[asterisk-dev] [Code Review] Multicast RTP Paging Support

[Joshua Colp]

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http://reviewboard.digium.com/r/270/
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Review request for Asterisk Developers.


Summary
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This patch implements two new modules: res_rtp_multicast and chan_multicast_rtp. The resource module is an RTP engine which can be used by any developer to send multicast RTP. The channel driver uses the RTP engine and configures it based on input given from the user in the Dial line. Any audio sent to the RTP engine is broadcast out.


This addresses bug 11797.
    https://issues.asterisk.org/view.php?id=11797


Diffs
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  /trunk/channels/chan_multicast_rtp.c PRE-CREATION 
  /trunk/res/res_rtp_multicast.c PRE-CREATION 

Diff: http://reviewboard.digium.com/r/270/diff


Testing
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Confirmed that audio is sent out as expected by dialing using the channel driver in the dialplan.


Thanks,

Joshua