[asterisk-users] Request for feedback/testing on Multicast RTP Paging

[Joshua Colp]

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