[Asterisk-Users] Sipura g729 call quality to PSTN

[Keith Burns]

Next thing I would check are the de-jitter buffers if possible on the
Sipura, or jitter in general.

Do you have control of the PSTN gateway ? Measure the jitter on ingress to
the gateway. You can do this crudely by using Ethereal and looking at the
delta between timestamps on RTP packets from Sipura to PSTN.



> -----Original Message-----
> From: Pedro [mailto:traci.asterisk at gmail.com]
> Sent: Wednesday, February 16, 2005 1:37 PM
> To: Keith Burns
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Jeffrey Chan
> Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
> 
> Thanks for the suggestion.  Changing the RTP Packet Size in the Sipura
> to 40ms did improve the call quality "slightly", but still well below
> par compared to the Cisco 7960.
> 
> In my ethereal captures, I did notice something interesting.  While
> the RTP stream from the Cisco to asterisk seemed to have a 160
> diffference in timestamps, the Sipura showed a 320 difference:
> 
> Cisco:
> RTP      Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094,
> Time=40666896
> RTP      Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095,
> Time=40667056
> 
> Sipura:
> RTP      Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784,
> Time=434932771
> RTP      Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784,
> Time=434933091
> 
> 
> On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns
> <kburns at porchlightcom.com> wrote:
> > What is your sample size?
> >
> > I believe the 7960 supports 40ms (2 samples) per packet by default.
> >
> > Do you have an ethereal trace? Look at the timestamps between RTP
packets if
> > you can't see/modify this setting.
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > > bounces at lists.digium.com] On Behalf Of Pedro
> > > Sent: Tuesday, February 15, 2005 6:30 PM
> > > To: Jeffrey Chan
> > > Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
> > >
> > > Actually the SPA-2100 supports 2 g729 channels which is why I bought
> > > it.  Unfortunately, the call quality is just as poor on the 2100 as it
> > > is on the 2000.
> > >
> > > - Pedro
> > >
> > >
> > > On Tue, 15 Feb 2005 23:12:51 +0000, Jeffrey Chan
<mutualphone at gmail.com>
> > > wrote:
> > > >  Is it just a bad implementation of g729 compression with the Sipura
> > > > > > > product line?
> > > > > > >
> > > > > >
> > > >  That would be my guess too . why SPA-2000 supports G729 for one
> > > > channel only? no enough CPU power to code/decode G.729 for two
> > > > channels?
> > > >
> > > > Jeffey
> > > >
> > > > www.mutualphone.com
> > > >
> > > >
> > > > On Tue, 15 Feb 2005 16:31:59 -0500, Pedro <traci.asterisk at gmail.com>
> > wrote:
> > > > > uggg.
> > > > >
> > > > > Is anyone out there having any luck with the SPA-2000 or SPA-2100
> > > > > using the g729 codec with decent call quality?
> > > > >
> > > > >
> > > > > On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler <mark at mixtur.com>
> > wrote:
> > > > > >
> > > > > > On Feb 14, 2005, at 1:25 PM, Pedro wrote:
> > > > > >
> > > > > > >
> > > > > > > Is it just a bad implementation of g729 compression with the
> > Sipura
> > > > > > > product line?
> > > > > > >
> > > > > >
> > > > > > That would be my guess.
> > > > > >
> > > > > > -mark
> > > > > >
> > > > > > --
> > > > > > Mark Eissler, mark at mixtur.com
> > > > > > Mixtur Interactive, Inc. - at - http://www.mixtur.com
> > > > > >
> > > > > >
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