[asterisk-dev] [Code Review]: chan_jingle2: New Jingle + Google Talk channel driver

[Joshua Colp]

> On June 14, 2012, 3:39 p.m., opticron wrote:
> > /trunk/channels/chan_jingle2.c, line 342
> > <https://reviewboard.asterisk.org/r/1917/diff/3/?file=28824#file28824line342>
> >
> >     Blob.

Fixed all whitespace. IN THE WORLD.


> On June 14, 2012, 3:39 p.m., opticron wrote:
> > /trunk/channels/chan_jingle2.c, line 343
> > <https://reviewboard.asterisk.org/r/1917/diff/3/?file=28824#file28824line343>
> >
> >     Should this use jingle_endpoint_find() since it is available?

Yeah, why not. Changed!


> On June 14, 2012, 3:39 p.m., opticron wrote:
> > /trunk/channels/chan_jingle2.c, line 823
> > <https://reviewboard.asterisk.org/r/1917/diff/3/?file=28824#file28824line823>
> >
> >     "break;" should be sufficient here.

Changed.


- Joshua


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On June 8, 2012, 3:32 p.m., Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1917/
> -----------------------------------------------------------
> 
> (Updated June 8, 2012, 3:32 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This is a new channel driver written from scratch for the Jingle, Google Jingle, and Google Talk protocols. It has been written to the specs available and tested extensively.
> 
> ICE and STUN support for Jingle uses the new ICE/STUN/TURN support which is present in another review. (Please do not review any of that code in this review)
> STUN support for Google uses the existing STUN implementation, as the new support is not compatible with it.
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_jingle2.c PRE-CREATION 
>   /trunk/channels/chan_sip.c 368682 
>   /trunk/configs/jingle2.conf.sample PRE-CREATION 
>   /trunk/configs/rtp.conf.sample 368682 
>   /trunk/include/asterisk/jabber.h 368682 
>   /trunk/include/asterisk/jingle.h 368682 
>   /trunk/include/asterisk/rtp_engine.h 368682 
>   /trunk/main/rtp_engine.c 368682 
>   /trunk/res/Makefile 368682 
>   /trunk/res/res_jabber.c 368682 
>   /trunk/res/res_rtp_asterisk.c 368682 
> 
> Diff: https://reviewboard.asterisk.org/r/1917/diff
> 
> 
> Testing
> -------
> 
> Tested audio calls with following:
> 
> GMail Google Talk Plug-in (and video)
> Google Voice
> Jitsi (and video)
> Psi
> OneTeam
> 
> * Included varying codecs (ulaw, speex, g722, etc)
> 
> Tested ringing, hold, and unhold with following:
> 
> Jitsi
> 
> Other clients do not support this.
> 
> 
> Thanks,
> 
> Joshua
> 
>

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