no GDB (gdb is for segfaults)complete debug log from console, from fs_cli:fs_cli>sofia global siptrace onfs_cli>sofia debug all 9fs_cli>fsctl loglevel 9fs_cli>fsctl console 9then reproduce what is your problemthen attach it all, complete of all relevant info like configurations, topology, whateverOn Thu, Jul 30, 2015 at 4:27 PM, Chris Mandra <mandra@gmail.com> wrote:Let me make sure I understand: You want me to use the GDB and get that info to you as described here:Do you want wireshark info as well? I know that there have been some issues with IPv6 - perhaps this is related?thank you,chris_________________________________________________________________________---------- Forwarded message ----------
From: Michael Jerris <mike@jerris.com>
Date: Thu, Jul 30, 2015 at 10:07 AM
Subject: Re: [Freeswitch-users] webrtc/Sip issue
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>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.On Jul 30, 2015, at 7:22 AM, Chris Mandra <mandra@gmail.com> wrote: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.2015-07-30 10:42:56.431486 [ALERT] switch_core_session.c:2760 sofia/internal/1000@xxx.xxxxx.xxx receive message [PROGRESS]2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early media2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5412 STUN Success [104.197.44.217]:[32306]2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5884 AUDIO RTP [sofia/internal/1000@xxx.xxxxx.xxx] 10.240.125.130 port 32306 -> 2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 202015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms2015-07-30 10:42:57.741478 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS ERROR: [Remote Address Error!]2015-07-30 10:42:57.741478 [NOTICE] switch_core_media.c:6567 Hangup sofia/internal/1000@xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]2015-07-30 10:42:57.741478 [ALERT] switch_channel.c:3277 Send signal sofia/internal/1000@xxx.xxxxx.xxx [KILL]2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal sofia/internal/1000@xxx.xxxxx.xxx [BREAK]Startup ipv6 complaints:015-07-30 10:57:10.336255 [ERR] sofia.c:2935 Error Creating SIP UA for profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC)
2015-07-30 10:57:10.436224 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC)
freeswitch@internal>
2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000@xxx.xxxxx.xxx] from ip 50.242.9.206
2015-07-30 10:57:13.616221 [CONSOLE] mod_voicemail.c:4066 Event Thread Started
2015-07-30 10:57:15.336251 [ERR] sofia.c:2935 Error Creating SIP UA for profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)
2015-07-30 10:57:15.336251 [ERR] sofia.c:2945 Error Creating SIP UA for profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp)
The likely causes for this are:
1) Another application is already listening on the specified address.
2) The IP the profile is attempting to bind to is not local to this system.
2015-07-30 10:57:15.436256 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)
2015-07-30 10:57:15.436256 [ERR] sofia.c:2945 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp)
The likely causes for this are:
1) Another application is already listening on the specified address.
2) The IP the profile is attempting to bind to is not local to this system.
And after disabling ipv62015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application ladspa_run Requires media! pre_answering channel sofia/internal/1000@xxx.xxxxx.xxx
2015-07-30 11:12:50.116372 [INFO] switch_core_session.c:2760 Sending early media
2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5412 STUN Success [xxx.197.44.217]:[23022]
2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5884 AUDIO RTP [sofia/internal/1000@xxx.xxxxx.xxx] 10.xxx.xxx.130 port 23022 -> fdae:f681:faac:a135:b84e:e923:21d0:144a port 52098 codec: 111 ms: 20
2015-07-30 11:12:50.176368 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms
2015-07-30 11:12:50.176368 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
2015-07-30 11:12:50.176368 [NOTICE] switch_core_media.c:6567 Hangup sofia/internal/1000@xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
On Wednesday, July 29, 2015, Michael Jerris <mike@jerris.com> wrote:a debug trace including the sip traffic might tell us more.On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra@gmail.com> wrote: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.2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
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:
*********************
Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection
sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed
sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected
sc-webrtc.js:931 _rejected called
*******************
Any ideas? We're using latest FS and sip.js
thanks,
chris
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