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OMAR RAMADAN omar.ramadan at berkeley.eduGood 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 at 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 at 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 at freyther.de> >> wrote: >> >>> >>> > On 25 Jan 2017, at 18:06, OMAR RAMADAN <omar.ramadan at 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 >> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20170126/ae13018d/attachment.htm>