[asterisk-dev] Reducing RTP overhead
John Todd
jtodd at loligo.com
Mon Feb 13 00:10:06 MST 2006
>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
There is the protocol called RTPC (RFC2508) which is rumored to
compress ~40 byte IP/UDP/RTP headers down to <5 bytes when used with
RFC2509. I haven't seen it in in action other than on T1 connections
point-to-point, and I don't know what the requirements are for
intermediate systems to understand and route the IP layer datagrams
with the shortened packet data. I have only been a witness to what I
was told was a compressed link, and I've not actually set it up
myself. Cisco seems to be the only player in this field currently,
though it is an open standard.
Interleaving data into single, larger packets MUST be the best way
to do it with VoIP, though I read that there are "drafts"
(un-discoverable) about a better way called Enhanced CRPT (ECRPT)
which describe a better way to do it with VoIP packets - links,
anyone?
http://www.faqs.org/rfcs/rfc2508.html
http://www.faqs.org/rfcs/rfc2509.html
http://www.opalsoft.net/qos/VoIP.htm
There may be a typo on some Grandstream literature that was perhaps
supposed to say "RTCP" instead of "RTPC", but there is an off-chance
that the Budgetones support this protocol (though it is almost
certainly "RTCP" which is a whole different bugtracker subject.)
JT
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