Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because
of bandwidth failure.
?
thanks in advance
Mani
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm
Igor Goncharovsky
2008-Nov-10 11:30 UTC
[asterisk-users] changing the size of voice packets
Hi! On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali <pezhman_lali at yahoo.com>wrote:> Dear, > is any way to change , the size of voice packets? > I want to increase the quality by decreasing the size of each packets, > because of bandwidth failure. >You can specify size of voice packets in allow line of sip.conf peer configuration. ex.: allow=alaw:30,g729:50 For more information: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt -- Best regards, Igor Goncharovsky -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/b9919169/attachment.htm
Igor Goncharovsky wrote:> You can specify size of voice packets in allow line of sip.conf peer > configuration. > ex.: allow=alaw:30,g729:50If you are interfacing with any commercial VoIP equipment/media gateway equipment, do be aware that packetisation durations for G.711u and G.729 that are not 10, 20, or 30 ms are not widely supported, and it may be that 20 ms is the only duration supported -- the most common. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
--- On Mon, 11/10/08, Igor Goncharovsky <igor.goncharovsky at gmail.com>
wrote:
From: Igor Goncharovsky <igor.goncharovsky at gmail.com>
Subject: Re: [asterisk-users] changing the size of voice packets
To: pezhman_lali at yahoo.com, "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Date: Monday, November 10, 2008, 3:00 PM
Hi!
On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali <pezhman_lali at yahoo.com>
wrote:
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because
of bandwidth failure.
You can specify size of voice packets in allow line of sip.conf peer
configuration.
ex.: allow=alaw:30,g729:50
thanks for your great help
2 questions
1- in allow=g729:50
50 is time or size of packet
?for smaller packet, 40 or 60 ?
2- what about for asterisk 1.2
For more information:
http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt
--
Best regards,
Igor Goncharovsky
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/08482c6c/attachment.htm
Kristian Kielhofner
2008-Nov-10 19:41 UTC
[asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 2:40 PM, Kristian Kielhofner <kkielhofner at star2star.com> wrote:> > "Depends?" > > What is the status of maxptime in Asterisk? >... or the remote end, for that matter... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
Well, I should say "irreconcilably different," not "different." On Mon, November 10, 2008 2:40 pm, Kristian Kielhofner wrote:> On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov > <abalashov at evaristesys.com> wrote: >> >> If the packetisation durations are different between endpoints, the SDP >> offer/answer should fail with a "488 Not Acceptable Here." Right? >> > > "Depends?" > > What is the status of maxptime in Asterisk? > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
is any command , shows the current rate of each channel? ? --- On Mon, 11/10/08, Kristian Kielhofner <kkielhofner at star2star.com> wrote: From: Kristian Kielhofner <kkielhofner at star2star.com> Subject: Re: [asterisk-users] changing the size of voice packets To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Date: Monday, November 10, 2008, 11:10 PM On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov <abalashov at evaristesys.com> wrote:> > If the packetisation durations are different between endpoints, the SDP > offer/answer should fail with a "488 Not Acceptable Here."Right?>"Depends?" What is the status of maxptime in Asterisk? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081111/02d41199/attachment.htm