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

[Nir Simionovich]

Tried initially with 13.2 - was exactly the same. I'll try latest 13 stable
and see if it re-creates.



On Tue, Apr 5, 2016 at 3:39 PM, Joshua Colp <jcolp at digium.com> wrote:

> Nir Simionovich wrote:
>
> <snip>
>
>
>> 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.
>
> Cheers,
>
> --
> 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
>
>
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