I have a similar call
quality problem that Ray related when using asterisk is in the middle of media
path specially, when asterisk is performing codec transcoding. I am not sure
this problem is related to asterisk lack of jitter buffer but it seems to be.
How can we make sure
asterisk is not screwing up the jitter buffer? Someone on this list knows
exactly how asterisk performs the “RTP proxing” ?
Juliano
De:
serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org]
Enviada em: domingo, 30 de outubro
de 2005 17:41
Para: Ray Van Dolson
Cc: serusers@lists.iptel.org
Assunto: Re: [Serusers] Inserting
SER into my voice network
I cant see that at all
from your diagram. I see only an ATA and Media Gateway doing final conversion
where jitter buffer would be useful. If turing on a jitter buffer in Asterisk
helps then one of the other 2 is broke.
On 10/30/05, Ray Van
Dolson <rayvd@digitalpath.net>
wrote:
When I take Asterisk out of the media path, this is
correct. And I believe my
ISP's media gateway *does* have a jitter buffer.
Since Asterisk was an media endpoint before (it doesn't just proxy the rtp
on), its lack of jitter buffer was hurting us in some cases.
Ray
On Sun, Oct 30, 2005 at 08:55:13AM -0600, Mark Aiken wrote:
>
> The only jitter buffers that matter in your diagram
are the SIP ATA and
> Media Gateway. Both should have jitter buffers at
the point where they
> convert RTP to PCM. If adding a jitter
buffer inside the network path
> somewhere helps then something else is broken.
>
>
>
> Mark
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