For the case where Asterisk is handling RTP bridging, and not TDM/DAC conversion, its not a jitter buffer you need but real-time platform performance.

 

If Asterisk is the cause of jitter when its relaying media then its a flaw in Asterisk, just like a slow/broken router will cause jitter when forwarding from one interface to another.

 

The fix is not to add a jitter buffer, but quite the opposite and all buffering and delays should be minimized.

 

You would need to insure that real-time priority for the entire RTP path though Asterisk is guaranteed. I expect potential deadlock conditions would require much work for this to be successful.

 

Mark 

 

On 10/31/05, Juliano Duque da Silva <juliano.duque@terra.com.br> wrote:

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@lists.iptel.org [mailto: serusers-bounces@lists.iptel.org] Em nome de Mark Aiken
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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