RES: [Serusers] Inserting SER into my voice network

[Juliano Duque da Silva]

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 at iptel.org [mailto:serusers-bounces at lists.iptel.org] Em nome
de Mark Aiken
Enviada em: domingo, 30 de outubro de 2005 17:41
Para: Ray Van Dolson
Cc: serusers at 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 <HYPERLINK
"mailto:rayvd at digitalpath.net"rayvd at 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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