<div dir="ltr"><div class="gmail_quote">Let me make sure I understand: You want me to use the GDB and get that info to you as described here: </div><div class="gmail_quote"><a href="https://wiki.freeswitch.org/wiki/Debugging_Freeswitch">https://wiki.freeswitch.org/wiki/Debugging_Freeswitch</a> ?<br></div><div class="gmail_quote"><br></div><div class="gmail_quote">Do you want wireshark info as well? I know that there have been some issues with IPv6 - perhaps this is related?</div><div class="gmail_quote"><br></div><div class="gmail_quote">thank you,</div><div class="gmail_quote">chris</div><div class="gmail_quote"><br></div><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Michael Jerris</b> <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span><br>Date: Thu, Jul 30, 2015 at 10:07 AM<br>Subject: Re: [Freeswitch-users] webrtc/Sip issue<br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br><br><br><div style="word-wrap:break-word">As i said before. a debug trace including the sip traffic might tell us more. A partial log that you have edited to remove data, without sip trace will not tell us anything useful.<div><br><div><blockquote type="cite"><span class=""><div>On Jul 30, 2015, at 7:22 AM, Chris Mandra <<a href="mailto:mandra@gmail.com" target="_blank">mandra@gmail.com</a>> wrote:</div><br></span><div><div dir="ltr"><span class="">Hi Michael, here's some more debug info: I just want to point out that only a couple users have this issue. The same hardware and software with diff users doesn't exhibit this issue. </span><div><br><div>2015-07-30 10:42:56.431486 [ALERT] switch_core_session.c:2760 <a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a> receive message [PROGRESS]</div><span class=""><div>2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early media</div><div>2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5412 STUN Success [104.197.44.217]:[32306]</div></span><div>2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5884 AUDIO RTP [<a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a>] 10.240.125.130 port 32306 -> 2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 20</div><span class=""><div>2015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms</div><div>2015-07-30 10:42:57.741478 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS ERROR: [Remote Address Error!]</div></span><div>2015-07-30 10:42:57.741478 [NOTICE] switch_core_media.c:6567 Hangup <a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]</div><div>2015-07-30 10:42:57.741478 [ALERT] switch_channel.c:3277 Send signal <a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a> [KILL]</div><div>2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal <a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a> [BREAK]</div><div><br></div><div><br></div><div>Startup ipv6 complaints:</div><div><br></div><div><span class=""><p><span>015-07-30 10:57:10.336255 [ERR] sofia.c:2935 Error Creating SIP UA for profile: external-ipv6 (<a>sip:mod_sofia@[::1]:5080;transport=udp,tcp</a>) ATTEMPT 2 (RETRY IN 5 SEC)</span></p><p><span>2015-07-30 10:57:10.436224 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal-ipv6 (<a>sip:mod_sofia@[::1]:5060;transport=udp,tcp</a>) ATTEMPT 2 (RETRY IN 5 SEC)</span></p><p><span>freeswitch@internal></span></p></span><p><span>2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge (REGISTER) on sofia profile 'internal' for [<a href="mailto:1000@xxx.xxxxx.xxx" target="_blank">1000@xxx.xxxxx.xxx</a>] from ip 50.242.9.206</span></p><span class=""><p><span>2015-07-30 10:57:13.616221 [CONSOLE] mod_voicemail.c:4066 Event Thread Started</span></p><p><span>2015-07-30 10:57:15.336251 [ERR] sofia.c:2935 Error Creating SIP UA for profile: external-ipv6 (<a>sip:mod_sofia@[::1]:5080;transport=udp,tcp</a>) ATTEMPT 3 (RETRY IN 5 SEC)</span></p><p><span>2015-07-30 10:57:15.336251 [ERR] sofia.c:2945 Error Creating SIP UA for profile: external-ipv6 (<a>sip:mod_sofia@[::1]:5080;transport=udp,tcp</a>)</span></p><p><span>The likely causes for this are:</span></p><p><span>1) Another application is already listening on the specified address.</span></p><p><span>2) The IP the profile is attempting to bind to is not local to this system.</span></p><p><span>2015-07-30 10:57:15.436256 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal-ipv6 (<a>sip:mod_sofia@[::1]:5060;transport=udp,tcp</a>) ATTEMPT 3 (RETRY IN 5 SEC)</span></p><p><span>2015-07-30 10:57:15.436256 [ERR] sofia.c:2945 Error Creating SIP UA for profile: internal-ipv6 (<a>sip:mod_sofia@[::1]:5060;transport=udp,tcp</a>)</span></p><p><span>The likely causes for this are:</span></p><p><span>1) Another application is already listening on the specified address.</span></p><p><span>2) The IP the profile is attempting to bind to is not local to this system.</span></p></span></div><div><br></div><div><br></div><div><br></div><div>And after disabling ipv6</div><div><p><span>2015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application ladspa_run Requires media! pre_answering channel sofia/internal/</span><a href="mailto:1000@xxx.xxxxx.xxx" target="_blank">1000@xxx.xxxxx.xxx</a></p><span class=""><p>2015-07-30 11:12:50.116372 [INFO] switch_core_session.c:2760 Sending early media</p><p><span>2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5412 STUN Success [xxx.197.44.217]:[23022]</span></p></span><p><span>2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5884 AUDIO RTP [<a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a>] 10.xxx.xxx.130 port 23022 -> fdae:f681:faac:a135:b84e:e923:21d0:144a port 52098 codec: 111 ms: 20</span></p><span class=""><p><span>2015-07-30 11:12:50.176368 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms</span></p><p><span>2015-07-30 11:12:50.176368 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS ERROR: [Remote Address Error!]</span></p></span><p><span>2015-07-30 11:12:50.176368 [NOTICE] switch_core_media.c:6567 Hangup <a href="mailto:sofia/internal/1000@xxx.xxxxx.xxx" target="_blank">sofia/internal/1000@xxx.xxxxx.xxx</a> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]</span></p></div><span class=""><div><br></div><div><br></div><div><br></div>On Wednesday, July 29, 2015, Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">a debug trace including the sip traffic might tell us more.<div><br><div><blockquote type="cite"><div>On Jul 29, 2015, at 3:21 PM, Chris Mandra <<a>mandra@gmail.com</a>> wrote:</div><br><div><div dir="ltr">Hi guys - one of my users keeps getting this error in fs_cli when he tries to connect via Chrome (web-rtc) on a macbook running yosemite.<div><p><span>2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!]</span></p><p><span>I'm using essentially the same setup and never have an issue. Any ideas? He doesn't get anything very useful in the chrome console, just this:</span></p><p><span>*********************</span></p><p><span>Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m</span></p><p><span>sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection</span></p><p><span>sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted</span></p><p><span>sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed</span></p><p><span>sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected</span></p><div><span>
</span><br></div><p><span>sc-webrtc.js:931 _rejected called</span></p><p><span>*******************</span></p><p><span><br></span></p><p><span>Any ideas? We're using latest FS and sip.js</span></p><p><span><br></span></p><p><span>thanks,</span></p><p><span>chris</span></p></div></div></div></blockquote></div></div></div></blockquote></span></div></div></div></blockquote></div><br></div></div><br>_________________________________________________________________________<br>
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