[Freeswitch-users] calls ending with MEDIA_TIMEOUT

Wed Jun 30 20:41:19 PDT 2010

you neet to get pcaps of the calls and look at the rtp and sip going on.

On Wed, Jun 30, 2010 at 9:39 AM, Dan <freeswitch-users at digitaldan.com>wrote:

> Thanks for your response, I put everything up on pastebin
> http://pastebin.freeswitch.org/13322 . The application in question is
> actually javascript, I'm using lua in production but was switching to the
> posted js version with the upgrade.
>
> Now that I posted it i realized I have
>
> <action application="set" data="rtp_timer_name=none"/>
>
> in my dial plan, I believed I used it in the older version to get around
> some dtmf issues or choppy playback (can't remember), not sure if this could
> be part of the issue (although it works fine in the production version I'm
> running)
>
>
> So I pulled one of the recordings that hung up after 4 minutes, but was
> only 24 seconds long, it sounded fine (but obviously too short).  But
> another one that dropped after 5 minutes and only 19 seconds in length was
> very choppy and  included short spurts of audio from parts of the call that
> were much longer then 19 seconds.
> ------------------------------
> *From: *"Anthony Minessale" <anthony.minessale at gmail.com>
> *To: *freeswitch-users at lists.freeswitch.org
> *Sent: *Tuesday, June 29, 2010 12:54:21 PM
> *Subject: *Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT
>
>
> it's not 100% accurate in the media timeout.
> It would be too expensive to use actual timers, it uses the number of
> samples you should be getting from rtp
> and a number of loops where no media was received.
>
> Migrating from svn 13000 range to GIT is a big step and you may have to
> adjust to some new behaviors.
> media_timeout may not even have existed that long ago I don't recall.
>
> If you don't need media timeouts turn off the param or turn it up to
> longer.
>
>
> On Tue, Jun 29, 2010 at 1:09 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Pastebin your dialplan and the lua script for starters. Also, is it the
>> 5300 that is responding with the media timeout?
>> -MC
>>
>> On Tue, Jun 29, 2010 at 10:15 AM, Dan <freeswitch-users at digitaldan.com>wrote:
>>
>>> Hi guys, I have been running two freeswitch boxes (13754M)  that answer
>>> calls from a cisco 5300 (both on the same network) and records them to disk
>>> with a small lua application.  This has been working well for the past few
>>> months.  I decided to upgrade one of them to trunk (  git-3fbd9e2 2010-06-11
>>> 11-08-51 -0500 ) and have run into a problem.  Some calls will fail with a
>>> MEDIA_TIMEOUT  after a few minutes, the time it takes to fail ranges from 4
>>> minutes to 10 minutes,  I don't have a full sip trace or pcap dump yet, I
>>> reverted back to the old freeswitch version (on the same hardware) and have
>>> not been able to reproduce it in a test environment yet ( I continue to
>>> try).   Below are the relevant lines from the log files for one of the
>>> calls:
>>>
>>> 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/
>>> nobody at 192.168.21.4) Callstate Change ACTIVE -> HANGUP
>>> 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup
>>> sofia/external/nobody at 192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT]
>>> 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal
>>> sofia/external/nobody at 192.168.21.4 [KILL]
>>> 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal
>>> sofia/external/nobody at 192.168.21.4 [BREAK]
>>> 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146
>>> sofia/external/nobody at 192.168.21.4 Restore previous codec PCMU:0.
>>>
>>> My configuration is bone stock, so the rtp timeout value is at 300,  but
>>> I have some calls that have lasted only 4 minutes.  One other piece of
>>> information is that on one of the recordings that was hung up after 4
>>> minutes and 17 seconds the recorded file was only 24 seconds long (it
>>> stopped recording after the first 24 seconds) , so I'm assuming freeswitch
>>> did not think there were any rtp packets to record.
>>>
>>> Any ideas on where to start debugging this?  I have setup a new
>>> freeswitch box connected to the same 5300 to reproduce, but have not been
>>> able to generate the call volume ( there where around 30 calls being
>>> recorded) yet, but I'm working on it.
>>>
>>> Thanks!
>>>
>>> _______________________________________________
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>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
>>>
>>
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>>
>
>
> --
> Anthony Minessale II
>
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-- 
Anthony Minessale II

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