[asterisk-dev] Asterisk Multicast RTP for paging - bounty?

[Kristian Kielhofner]

Hello,

  I'm very interested in a multicast RTP implementation for Asterisk.
I'm having some paging problems with app_page on smaller systems.
Having to setup a meetme, a new SIP channel for each participant, and
handle all of the RTP for EVERY RTP seems a little unnecessary.

  Newer Snom (firmware v7), Linksys, and (maybe) Polycom phones
support RTP multicast.  A phone can listen on a multicast address and
port.  It will then play any RTP that arrives there.  I have tested
this with pbxnsip and a Snom 360 and Snom 300.  It works quite well.
I sniffed the traffic on the multicast group and there is no SIP
session setup.  No INVITE, nothing.  Just good 'ol spray and pray UDP
RTP to a multicast group and port.  The Snoms then display "PA" on the
display and turn on their speaker.  It works quite well.

  My question is this...  How tough would it be to integrate this
functionality into Asterisk?  I'm thinking of a few different ways:

1)  Add-on to app_page to just spray rtp.  This might be the simplest
(and most hackish).

2)  Maybe a chan_rtp?:

exten => page,1,Page(SIP/102,RTP/233.64.133.10:7000,IAX2/remote,q)

  Does that make any sense?

3)  Avoid Asterisk all together...  Not really an Asterisk question,
but what about a standalone app...  Use a SIP library to enable it to
setup the initial session (handle INVITE from anywhere - even Asterisk
on the same machine if running on a different port) and standard
unicast RTP stream from any SIP device.  Then take that stream and
send it out RTP multicast based on configuration or (ideally) info
from the initial SIP invite (To:, custom header, etc)

  Any ideas from Asterisk devs?  Thanks!

-- 
Kristian Kielhofner