Use a timer to send inside dialog request outside to SIP server is better than SIP server call checking. Some NAT devices still will cut mapping if only sequence packets first from outside. regards, Gang 2011/11/11 R?gis Montoya <r3gis.3r at gmail.com>: > Ok, > Thanks all for the reply. > > Indeed it makes sense for mainstream use cases. > > So I think that the feature should rather be a pjsip plugin (or an app > feature) that would be able to send keep alive OPTION as Sa?l said. > It will be active if the call is established from a local account or an > account without sip proxy. > I'll try to do that on my side and I'll ask if I need help ;). > > Thx, > Regards, > R?gis > > > On 10/11/2011 12:36, Benny Prijono wrote: > > 2011/11/10 R?gis Montoya <r3gis.3r at gmail.com> >> >> Hi, >> >> For now pjsip KA mechanism for UDP sig is to send keep alive packets only >> on the registration stream. When pjsip make an INVITE there is not such a >> mechanism for control channel. - At least I didn't found one ;) -. >> > > I think when you're behind NAT, you should always send the call via the > server (i.e. not directly to callee). If you call direct, your Contact URI > will be wrong if you're behind symmetric NAT. And anyway your incoming call > will also come from the server, so doing it this way creates a good symmetry > there between incoming and outgoing calls. At least this is the model that > I've thought about when doing the KA. > > ?Benny > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
