[asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

Benjamin Jacob <ben4asterisk at yahoo.com> wrote: 

Tony Mountifield <tony at softins.clara.co.uk> wrote: In article <841715.98554.qm at web46405.mail.sp1.yahoo.com>,
Benjamin Jacob  wrote:
> 
> One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback
> of gsm files.
> So scouring the internet gave me the solution of installing ztdummy and loading it as a module.
> Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill
> no effect.
> 
> Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As
> far as I remember (am coming back to Asterisk after quite some time now), you don't really
> need to set anything over there for any zaptel specific compilation?
> 
> And yes, all the files are  gsm files and the codec used for the calls is ulaw.
> 
> I even tried converting those gsm files to wav using sox and then playing them, but the
> behaviour is the same.
> 
> Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI> zap show channels
   Chan Extension  Context         Language   MusicOnHold         
 pseudo            default
*CLI>

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If  ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line "internal_timing=yes". That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org

Thanks Tony for the response.
zap show channels shows that things are fine, as you said :
*CLI> zap show channels
   Chan Extension  Context         Language   MOH Interpret       
 pseudo            default                    default

Tried setting internal_timing to yes as well. Still  no difference.

Also,  I don't think my SIP gateway uses Silence suppression, because the same SIP gateway connections work fine with another Asterisk server.

This is getting seriously irritating now!!! Have tried all the tricks and tips I've been finding on the net.

Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to timing. But I am not the expert. 

- Ben.
         

---------------------------------
Be a better friend, newshound, and  know-it-all with Yahoo! Mobile.  Try it now._______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-usersBtw, I am on CentOS 5, with uname showing as:
Linux mserver.org 2.6.18-53.1.13.el5 #1 SMP Tue Feb 12 13:01:45 EST 2008 i686 i686 i386 GNU/Linux

And it is not a multiprocessor machine. Will the SMP option affect the working in any way?

- Ben.

       
---------------------------------
Be a better friend, newshound, and know-it-all with Yahoo! Mobile.  Try it now.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080425/9d5f84ad/attachment.htm