Re: [Asterisk-Users] RTP frame size location?
On 03/29/06 13:06 Andres said the following: It works perfectly with other values we have tested of 40 and 60. We currently use 60 on all our servers. It cuts down on bandwidth for a G279 call to about 15Kbps. with 60ms packets, is a packet loss or two noticable ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP frame size location?
Dinesh Nair wrote: On 03/29/06 13:06 Andres said the following: It works perfectly with other values we have tested of 40 and 60. We currently use 60 on all our servers. It cuts down on bandwidth for a G279 call to about 15Kbps. with 60ms packets, is a packet loss or two noticable ? Depends on the UA. The ones we use are all Sipura/Linksys. We have measured up to 6% packet loss and the call still sounds pretty good (cell phone quality). A 1-2% packet loss will not degrade the call at all. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP frame size location?
Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to force my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a peer-by-peer or user-by-user fashion, but I'm just looking for a quick hack right now.) Anyone know where this is in rtp.c or other RTP-associated files? JT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP frame size location?
Check out bugid 5162 on Mantis. It allowed per peer/user packetization settings. It is in need of much love, but as-is should not be too far from applying to the 1.2.X series. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Tuesday, March 28, 2006 2:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RTP frame size location? Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to force my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a peer-by-peer or user-by-user fashion, but I'm just looking for a quick hack right now.) Anyone know where this is in rtp.c or other RTP-associated files? JT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP frame size location?
John Todd wrote:
Google has given me too many responses, so I'll ask the list:
Where in the Asterisk rtp source code can I find the default
millisecond frame size? I've looked around for obvious pointers, but
it's not clear. I'd like to force my Asterisk server to use a
certain frame size all the time. (Of course, ideally I'd like to
prefer or even force that frame size in a peer-by-peer or user-by-user
fashion, but I'm just looking for a quick hack right now.)
Anyone know where this is in rtp.c or other RTP-associated files?
You can force that by codec in rtp.c. For example for G729 find the
following part of the code:
case AST_FORMAT_G729A:
if (!rtp-smoother) {
rtp-smoother = ast_smoother_new(20);
This 20 is the fame size in milliseconds, just change it to something else.
It works perfectly with other values we have tested of 40 and 60. We
currently use 60 on all our servers. It cuts down on bandwidth for a
G279 call to about 15Kbps.
--
Andres
Technical Support
http://www.telesip.net
___
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