I am using
chan_sip driver for SIP calling and I am trying to do an implementation where one of the call legs (WebRTC) between a two-party call never disconnects and used again and again for new calls from a Softphone endpoint.
The problem here that I am facing is that If the “always connected” WebRTC channel endpoint somehow got abruptly closed or say I closed the browser tab and the Asterisk didn’t receive any SIP Bye packet then this WebRTC channel will never disconnect and all the future calls will be bridged to this channel (which is of no use now).
I tried using
session-timers in the
originate mode but that didn’t work because If Asterisk is not able to write the INVITE on the WebSocket it will not take any action.
ERROR: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Also, I can’t use the
rtptimeout here because I am using
jitter-buffer which kind of disables the