If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side (the audio on the phone connected to the Sipura sounds fine). My guess is that the Sipura does not compress the outbound audio very effectively and since the incoming audio from the PSTN is already compressed by the VoIP provider, it is just delivering the good-sounding g729 stream. It is worth noting that call quality on both the IP and PSTN side is great when using the Cisco 7960 with g729. It is just with the Sipura that the sound quality on the PSTN-side sounds like a bad quality cell phone call. I even got an SPA-2100 in hopes that the g729 would sound better on that unit, but the same issue is present there as well. Is it just a bad implementation of g729 compression with the Sipura product line? Any thoughts or recommendations are appreciated :) Thanks! - Pedro
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@mixtur.com Mixtur Interactive, Inc. -@- http://www.mixtur.com
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@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@mixtur.com > Mixtur Interactive, Inc. -@- http://www.mixtur.com > >
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@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@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@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@mixtur.com > > > Mixtur Interactive, Inc. -@- http://www.mixtur.com > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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 > > >
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@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@lists.digium.com [mailto:asterisk-users- > > bounces@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@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@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@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@mixtur.com > > > > > Mixtur Interactive, Inc. -@- http://www.mixtur.com > > > > > > > > > > > > > > _______________________________________________ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@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@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 > >
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@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@lists.digium.com [mailto:asterisk-users- > > bounces@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@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@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@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@lists.digium.com > [mailto:asterisk-users- > > > > > > bounces@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@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@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@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@mixtur.com > > > > > > > > > Mixtur Interactive, Inc. -@- http://www.mixtur.com > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > Asterisk-Users mailing list > > > > > > > > Asterisk-Users@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@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@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 > >