[Serusers] Inserting SER into my voice network

[Mark Aiken]

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 at 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 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* <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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