[Asterisk-Users] Sipura g729 call quality to PSTN

[Pedro]

Actually - jitter does not seem to be the issue (sound is not garbled
and does not drop out, it was just very low and "fuzzy"/"staticy" when
not set to 10 ms).

It is weird that I have to drop to 10ms, but I have tested some more
and the general consenses from the people I have called said it sounds
fine now with 10ms setting.

Thanks for your help though.

Here is the result set from the ethereal trace using 10ms (RTP stream
sent from Sipura to asterisk):

RTP      Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604121
RTP      Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604201

As you can see there is now a difference of 80 between the Time stamps
 (now to sound dumb, but it would be 80 what?)


On Wed, 16 Feb 2005 19:42:19 -0700, Keith Burns
<kburns at porchlightcom.com> wrote:
> Hmmm, that worked?
> 
> Interesting that you can change the sample size to 10ms since the "standard"
> is 20ms that most people don't go below. I know you *can* do below 20 but if
> you are doubt the technical ability of the box it seems strange they are
> capable of that.
> 
> This seems to smack of bad de-jitter buffers on the egress gateway... are
> you receiving 20ms sampled RTP ?
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of Pedro
> > Sent: Wednesday, February 16, 2005 3:20 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
> >
> > FYI - Seems the latest firmware in conjunction with changing the
> > packet size to 10ms improved the call quality to usable.  The Cisco
> > 7960 is stell superior, but now at least the SPA-2100 is acceptable
> > (and with 2 working g729 channels including 3-way calling).
> >
> >
> > On Wed, 16 Feb 2005 15:44:58 -0500, Pedro <traci.asterisk at gmail.com>
> wrote:
> > > Forgot to mention that when I set the RTP Packet Size to 20ms that the
> > > difference was 160 (like the Cisco) but call quality was much worse.
> > >
> > >
> > > On Wed, 16 Feb 2005 15:36:44 -0500, Pedro <traci.asterisk at gmail.com>
> wrote:
> > > > 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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