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 <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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