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.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web:
http://www.evaristesys.com/,
http://www.csrpswitch.com/