Hi, Please see pjsua_call_dump() as the starting reference. For, the jitter buffer size (or other info) please check pjmedia_session_get_stream_stat_jbuf(). BR, nanang On Tue, Apr 6, 2010 at 5:44 PM, Re Mo <remo9071 at gmail.com> wrote: > HI all > > I would like to retrieve the following parameters regarding an active SIP > call on an iPhone SIP client I've built: > > > Which Codec is being used > > Amount of packet loss > > Jitter buffer size > > Round trip delay > > > > Your asssitance on how this can be made would be greatly apprecaited. > > > > Thx > > Remo > > remo9071 at gmail.com > > > > _______________________________________________ > 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 > >
