[Freeswitch-users] webrtc/Sip issue

Michael Jerris mike at jerris.com
Sat Aug 1 02:38:06 MSD 2015


Does your host have IPv6 running and routable?  What version of freeswitch are you using again?

2015-07-31 22:08:00.106489 [DEBUG] switch_core_media.c:3261 Choose rtp candidate, index 0, fdae:f681:faac:a135:41e8:9d44:1cdb:503e:53654


I think this issue was fixed in this commit:

commit 2815994647ef03fda488685899262450eb4b935c
Author: Brian <brian at freeswitch.org>
Date:   Thu Jun 11 08:24:31 2015 -0500

    FS-7638 properly match ACL with IPv4 mapped IPv6 addresses



> On Jul 31, 2015, at 6:20 PM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Hi guys - I've tried to debug as you asked - attached an rtf of the troubled session. (pasting into gmail screwed up formatting)
> 
> Let me know what you think. 
> 
> Thanks!
> 
> chris
> 
> On Thu, Jul 30, 2015 at 11:02 AM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
> no GDB (gdb is for segfaults)
> 
> complete debug log from console, from fs_cli:
> 
> fs_cli>sofia global siptrace on
> fs_cli>sofia debug all 9
> fs_cli>fsctl loglevel 9
> fs_cli>fsctl console 9
> 
> then reproduce what is your problem
> 
> then attach it all, complete of all relevant info like configurations, topology, whatever
> 
> 
> On Thu, Jul 30, 2015 at 4:27 PM, Chris Mandra <mandra at gmail.com <mailto:mandra at 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: 
> https://wiki.freeswitch.org/wiki/Debugging_Freeswitch <https://wiki.freeswitch.org/wiki/Debugging_Freeswitch> ?
> 
> 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 at jerris.com <mailto:mike at jerris.com>>
> Date: Thu, Jul 30, 2015 at 10:07 AM
> Subject: Re: [Freeswitch-users] webrtc/Sip issue
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at gmail.com <mailto:mandra at 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 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at xxx.xxxxx.xxx> receive message [PROGRESS]
>> 2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early media
>> 2015-07-30 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 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at xxx.xxxxx.xxx>] 10.240.125.130 port 32306 -> 2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 20
>> 2015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms
>> 2015-07-30 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 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at 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 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at xxx.xxxxx.xxx> [KILL]
>> 2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal sofia/internal/1000 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at 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 at internal>
>> 
>> 2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at xxx.xxxxx.xxx <mailto:1000 at 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 ipv6
>> 2015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application ladspa_run Requires media! pre_answering channel sofia/internal/1000 at xxx.xxxxx.xxx <mailto:1000 at 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 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at 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 at xxx.xxxxx.xxx <mailto:sofia/internal/1000 at xxx.xxxxx.xxx> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>> 
>> 
>> 
>> 
>> On Wednesday, July 29, 2015, Michael Jerris <mike at jerris.com <mailto:mike at 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 at 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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> 
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> 
> 
> 
> -- 
> mandra
> c:410.258.5281 <tel:410.258.5281>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618 <tel:%2B39-347-2665618>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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> 
> 
> 
> -- 
> mandra
> c:410.258.5281
> <badSip2.rtf>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
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> 
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