[asterisk-dev] Reducing RTP overhead

[asterisk at ntplx.net]

>From what I was reading in the RFC...

CRTP compresses the header by removing most of it. This is done on point
to point links where the routers know the info in the header and then
remove it over the link and add it back in on the other side. So after
compression there is not enough left in the header to route the packet
by its self. Also if the routers loose packets they may drop even
more data and the resync the compressed stream.

So Asterisk can't send CRTP packets to a non-local phone. If the phone
is local then bandwidth should not be a problem....

Send more voice data in a packet so the overhead is less and don't use
UDP checksums (saves 2 more bytes).

For overhead (SIP per call RTP streams):
40 bytes header, 20 bytes voice data (20ms G.729) does not look good.
40 bytes header, 160 bytes voice data (20ms G.711/G.722) looks ok.

If you up it to 40ms then for G.729 it's 50% header, 50% data.

Some systems run 10ms, but that's nuts (IMHO), 20ms is "standard" and
40ms is not too bad for jitter. 60-80ms is a bit long in systems that
have additional delay (ie, anything non-local).



Quoting Kristian Kielhofner <kris at krisk.org>:

> 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
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