Hi, I have a pjsip application similar to simple pjsua example ( http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm) . I am calling my application from a sip phone (xlite). i need bidirectional flow so I am connecting media to sound card on 'on_call_media_state' callback. ----------- static void on_call_media_state(pjsua_call_id call_id) { pjsua_call_info ci; pjsua_call_get_info(call_id, &ci); if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { /// When media is active, connect call to sound device. cout << "Connecting media to sound card" << endl; pjsua_conf_connect(ci.conf_slot, 0); pjsua_conf_connect(0, ci.conf_slot); } } ------------ I can receive sip phone's voice from my application succesfully. sip phone can also receive voice from my application succesfully. however, sip phone also receives its own voice. I think i am somehow looping media back to sip phone. How do i prevent looping media back to sip phone? Should i use echo cancellation api or is there any other trick that i don't know yet? Any help is appreciated! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100421/64d0c4b5/attachment.html>
