asterisk users - Aug 2003 - RTP header compression?

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

_________________________________________________________________
<b>Get MSN 8</b> and enjoy automatic e-mail virus protection.  
http://join.msn.com/?page=features/virus

>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

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@loligo.com>
Reply-To: asterisk-users@lists.digium.com
To: asterisk-users@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
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

_________________________________________________________________
The new MSN 8: smart spam protection and 2 months FREE*  
http://join.msn.com/?page=features/junkmail