Good point Alexander, I just realized that the connector is only handling
the signaling. What probably is happened is a call is trying to be
established to a dead SIP server. But we should still terminate the call
cleanly
On Jan 25, 2017 8:30 PM, "Alexander Chemeris"
<alexander.chemeris(a)gmail.com>
wrote:
Omar,
Just curious - is there any reason you're running RTP through the
osmo-sip-connector instead of directly to FreeSWITCH?
Please excuse typos. Written with a touchscreen keyboard.
--
Regards,
Alexander Chemeris
CEO Fairwaves, Inc.
https://fairwaves.co
On Jan 26, 2017 02:31, "OMAR RAMADAN" <omar.ramadan(a)berkeley.edu> wrote:
> I've seen it a few times in production already and it filled the disk.
> You should be able to reproduce it by killing an active RTP stream. I have
> been using freeswitch, but I don't imagine it is limited to this SIP
> server. It looks like sofia-sip is driven to continue to receiving media
> and gets nothing back while the call should be terminated.
>
> On Wed, Jan 25, 2017 at 12:06 PM, Holger Freyther <holger(a)freyther.de>
> wrote:
>
>>
>> > On 25 Jan 2017, at 18:06, OMAR RAMADAN <omar.ramadan(a)berkeley.edu>
>> wrote:
>> >
>> > If the SIP server dies in the middle of a call, osmo-sip-connector is
>> in a bad state and generates a never ending stream of error messages:
>>
>>
>> Can you reliable reproduce it? It seems sofia-sip is struggling with
>> some input to it and goes crazy after that. I lack a stable way to
>> reproduce it. The lack of \n in that message is annoying too. :(
>>
>> holger
>
>
>