[Asterisk-Users] Sipura g729 call quality to PSTN
Pedro
traci.asterisk at gmail.com
Wed Feb 16 15:20:19 MST 2005
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
> > > > > > >
> > > > > > >
> > > > > > _______________________________________________
> > > > > > Asterisk-Users mailing list
> > > > > > Asterisk-Users at lists.digium.com
> > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > > > To UNSUBSCRIBE or update options visit:
> > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > > >
> > > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
>
More information about the asterisk-users
mailing list