On 7/17/14, 3:41 PM, Frank Carmickle wrote: > > On Jul 16, 2014, at 4:05 PM, Andras FOGARASI <fogar...@fogarasi.com> wrote: > >> On 7/16/14, 10:00 PM, Frank Carmickle wrote: >>> >>> On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla <mico...@gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> I expect that the signaling is ok at least for call setup. >>>> >>>> From signling point of view, I can think of following situations: >>>> - endpoints send keep alive packets (or session updates) which are no >>>> answered. You can add an xlog(...) at the top of request_route{} and >>>> reply_route{} blocks printing at least the method, call-id, cseq, from and >>>> to header, plus the response code for reply block. In this case you can >>>> see if there is some signaling before call is dropped. >>> >>> Is this happening just on calls between two phones in your domain, or is >>> there a carrier/federation involved? >>> >> >> No other parties are involved, only the two phones involved (and the >> proxy of course). >> > > I would expect that if it was a NAT issue you would see it much sooner than > 15 minutes, 30-60 seconds. Are session timers being stripped by Kamailio? > You say it's a TURN server or is it acting more like a media relay where it > is signaled into the path? What TURN server are you using? How is it > configured? >
The problem occurs even without TURN, in pure peer-to-peer mode. We use TURN only in emergency case (symmetric NAT and like that...). I do nothing with session timers - i didn't think about it until now... Andras _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users