[asterisk-users] audio glitches in conference

[marco.mouta at gmail.com]

It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it.

What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme..

You need to have chan_dahdi dummie... 

Hope it helps.
Marco Mouta
Enviada do dispositivo sem fios BlackBerry®

-----Original Message-----
From: "Jeff Brower" <jbrower at signalogic.com>
Date: Wed, 24 Feb 2010 18:25:07 
To: Jonathan Addleman<jono at redowl.ca>
Cc: <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] audio glitches in conference

Jonathan-

> I'm having a problem with conferences both meetme and app_conference,
> though I've done most of the testing with meetme.
>
> Essentially, I get little gaps in the audio - usually fewer than a dozen
> or so samples, though it does vary. They seem to occur at random, but I
> usually get one ever few seconds, on average. They also seem to delay
> some buffer somewhere, so that if I start recording (via eagi) after the
> conference has been established for half an hour or so, the stream
> received by the eagi script delayed by about 10 seconds.

How did you measure the gaps?  Using signal or speech analysis software to display the recording?  If you measure
number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8
kHz G711 multiples of 80 samples between gaps) ?

-Jeff

> First, the preliminaries: I'm on a debian lenny system, using the
> 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was
> running xen, but I've shut down all the domU's to test if they were
> interfering, so now there's no sharing going on.
>
> I've been testing with a simple eagi script to grab the audio from the
> conference:
> #!/bin/sh
> cat /dev/fd/3 > /tmp/audio.raw
>
> I've been testing it using the following dialplan extensions:
> [test]
> exten => testeagi,1,Answer
> exten => testeagi,n,Wait(3)
> exten => testeagi,n,EAGI(testeagi.sh)
>
> exten => testmeet,1,Answer
> exten => testmeet,n,MeetMe(testconf,1qd)
>
> exten => testsound,1,Answer
> exten => testsound,n,Playback(testbeep-asterisk)
>
> (testbeep is just 30 seconds of sine wave)
>
> I've been trying things like this:
>
>
>
> originate Local/testsound at test extension testeagi at test
>
> The recorded audio plays back fine - no glitches.
> (an example is at http://www.vecotourism.org/audio17.wav)
>
> originate Local/testeagi at test extension testmeet at test
> originate Local/testsound at test extension testmeet at test
>
> This does have the glitches.
> (an example is at http://www.vecotourism.org/audio18.wav)
>
> What could be causing this? And is there anything else I could be doing
> to debug it? Thanks.
>
> --
> Jon-o Addleman - http://www.redowl.ca


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