publishing a sip call to rtmp server

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I am looking for a way to do a live webcast of a sip call.
Is there a way I can use sip2rtmp to make a sip call and stream out the audio 
as a rtmp stream (or publish it to an sdp based encoder like wowza) so that 
other listeners can listen to the call?


Reply
Hi Madhusudana,

Sorry about the delay.
The existing implementations in rtmplite do not have this feature.
For this feature, there are several options within the scope of rtmplite:

1) create a client application that combines the features from
siprtmp.py and rtmpclient.py, so that it makes a SIP call as well as
makes an RTMP connection, and connects the audio/video path of the
two.

2) Modify siprtmp so that it can accept connection from your media
server (wowza) and auto-accept call from your SIP webcast server, or
auto-dial call to your webcast server.

This sounds like a good use case of the application. If you can create
a new enhancement request on the project "issues" page, it will be
great to keep track of the progress... 

Original issue reported on code.google.com by madhu...@bluewoodcapital.com on 29 Jan 2013 at 4:20