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

[Andreas Brodmann]

2007/12/24, Kristian Kielhofner <kristian.kielhofner at gmail.com>:
>
> 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.


I have a working prototype for the group1 stuff. group3 isn't much
additional
work. Haven't looked at group2 style in depth yet. Could you test it with
the snoms, if I send you the app?
(it's just copying it into the 1.4.x source tree and recompile
asterisk from scratch).

-Andreas
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