[Asterisk-Users] RTP header compression?

[Kevin K]

Uh oh.  I think I may be looking at the wrong tool.

My goal is to implement (in an open source software suite) an RTP/UDP/IP 
header compression algorithm that would save bandwidth used by voice traffic 
packets.  So a 5ms G.711 packet that would otherwise be 98 bytes, could be 
reduced to 62 bytes when its RTP/UDP/IP header is compressed.  This would 
require me to get into the RTP stack used to packetize the actual data 
(voice) packets.  From what I am reading, * is not the right software for 
that kind of work because it don't really touch the voice packets themselves 
but rather enables the two endpoints to talk to negotiate and connect with 
each other (as well as utilize a number of other features - vmail, callerID, 
etc).  Is this correct?

Is what I am looking for an open-sourced SIP user agent?  Or is it more than 
that?

Any help would be great, as I appear to be confused.

Thanks,
Kevin

----Original Message Follows----
From: John Todd <jtodd at loligo.com>
Reply-To: asterisk-users at lists.digium.com
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] RTP header compression?
Date: Wed, 20 Aug 2003 13:38:22 -0700

>I sent this to the asterisk-dev by accident...
>
>----Original Message Follows----
>
>Hi all,
>
>I have a couple questions about RTP header compression with Asterisk:
>
>1) Has this been implemented before or is it included in the Asterisk 
>package?
>2) If the answer to (1) is no, is there an RTP stack that this can be 
>logically implemented into?  Where would that be?
>
>Thanks,
>Kevin


As I mentioned before in a reply to this:

1) No.
2) Take a look at the source code; you should be able to find the rtp 
sections in it readily enough.

3) If you are asking about RTP compression between Asterisk servers, look at 
IAX2 and trunking mode.  Questions 1 and 2 become less meaningful.

JT
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