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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