side telling you or examining the media stream to detect a long amount of silence. The detection of silence to indicate hangup isn’t currently supported.