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

[Joshua Colp]

> On May 16, 2012, 3:09 p.m., Paul Belanger wrote:
> > What sort of upgrade path are we expecting between chan_jingle and chan_jingle2? I didn't see anything specific on the wiki pages. If I remember right, this is not a drop in replacement so it might be good to have it documented on the wiki what is and what is not.
> > 
> > Additionally, I mentioned this in passing, while it is cool we have version 2 of jingle, I'm not a fan of appending 2 to the channel name.  Perhaps something like chan_xmpp_jingle? At first it is kinda ugly, but seems to grow on me as I look at it more.
> 
> Leif Madsen wrote:
>     I also dislike adding '2' to the end. I'm not really picky, and what Paul has suggested seems reasonable.

I really really really dislike chan_xmpp_jingle. It's long and no other channel driver (besides multicast RTP) uses two words to describe it like that. An alternative word to jingle... hrm.


- Joshua


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On June 8, 2012, 3:32 p.m., Joshua Colp wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1917/
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> 
> (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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