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

[Johansson Olle E]

20 dec 2007 kl. 20.42 skrev 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).

Since there's no signalling involved at all, not "call setup", I would
take this the same route as the icecast integration - in an application.

I think the SNOM phones also can take multicast as a source
for music on hold.

/O