[asterisk-bugs] [Asterisk 0011797]: [patch] app_rtpstream: Application to Page Multicast capable receivers (e.g. Snom, Linksys, Cisco, Barix devices)

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Apr 13 08:43:38 CDT 2009


The following issue is now READY FOR TESTING. 
====================================================================== 
http://bugs.digium.com/view.php?id=11797 
====================================================================== 
Reported By:                macbrody
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11797
Category:                   Applications/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 99188 
Request Review:              
====================================================================== 
Date Submitted:             2008-01-19 04:46 CST
Last Modified:              2009-04-13 08:43 CDT
====================================================================== 
Summary:                    [patch] app_rtpstream: Application to Page Multicast
capable receivers (e.g. Snom, Linksys, Cisco, Barix devices)
Description: 
app_rtpstream is an application that reads the input channel's voice frames
and does rtp stream them to either unicast or multicast addresses defined
as groups in the config file.

This can be used for example with the Snom and Linksys IP Phones' feature
to do paging to multicast receivers.
====================================================================== 

---------------------------------------------------------------------- 
 (0103136) file (administrator) - 2009-04-13 08:43
 http://bugs.digium.com/view.php?id=11797#c103136 
---------------------------------------------------------------------- 
I've created a branch at
http://svn.digium.com/svn/asterisk/team/file/issue11797 to bring this work
to completion based on some new things available in trunk.

The multicast RTP paging has now been turned into a module that is
available to any developer through the RTP engine API.

I have also created a channel driver that uses this to page much like the
attached application does. This was done so it could transparently be used
with the existing Page() application so that mixed environments can work
equally the same. That does not mean you have to use the Page application,
you can use Dial if you wish.

I would appreciate some testing on this branch.

The format for the dial string is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys
The control address is optional and only used for the linksys type 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-13 08:43 file           Note Added: 0103136                          
2009-04-13 08:43 file           Status                   assigned => ready for
testing
======================================================================




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