I have found that calls are very inconsistent. I use Kamailio 5, Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls into a WebRTC client (tryit-jssip), sometime the calls stay up until I close them (10-15 minutes), others times those calls drop in 30 seconds. This is extremely confusing...does anyone else experience this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of moving parts with WebRTC. However, the typical reason why an established call would drop after ~30 sec (32, to be precise) is that the end-to-end ACK from the caller, which completes the required "three-way handshake" for call establishment, does not reach the callee. This is because it's not constructed correctly by the calling UA, not routed correctly by intermediate entities, or isn't sent at all by the calling UA.