[asterisk-dev] Reducing RTP overhead

Derek Smithies derek at indranet.co.nz
Sun Feb 12 20:13:50 MST 2006


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.


===============================================================
On Sun, 12 Feb 2006, Kristian Kielhofner wrote:

> Hello everyone,
> 
> 	A long, long time ago someone somewhere told me about a feature with 
> RTP to reduce overhead (and bandwidth).  By trunking I mean trunking as 
> in "IAX trunking" - stuffing multiple voice channels into the same UDP 
> packet on connections between the same systems to reduce UDP/etc 
> overhead (you know what I am talking about). :)
> 
> 	I was told that such a feature exists for use with RTP.  The closest 
> thing that I have been able to find is RTP header compression (RFC 
> 2508).  With the calculations that I have seen, RTP header compression 
> can increase call capacity by over %50:
> 
> http://www.connect802.com/voip_bandwidth.php
> 
> 	Are there any other ways to improve bandwidth usage with SIP/RTP? 
> Perhaps something more like IAX trunking?  RFC 2508 appears to only 
> apply to PtP serial links (it also compresses the IP header, but that 
> may be optional).  I'll continue to read the spec.
> 
> 	Lets just say that RFC 2508 (or something like it) is the only way to 
> reduce RTP bandwidth usage.  I have several questions:
> 
> 	If an ideal implementation for Asterisk was created, would it stand a 
> chance of being put in CVS?  What equipment/vendors also support it? 
> This is key.  If I'm using just Asterisk (I wish) I would just use IAX2!
> 
> Thanks!
> 
> 

-- 
Derek Smithies Ph.D.                 Any fool can write code that 
IndraNet Technologies Ltd.                a computer can understand.        
Email: derek at indranet.co.nz         Good programmers write code 
ph +64 3 365 6485                          that humans can understand.
Web: http://www.indranet-technologies.com/            Martin Fowler




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