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@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@freyther.de> wrote:

> On 25 Jan 2017, at 18:06, OMAR RAMADAN <omar.ramadan@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