[Freeswitch-users] webrtc/Sip issue

Thu Jul 30 19:02:06 MSD 2015

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> 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 ?
>
> 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>
> 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>
>
>
> 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> 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 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] 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 [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 [KILL]
> 2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal
> 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] 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
>
> 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] 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 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
>
>
> On Wednesday, July 29, 2015, Michael Jerris <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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>
>
> --
> mandra
> c:410.258.5281
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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