[asterisk-users] Multicast RTP Paging

[Krishna Sumanth Chava]

HI Guys,

I am trying to use the RTPPage application on asterisk 1.4 using the Snom
320's?? My goal is to do the paging using a multicast IP address.

I tried the app_rtppage.c and i can only do unicast on the snom's and i was
unable to do a multicast.

https://issues.asterisk.org/view.php?id=11797
http://svnview.digium.com/svn/asterisk?revision=101218&view=revision

My dialplan command is as below.

exten => 1234,1,RTPPage(basic|224.1.1.1:7000|ulaw|ef)

i have the same IP/Port to be listened on for multicast traffic on the Snom
320's. But when i make a call to 1234, the snom 320 does not get answered at
all.

If i use the same command and the IP of the Snom instead of the multicase
IP, i was able to have the snom auto answer the call on Speaker.

I would like get assistance from the community in this issue.

Thanks as always

Regards
Krishna
On Wed, May 13, 2009 at 9:21 AM, Joshua Colp <jcolp at digium.com> wrote:

> Hello everyone,
>
> A month ago I took on an issue on the Asterisk issue tracker (
> https://issues.asterisk.org/view.php?id=11797) dealing with multicast RTP
> paging.
>
> This is the ability to send audio to phones (the phone must support it) and
> have it played out the speakerphone. Using multicast RTP is great for
> this because it does not incur the cost and weight of setting up a
> potentially short call. Depending on the setup this can actually get to be
> quite
> a big problem because when you involve phones subscribed to the state of
> another they get told that the phone is in use. The amount of SIP traffic
> can
> just spiral out of control.
>
> Originally this issue was filed with a new application that performed the
> paging. I took this application and turned it into a channel driver. This
> means
> that instead of having a dedicated paging application for it you can just
> use Dial(). This also means that in mixed environments you can use the
> Page()
> application along with other phones that do not support the multicast RTP
> paging.
>
> So far I have gotten very little response on the issue so I am asking
> anyone on this mailing list who is interested and has the time to test to
> please test
> and provide some feedback.
>
> A branch based off of trunk (as that is where the channel driver will go)
> is available at http://svn.asterisk.org/svn/asterisk/team/file/issue11797
>
> The dial string for the channel driver is in the form of
> MulticastRTP/<type>/<destination>/<control address> where type is either
> basic or linksys. The
> control address is only needed for the linksys type.
>
> Any feedback is welcome as a note on
> https://issues.asterisk.org/view.php?id=11797 and will help to getting
> this into the tree.
>
> Thanks!
>
> --
> Joshua Colp
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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