thanks, worked fine. regards klaus Benny Prijono wrote: > On Fri, Aug 1, 2008 at 11:56 AM, Klaus Darilion > <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote: > > Hi! > > For some interoperability testing I need a SIP client which changes > media ports during reINVITE. Can I implement such a client based on > pjsua API (e.g. somehow force a new RTP port)? > > > Not with pjsua API. But for a quick test you can put this hack in > pjsua_media.c:1025: > > /* Recreate transport */ > if (call->med_tp_st != PJSUA_MED_TP_IDLE) { > pjsua_media_channel_deinit(call_id); > status = pjmedia_transport_udp_create(pjsua_var.med_endpt, NULL, > 40000+(pj_rand()%20000), 0, > &call->med_tp); > pj_assert(status==PJ_SUCCESS); > } > > Cheers > 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 tha
