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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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<p style="margin-top: 0;">On June 26th, 2012, 5:24 p.m., <b>Matt Jordan</b> wrote:</p>
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<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's comment on res_xmpp, this could use another pair of eyes before it gets committed, but this looks good to me.</pre>
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<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>
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<p>- Matt</p>
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<p>On June 21st, 2012, 8:32 a.m., Joshua Colp wrote:</p>
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<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>
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<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>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<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>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<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>
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<p><a href="https://reviewboard.asterisk.org/r/1917/diff/" style="margin-left: 3em;">View Diff</a></p>
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