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

[Andreas Brodmann]

2007/12/20, Kristian Kielhofner <kristian.kielhofner at gmail.com>:
>
> 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.


I am currently working on a dialplan app that does just about that.

If you could provide me specs/info about how your phones
expect the data (codec, packetization time) etc, I'll try to keep the
app as generic as possible.

The linksys phones do expect a start/stop signal to accept this
paging feature as far as I found out by capturing traffic. Unfortunately
Linksys didn't bother letting me have the data/information on how
these packets have to be formated.

The Cisco 79xx series phones do have a similar feature. You do have
to authenticate against each phone's internal webserver though to
have them listen to your multicast traffic.

If anyone has information about similar devices, be it phones or
automation systems like the barix.com series devices, which can
handle rtp streams, please let me know.

-Andreas
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