[asterisk-dev] Reducing RTP overhead

[Kristian Kielhofner]

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!

-- 
Kristian Kielhofner