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? --FC _______________________________________________ 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