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

[Russell Bryant]

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/trunk/channels/chan_multicast_rtp.c
<http://reviewboard.digium.com/r/270/#comment2009>

    Don't forget to set AF_INET.



/trunk/channels/chan_multicast_rtp.c
<http://reviewboard.digium.com/r/270/#comment2010>

    Probably LOAD_DECLINE ..



/trunk/res/res_rtp_multicast.c
<http://reviewboard.digium.com/r/270/#comment2011>

    You can just use structure assignment here.



/trunk/res/res_rtp_multicast.c
<http://reviewboard.digium.com/r/270/#comment2012>

    Space after casts.
    
    Also, if this fails, it would be good to log the strerror(errno).



/trunk/res/res_rtp_multicast.c
<http://reviewboard.digium.com/r/270/#comment2013>

    Isn't this equivalent to the STANDARD macro?


- Russell


On 2009-06-01 08:07:07, Joshua Colp wrote:
> 
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> 
> (Updated 2009-06-01 08:07:07)
> 
> 
> 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
> -----
> 
>   /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
> 
>