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 <[EMAIL PROTECTED]> 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 > <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:asterisk-users- > > > [EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> > > > 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 <[EMAIL PROTECTED]> > > 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 <[EMAIL PROTECTED]> > > 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, [EMAIL PROTECTED] > > > > > > Mixtur Interactive, Inc. [EMAIL PROTECTED] 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