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

Joshua Colp jcolp at digium.com
Thu Jun 25 12:31:11 CDT 2009



> On 2009-06-02 12:54:14, Russell Bryant wrote:
> > /trunk/channels/chan_multicast_rtp.c, line 114
> > <http://reviewboard.digium.com/r/270/diff/1/?file=5485#file5485line114>
> >
> >     Don't forget to set AF_INET.

Done.


> On 2009-06-02 12:54:14, Russell Bryant wrote:
> > /trunk/channels/chan_multicast_rtp.c, line 170
> > <http://reviewboard.digium.com/r/270/diff/1/?file=5485#file5485line170>
> >
> >     Probably LOAD_DECLINE ..

Done.


> On 2009-06-02 12:54:14, Russell Bryant wrote:
> > /trunk/res/res_rtp_multicast.c, lines 256-259
> > <http://reviewboard.digium.com/r/270/diff/1/?file=5486#file5486line256>
> >
> >     Isn't this equivalent to the STANDARD macro?

Done.


> On 2009-06-02 12:54:14, Russell Bryant wrote:
> > /trunk/res/res_rtp_multicast.c, line 224
> > <http://reviewboard.digium.com/r/270/diff/1/?file=5486#file5486line224>
> >
> >     Space after casts.
> >     
> >     Also, if this fails, it would be good to log the strerror(errno).

Done.


> On 2009-06-02 12:54:14, Russell Bryant wrote:
> > /trunk/res/res_rtp_multicast.c, line 154
> > <http://reviewboard.digium.com/r/270/diff/1/?file=5486#file5486line154>
> >
> >     You can just use structure assignment here.

Done.


- Joshua


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On 2009-06-01 08:07:07, Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> http://reviewboard.digium.com/r/270/
> -----------------------------------------------------------
> 
> (Updated 2009-06-01 08:07:07)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> 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
> -------
> 
> Confirmed that audio is sent out as expected by dialing using the channel driver in the dialplan.
> 
> 
> Thanks,
> 
> Joshua
> 
>




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