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

[Kristian Kielhofner]

On Dec 24, 2007 5:59 AM, Andreas Brodmann <andreas.brodmann at gmail.com> wrote:
>
> 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
>

Andreas,

 The possibility of app_page enhancements and (maybe) a Page
pseudo-channel intrigues me...

-- It would be simpler to mix/match multiple device support.  For
instance, you could build a page command line as follows:

exten => page,1,Page(Page/group1&Page/group2&Page/group3&SIP/105&Zap/4)

where page.conf could look like:

[group1]
method = multicast_rtp ; SNOM style RTP spray
address = 233.64.133.10:7000/1 ; (IP:port/ttl)

[group2]
method = unicast_rtp ; unicast INVITEs to each recipient with
multicast address in SDP for media (c=) -  RFC compliant but a little
more work for Asterisk
member => SIP/polycom1
member => SIP/grandstream1
member => SIP/generic1

[group3]
method = linksys_marker ; whatever they need, some variation of SNOM
member => SIP/linksys1
member => SIP/linksys2

  This way you could page multiple SIP device types using entries from
page.conf while specifying additional technologies on the command
line.  There would be no reason why you couldn't specify additional
methods that didn't use SIP at all - you could have methods for any
channel type in page.conf.  Ideally, you would only have to mix audio
once for each stream type.

  This would be awesome.  Now the real question. Is it practical?
Would we need to rewrite Asterisk to support it?  ;) Probably not, but
these are the things I'd like to address before we head down this
road.

-- 
Kristian Kielhofner