On Thu, Dec 12, 2019 at 6:39 AM marek <cervaj...@gmail.com> wrote: > hi, > > i have following topology > > PSTN - Asterisk ---- internet ----- router - jssip client (wss) > > Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP > connection to PSTN > > router - public IP/private IP (NAT) > > jssip client - private IP - sip over websocket to Asterisk PJSIP > > > ~30% of calls has problem with no audio. reason is that Asterisk is > sending RTP to private IP of jssip > > SDP looks the same for good call and bad call too > > > i searched through res_rtp_asterisk.c but i'm not sure where to put > DEBUG info about which IP and why Asterisk pick for RTP > > any hint? > > > is it possible debug Asterisk STUN request/response ? or is it hidden in > pjsip internals? >
ICE is performed using pjnath, which is part of pjproject and not Asterisk itself. Looking at the SDP and the ICE candidates can tell you the possibilities for the paths, and a wireshark capture can show you the actual traffic going back and forth and what is being attempted. -- Joshua C. Colp Senior Software Developer Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
