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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2318/">https://reviewboard.asterisk.org/r/2318/</a>
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<p>Ship it!</p>
<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">The only objection I have to this is that some of the SDP handler callbacks in res_sip_sdp_audio are a bit long. But given that I think cleanup and such things are on the horizon, I'm more concerned with functionality in this review. So, ship it!</pre>
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<p>- Mark</p>
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<p>On February 12th, 2013, 8:31 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 Feb. 12, 2013, 8:31 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;">These changes clean up media handling, move some more stuff into res_sip_sdp_audio, fixes a few bugs, and adds some additional features.
The act of negotiating an SDP media stream and actually applying the media stream are now separate operations.
Hold/unhold works.
RTP over IPv6 works.
Use of the 'ptime' attribute works.
Local Packet2Packet bridging works.
Symmetric RTP can now be enabled per-endpoint.
Reduced memory pool usage.
Fixed bug where the RTP instance was never destroyed.</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;">1. Sent and received calls from a few different devices
2. Held/unheld a call
3. Attempted to set up incompatible calls (only configured for gsm, but offering ulaw only)</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/team/group/pimp_my_sip/channels/chan_gulp.c <span style="color: grey">(381276)</span></li>
<li>/team/group/pimp_my_sip/configs/res_sip.conf.sample <span style="color: grey">(381276)</span></li>
<li>/team/group/pimp_my_sip/include/asterisk/res_sip.h <span style="color: grey">(381276)</span></li>
<li>/team/group/pimp_my_sip/include/asterisk/res_sip_session.h <span style="color: grey">(381276)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip/sip_configuration.c <span style="color: grey">(381276)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip_sdp_audio.c <span style="color: grey">(381276)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip_session.c <span style="color: grey">(381276)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2318/diff/" style="margin-left: 3em;">View Diff</a></p>
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