[asterisk-dev] Deadlock in chan_sip, caused by weird media re-invite from remote side

Joshua Colp jcolp at digium.com
Tue Apr 5 07:39:17 CDT 2016

Nir Simionovich wrote:


> Soft Phone -> Asterisk A -> Asterisk B -> Carrier
>    Soft phone is behind a NAT. Asterisk servers are not, same as the
> carrier.
>    We've noticed that the carrier tries to run a media re-invite, after
> the call had basically
> dropped from Asterisk B, and tries to do it over and over again, without
> stopping. Eventually,
> that would dead-lock chan_sip completely, requiring a full blown
> asterisk restart.
>    Any of you ever encountered anything like this?
>    I've mitigated the issue by forcing two different codecs on the two
> sides of Asterisk B, basically,
> preventing the media re-invite - but it isn't the proper solution.

I can't say I've heard of anyone running into this problem and it's a 
common enough scenario. I'd suggest trying against the latest 13 and if 
not resolved then filing an issue with a description and backtrace.


Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

More information about the asterisk-dev mailing list