Here's what is in mine verbatim.

Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it.

When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful.

Here's the online version : http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html

It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find.

Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached:

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
; These are the addresses where your system will RECEIVE audio and video stream$
; If you have connections across a firewall, make sure that these are open.
;

rtpstart=10000
rtpend=10999

And here's a link to a page concerning rtp.conf : http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf

Hope this helps.