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List: freeswitch-users
Subject: [Freeswitch-users] calls ending with MEDIA_TIMEOUT
From: freeswitch-users () digitaldan ! com (Dan)
Date: 2010-06-30 14:39:27
Message-ID: 23029141.3084.1277908748262.JavaMail.daniel () radio
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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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Anthony Minessale II
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