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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11797 
====================================================================== 
Reported By:                macbrody
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11797
Category:                   Applications/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 99188 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             01-19-2008 04:46 CST
Last Modified:              01-23-2008 08:52 CST
====================================================================== 
Summary:                    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.
====================================================================== 

---------------------------------------------------------------------- 
 macbrody - 01-23-08 08:52  
---------------------------------------------------------------------- 
yes, I suggest though that we then make two applications from it, like:

1) RTPPage_Multicast(group1[&group2][, codec])
2) RTPPage_Unitcast(ip1 & ip2 & ip3[, codec])

Just where do we get the port to send it to from, if
we use SIP_PEER? And is there a SIP client who handles
unicast RTP without signalling?

The other use for unicast RTP are clients like the Barix Extreamer.

What do you think? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-23-08 08:52  macbrody       Note Added: 0081091                          
======================================================================