Re: [Asterisk-Users] C7960 g729 question
Not sure I understood the comments... the existing situtation (as shown in the original email) is a remote 7960 uses g729, and an internal 7960 that uses g711 (both forced via sip.conf entries). The audio is oftentimes choppy, "but" changing the g729 side to g711 (still passing through * with canreinvite=no) clears up the audio. Same exact path in both cases without a doubt. CPU is 2.2ghz, 100 meg full duplex, <2% utilization during a "single" call in the * system. Logic would suggest the g711-to-g711 (with canreinvite=no) should have been a lower quality then the g729-to-g711 (due to any form of bandwidth contraints that might impact either approach). But, its the other way around; the g729-to-g711 is choppy. I'm searching for "where" to go look for the problem and the only logical conclusion that I've come up with is that it appears to be something associated with the transcoding. Thoughts? Rich > Have you checked the CPU load on your asterisk server? > > If you have no change of codec between the two sides, asterisk just > pasts the audio straight through. If there is a change, conversion has > to take place and conversion from g729 to anything else does require cpu > power. > > Another thing you could do however is allow the C7960 to use g729 inside > sip.conf just not as the first codec, that way it will use g711 but pass > on calls direct if g729. > > Hope that helps > > Matthew Enger > [EMAIL PROTECTED] > > > On Sat, 2004-06-19 at 01:12, Rich Adamson wrote: > > I have multiple voiceage g729 licenses installed on a RH9 box, and have > > a remote C7960 configured to use it (low bandwidth). In calls like: > > > > Remote C7960 -> g729 -> asterisk -> g711 -> C7960 > > > > the audio is oftentimes rather choppy. Changing the remote 7960 to use > > g711 seems to eliminate/reduce the choppyness. Any ideas on what might > > be behind this? > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Matthew Enger <[EMAIL PROTECTED]> > Xintegration > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] C7960 g729 question
Hello, Have you checked the CPU load on your asterisk server? If you have no change of codec between the two sides, asterisk just pasts the audio straight through. If there is a change, conversion has to take place and conversion from g729 to anything else does require cpu power. Another thing you could do however is allow the C7960 to use g729 inside sip.conf just not as the first codec, that way it will use g711 but pass on calls direct if g729. Hope that helps Matthew Enger [EMAIL PROTECTED] On Sat, 2004-06-19 at 01:12, Rich Adamson wrote: > I have multiple voiceage g729 licenses installed on a RH9 box, and have > a remote C7960 configured to use it (low bandwidth). In calls like: > > Remote C7960 -> g729 -> asterisk -> g711 -> C7960 > > the audio is oftentimes rather choppy. Changing the remote 7960 to use > g711 seems to eliminate/reduce the choppyness. Any ideas on what might > be behind this? > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger <[EMAIL PROTECTED]> Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] C7960 g729 question
My sip.conf for this "single" extension, looks like: [3014] username=3014 ...snip... canreinvite=no disallow=all allow=g729 ...snip... which works fine, and does force the use of g729 as wanted. That's not the issue here. The issue is: once a call is established, it appears the g729 -> g711 transcoding seems to be choppy while g711 -> g711 is much better. In a somewhat limited bandwidth situation, I would have expected the other way around; g711 should have been worse then g729 transcoding. That implies a trancoding issue might be lurking. Thoughts? > What does your sip.conf look like? Always make sure that you have the > following codec order for G.729 pass-thru: > > [general] > disallow=all > allow=g729 > allow=ulaw > allow=alaw > > you don't need to force your C7960 (SIP settings) to use G.729 with the > above config. > > see also: > http://www.voip-info.org/tiki-index.php?page=Asterisk%20G.729%20pass-thru > > Dominique > > Rich Adamson wrote: > > > I have multiple voiceage g729 licenses installed on a RH9 box, and have > > a remote C7960 configured to use it (low bandwidth). In calls like: > > > > Remote C7960 -> g729 -> asterisk -> g711 -> C7960 > > > > the audio is oftentimes rather choppy. Changing the remote 7960 to use > > g711 seems to eliminate/reduce the choppyness. Any ideas on what might > > be behind this? > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Dominique Kull > The Old Lodge, London SW6 6EE UK > t: +44 207 731 1562 > v: fwd 268167 > e: [EMAIL PROTECTED] > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] C7960 g729 question
What does your sip.conf look like? Always make sure that you have the following codec order for G.729 pass-thru: [general] disallow=all allow=g729 allow=ulaw allow=alaw you don't need to force your C7960 (SIP settings) to use G.729 with the above config. see also: http://www.voip-info.org/tiki-index.php?page=Asterisk%20G.729%20pass-thru Dominique Rich Adamson wrote: I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dominique Kull The Old Lodge, London SW6 6EE UK t: +44 207 731 1562 v: fwd 268167 e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
