Hi, I am doing some experiment in our college using PJSIP. I am using OpenSER with RTPProxy as media relay. All audio and video traffics are relayed by RTPPROXY to destinations. Consider an example of video conference. Person A wants to call Person-B, Person-C, Person-D simultaneously. I knew that this can be done by calling following API function *multiple * times: pj_status_t<http://www.pjsip.org/pjlib/docs/html/group__PJ__BASIC.htm#gab43ba3167bd2a2ab4580509dbf79200e>pjsua_call_make_call ( pjsua_acc_id<http://www.pjsip.org/pjsip/docs/html/group__PJSUA__LIB__BASE.htm#ga01a78e17d7787f7e0ea5efd240f3e427> *acc_id*, const pj_str_t <http://www.pjsip.org/pjlib/docs/html/structpj__str__t.htm>* *dst_uri*, const pjsua_call_setting<http://www.pjsip.org/pjsip/docs/html/structpjsua__call__setting.htm>* *opt*, void * *user_data*, const pjsua_msg_data<http://www.pjsip.org/pjsip/docs/html/structpjsua__msg__data.htm>* *msg_data*, pjsua_call_id<http://www.pjsip.org/pjsip/docs/html/group__PJSUA__LIB__BASE.htm#gad4eb99a78c98ddbd83aecf7e933fc684>* *p_call_id* ) But this means Person A has to upload 3 audio and 3 video streams to RTPPROXY (media relay). Instead of uploading 3 audio and 3 video streams to RTPPROXY, it can upload only one audio and one video streams to RTPPROXY. Now I have did some study on RTPPROXY and found that RTPPROXY uses Call-ID and From Tag from SIP packets to broadcast RTP traffic. RTPPROXY doesn't care about TO Tags in SIP Packets because It can relay same incoming traffic to multiple destinations. ( http://lists.rtpproxy.org/pipermail/devel/2008-June/000074.html) *Now my questions is "How can I place multiple video call using pjsua in such a way that SIP INVITE Packets will contain same Call-D header for multiple SIP INVITE packets ?"* Please advice me how to achieve this goal. Regards Tapas -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120808/85406b06/attachment-0001.html>
