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)?
thanks
klaus
On Fri, Aug 1, 2008 at 11:56 AM, Klaus Darilion <
klaus.mailinglists@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
thanks, worked fine.
regards
klaus
Benny Prijono wrote:
On Fri, Aug 1, 2008 at 11:56 AM, Klaus Darilion
<klaus.mailinglists@pernau.at mailto:klaus.mailinglists@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
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