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<br>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@2.15kbs and packetization time 60ms it should consume maximum of 10kbs bandwidth on these conditions)<br><br>because speex is an open source and configurable codec according to your bandwidth requirements.........any further help from your side would be really helpfull<br><br><br><br><br>> From: fred@teamforrest.com<br>> Date: Thu, 4 Nov 2010 09:46:57 -0400<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] Urgent Help Required<br>> <br>> On Nov 4, 2010, at 9:41 AM, C F wrote:<br>> <br>> > You see the problem is that asterisk will send as many packets as its<br>> > admin does on the list. There is no way to change that. I suggest you<br>> > first change the amount of packets per second you send.<br>> > <br>> > On Thu, Nov 4, 2010 at 5:38 AM, ali anjum <aliraza_anjum@hotmail.com> wrote:<br>> >> Hi,<br>> >> <br>> >> (I have install trixbox2.8 with asterisk 1.6)<br>> >> I am using speex codec for my Inter asterisk communication<br>> >> <br>> >> Question1: I want to configure speex on 2.15kbs and packetization of 60ms<br>> >> seconds for that is have configured "codecs.conf" for desired result and<br>> >> also placed a line in general section of "sip.conf" allow=speex:60 after<br>> >> disallow=all line .<br>> >> <br>> >> I have also configure SIP trunk between two asterisk to use speex:60<br>> >> During debugging I have checked that both side accept speex as a codec for<br>> >> call and ptime:60 but<br>> >> <br>> >> I am facing following unexpected results<br>> >> <br>> >> 1-> When I check the packet rate from one asterisk to other asterisk for one<br>> >> call its not (1000/60 == 17)?<br>> >> <br>> >> 2-> When ever I change the softphone result changes i.e. data ratae chages ?<br>> >> <br>> >> 3-> How can I use my own codec "xyz" in asterisk to place calls ?<br>> >> <br>> >> 4->if I change the codecs.conf then no results appears in packet size which<br>> >> is comming out of asterisk?<br>> <br>> Out of curiosity, is there a reason why you want to exceed 20ms?<br>> <br>> 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.<br>> <br>> This being said, doc/rtp-packetization will describe the acceptable payload sizes for different codecs.<br>> <br>> ---fred<br>> http://qxork.com<br>> -- <br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
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