On Wed, Aug 13, 2008 at 11:27 AM, Pedro Gon?alves <pedro.pandre at gmail.com>wrote: > Hi! > > I already managed to use ICE to establish a call between a client behind > symmetric NAT and a client directly (no NAT) connected to my SIP Server, > which proves that ICE is working fine :) > > However, I have a problem when ICE can't establish a connection (ex: > Port Restricted Cone NAT to Symmetric NAT). > I think I am missing something, because the media (RTP and RTCP) is not > starting in caller when the initial answer (200 OK INVITE) is received > by the caller. > > As I am using PJSIP at a low level, the process flow is like this: > [When starting a call] > > == Caller == > pjmedia_ice_create() > pjmedia_transport_media_create() > pjmedia_transport_encode_sdp() > pjsip_inv_create_uac() > pjsip_inv_invite() > pjsip_inv_send_msg() > > [INVITE goes to answerer] > > == Answerer == > pjsip_inv_answer() > pjmedia_ice_create() > pjmedia_transport_media_create() > pjmedia_transport_encode_sdp() > pjsip_inv_send_msg() > pjmedia_transport_media_start() > > [200 OK goes to caller] > > == Caller == > on_media_update(): > pjmedia_transport_media_start() > > on_ice_complete(): > pjmedia_endpt_create_sdp() > pjmedia_transport_encode_sdp() > pjsip_inv_reinvite() > pjsip_inv_send_msg() > (...) > > With this process flow, I can see that the default candidates (relayed > candidates) are chosen and negotiated; I can even see the media flow > going from answerer to relay sever to caller, but not the other way > around (caller to relay server to answerer). > Any idea why? > > Nope. > Am I missing something? > > Yes. The log. :) And btw, in your log, all SIP messages are written as one very long line and this makes it very hard to read. Can you fix that first? Cheers Benny > Cheers > Pedro Gon?alves > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080813/c596869e/attachment.html
