Unavailable ~120 seconds after a call

I’m at a loss here and I hope that somebody can point me in the right direction! :slight_smile:

I’m experiencing that approx. 120 seconds after a call, asterisk marks my sip user as unavailable.

When I enable the debug in jssip there’s no warning or error of sorts, the communication just cuts off. Then, after some time, jssip re-registers again and things work fine again (until i make/receive another call).

Also, when in call I can hear audio just fine. The call is OK.

Any pointers to where to look is greatly appreciated! :slight_smile:

Thank you!

Likely to be a dynamic NAT or firewall rule, in a router, timing out.

Thanks for the reply. The weird thing though, is that when I do nothing at all, the connection is fine and the SIP dialogs keeps on. The error only occurs after a call.

Weird that there is no warning or error.

What version of Asterisk are you using?
Did you mean pjsip (instead of jssip)?

Do you have rtp_timeout (or rtptimeout) set anywhere?

There might have been an issue with that setting:
But then you should have gotten an error on the console:
res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel for lack of video RTP activity in 120 seconds

Did you check the console with asterisk -rvvvvvvvvv (heavy verbosity)?

I think the OP: is looking at the browser client logging, when, of course, when asking here, they should be looking at the Asterisk logging.

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