[asterisk-users] Urgent Help Required

[ali anjum]

first of all thanks to all for help..............actually I want to increase the packetization time to save bandwidth actually I want to run the voip on a medium where I have to use only 15kbs data for placing a call i have calculated manually on a paper and also checked many forums a little bit delay is affordable(I have calculated required bandwidth using speex at 2.15kbs and packetization time 60ms it should consume maximum of 10kbs bandwidth on these conditions)

because speex is an open source and configurable codec according to your bandwidth requirements.........any further help from your side would be really helpfull




> From: fred at teamforrest.com
> Date: Thu, 4 Nov 2010 09:46:57 -0400
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Urgent Help Required
> 
> On Nov 4, 2010, at 9:41 AM, C F wrote:
> 
> > You see the problem is that asterisk will send as many packets as its
> > admin does on the list. There is no way to change that. I suggest you
> > first change the amount of packets per second you send.
> > 
> > On Thu, Nov 4, 2010 at 5:38 AM, ali anjum <aliraza_anjum at hotmail.com> wrote:
> >> Hi,
> >> 
> >> (I have install trixbox2.8 with asterisk 1.6)
> >> I am using speex codec for my Inter asterisk communication
> >> 
> >> Question1: I want to configure speex on 2.15kbs and packetization of 60ms
> >> seconds for that is have configured "codecs.conf" for desired result and
> >> also placed a line in general section of "sip.conf" allow=speex:60 after
> >> disallow=all line .
> >> 
> >> I have also configure SIP trunk between two asterisk to use speex:60
> >> During debugging I have checked that both side accept speex as a codec for
> >> call and ptime:60 but
> >> 
> >> I am facing following unexpected results
> >> 
> >> 1-> When I check the packet rate from one asterisk to other asterisk for one
> >> call its not (1000/60 == 17)?
> >> 
> >> 2-> When ever I change the softphone result changes i.e. data ratae chages ?
> >> 
> >> 3-> How can I use my own codec "xyz" in asterisk to place calls ?
> >> 
> >> 4->if I change the codecs.conf then no results appears in packet size which
> >> is comming out of asterisk?
> 
> Out of curiosity, is there a reason why you want to exceed 20ms?
> 
> The nice thing about 20ms is that in theory you can drop a packet and not notice (audibly). Over 20ms is supposed to be noticed by the human ear.
> 
> This being said, doc/rtp-packetization will describe the acceptable payload sizes for different codecs.
> 
> ---fred
> http://qxork.com
> -- 
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