On Fri, Aug 1, 2008 at 11:56 AM, Klaus Darilion < 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080801/a653f9bb/attachment.html