[asterisk-users] Urgent Help Required

[Fred Posner]

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