[asterisk-dev] Reducing RTP overhead

[Kristian Kielhofner]

Derek Smithies wrote:
> Hi,
>  the simplest method to reduce network usage is put more than 1 compressed 
> sound frame into each udp packet.
> 
> This does increase the latency, but it is not noticable. There is already 
> a 200 msec transmit time, 200ms jitter buffer.
>  An extra 40ms cause there are an extra is 2 gsm frames in an audio packet 
> will not be significant.
> 
> The chart at:
> http://www.voxgratia.org/docs/codecbw.html
> 
> illustrates this point.
> 
> I am told that IAX2 does have the option of allowing you to put more audio 
> frames into each packet.
> I know that H.323 and SIP have been doing this for years also.
> 
> Derek.
> 

Derek,

	Great, thanks for the link!  However, I am not looking for the simplest 
solution, but rather the "best" (after weighing all of the pros and 
cons, obviously).  Increasing latency (if there are other ways) does not 
seem like the best approach.

	It's good that you brought this up, though.  I'd like to see this type 
of functionality in Asterisk also! :)	

-- 
Kristian Kielhofner