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List:       asterisk-dev
Subject:    Re: [asterisk-dev] [Code Review]: chan_jingle2: New Jingle + Google Talk channel driver
From:       "Matt Jordan" <reviewboard () asterisk ! org>
Date:       2012-06-28 12:34:35
Message-ID: 20120628123435.27460.59061 () hotblack ! digium ! com
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> On June 26, 2012, 5:24 p.m., Matt Jordan wrote:
> > As with Mark's comment on res_xmpp, this could use another pair of eyes=
 before it gets committed, but this looks good to me.

We did realize this needs the new Call ID Logging functionality - but that =
could be done in a separate patch.


- Matt


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1917/#review6582
-----------------------------------------------------------


On June 21, 2012, 8:32 a.m., Joshua Colp wrote:
> =

> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1917/
> -----------------------------------------------------------
> =

> (Updated June 21, 2012, 8:32 a.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 availab=
le 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 thi=
s 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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      This is an automatically generated e-mail. To reply, visit:
      <a href="https://reviewboard.asterisk.org/r/1917/">https://reviewboard.asterisk.org/r/1917/</a>
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<blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; padding-left: \
10px;">  <p style="margin-top: 0;">On June 26th, 2012, 5:24 p.m., <b>Matt Jordan</b> \
wrote:</p>  <blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; \
padding-left: 10px;">  <pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; \
white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">As with \
Mark&#39;s comment on res_xmpp, this could use another pair of eyes before it gets \
committed, but this looks good to me.</pre>  </blockquote>







</blockquote>

<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: \
-pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">We did realize this \
needs the new Call ID Logging functionality - but that could be done in a separate \
patch.</pre> <br />








<p>- Matt</p>


<br />
<p>On June 21st, 2012, 8:32 a.m., Joshua Colp wrote:</p>






<table bgcolor="#fefadf" width="100%" cellspacing="0" cellpadding="8" \
style="background-image: \
url('https://reviewboard.asterisk.org/media/rb/images/review_request_box_top_bg.png'); \
background-position: left top; background-repeat: repeat-x; border: 1px black \
solid;">  <tr>
  <td>

<div>Review request for Asterisk Developers.</div>
<div>By Joshua Colp.</div>


<p style="color: grey;"><i>Updated June 21, 2012, 8:32 a.m.</i></p>




<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: \
1px solid #b8b5a0">  <tr>
  <td>
   <pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: \
-moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: \
break-word;">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.</pre>  </td>
 </tr>
</table>


<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: \
1px solid #b8b5a0">  <tr>
  <td>
   <pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: \
-moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: \
break-word;">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.</pre>
  </td>
 </tr>
</table>




<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">

 <li>/trunk/channels/chan_jingle2.c <span style="color: \
grey">(PRE-CREATION)</span></li>

 <li>/trunk/channels/chan_sip.c <span style="color: grey">(368682)</span></li>

 <li>/trunk/configs/jingle2.conf.sample <span style="color: \
grey">(PRE-CREATION)</span></li>

 <li>/trunk/configs/rtp.conf.sample <span style="color: grey">(368682)</span></li>

 <li>/trunk/include/asterisk/jabber.h <span style="color: grey">(368682)</span></li>

 <li>/trunk/include/asterisk/jingle.h <span style="color: grey">(368682)</span></li>

 <li>/trunk/include/asterisk/rtp_engine.h <span style="color: \
grey">(368682)</span></li>

 <li>/trunk/main/rtp_engine.c <span style="color: grey">(368682)</span></li>

 <li>/trunk/res/Makefile <span style="color: grey">(368682)</span></li>

 <li>/trunk/res/res_jabber.c <span style="color: grey">(368682)</span></li>

 <li>/trunk/res/res_rtp_asterisk.c <span style="color: grey">(368682)</span></li>

</ul>

<p><a href="https://reviewboard.asterisk.org/r/1917/diff/" style="margin-left: \
3em;">View Diff</a></p>




  </td>
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</table>








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