asterisk users - Feb 2005 - IAX Voice Quality Issues

I am running * 1.0.5 and have been having lots of problems with
outgoing calls and their sound quality. I am using ULAW for the codec
and sixtel for termination. Basically the problem is that portions of
the call seem to be lost and replaced with silence. Sometimes I can't
hear the person talking othertimes they can't hear me. This situation
comes and goes throughout the call. Bandwidth isn't an issue as I have
a 3MB/1MB connection and there is at most 2 concurrent connections.
Also using pingplotter to monitor iax2.sixtel.net shows little or no
packetloss.

Just as further info, I am using a SPA-2000 to connect to * with G711u
as the preferred codec.

Anyone else experience the like or have any suggestions on what may be
causing this or ideas on how to debug?

Brian

>
>Just as further info, I am using a SPA-2000 to connect to * with G711u
>as the preferred codec.
>
>
>  
>
Maybe you have silence suppression enabled on the SPA?  That does not 
play nice with Asterisk.

Silence Suppression is off or set to no of the SPA. I changed
jitterbuffer=no and things seem better. Will need to do some more
testing.


On Wed, 09 Feb 2005 13:03:06 -0500, Andres <andres@telesip.net>
wrote:
> 
> 
> >
> >Just as further info, I am using a SPA-2000 to connect to * with G711u
> >as the preferred codec.
> >
> >
> >
> >
> Maybe you have silence suppression enabled on the SPA?  That does not
> play nice with Asterisk.
> 
>

No quite sure what you mean by timimg source. But I have a T100P card
in the box and zaptel drivers loaded.


On Wed, 9 Feb 2005 13:49:15 -0500, Paul Rodan <asterisk@glitch.cc>
wrote:
> That jitter buffer has caused nothing but problems for me.
> 
> But that was a few months ago, haven't tried it lately.
> 
> What are you using as your timing source?

I have been qualityhaving problems although I don't have all the
details at this point - I currently suspect mine are due to Latency
issues as both ends are colo boxes which don't seem to drop packets in
testing (ping).

Latency right now seems to be in the 80ms range on each end of the server.

The problems on our lines tend to be slight breakups in the conversation. 

I am in the motions of finding a closer end point and hoping that will
help, otherwise I'm not too sure as bandwidth seems to be fine and
clean.

Anyway, let me know if you get any further ideas and I will let you
know what happens when I cut down the time between nodes.

On Wed, 9 Feb 2005 12:30:39 -0500, Brian Dingman <bdingman@gmail.com>
wrote:
> I am running * 1.0.5 and have been having lots of problems with
> outgoing calls and their sound quality. I am using ULAW for the codec
> and sixtel for termination. Basically the problem is that portions of
> the call seem to be lost and replaced with silence. Sometimes I can't
> hear the person talking othertimes they can't hear me. This situation
> comes and goes throughout the call. Bandwidth isn't an issue as I have
> a 3MB/1MB connection and there is at most 2 concurrent connections.
> Also using pingplotter to monitor iax2.sixtel.net shows little or no
> packetloss.
> 
> Just as further info, I am using a SPA-2000 to connect to * with G711u
> as the preferred codec.
> 
> Anyone else experience the like or have any suggestions on what may be
> causing this or ideas on how to debug?
> 
> Brian
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