RTP Timeout and SIP trunks

Asterisk Asterisk Support
1

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:

rtptimeout=30
rtpholdtimeout=600
rtpkeepalive=10

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

2

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.