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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1924/">https://reviewboard.asterisk.org/r/1924/</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;">Excellent! This will work great. I have one nit-picky optimization-esque suggestion that you are free to ignore if you want.</pre>
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<a href="https://reviewboard.asterisk.org/r/1924/diff/3/?file=28172#file28172line30388" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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(Diff revision 3)
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<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
<td colspan="2"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_channel *chan2, char *rtptype)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">        <span class="n">sip_pvt_lock</span><span class="p">(</span><span class="n">p2</span><span class="p">);</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">        <span class="k">if</span> <span class="p">(</span><span class="n">ast_test_flag</span><span class="p">(</span><span class="o">&</span><span class="n">p2</span><span class="o">-></span><span class="n">flags</span><span class="p">[</span><span class="mi">0</span><span class="p">],</span> <span class="n">SIP_DIRECT_MEDIA</span><span class="p">))</span> <span class="p">{</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                <span class="k">if</span> <span class="p">(</span><span class="o">!</span><span class="n">apply_directmedia_ha</span><span class="p">(</span><span class="n">p2</span><span class="p">,</span> <span class="n">p1_directmediaha</span><span class="p">,</span> <span class="n">rtptype</span><span class="p">))</span> <span class="p">{</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                        <span class="n">res</span> <span class="o">=</span> <span class="mi">0</span><span class="p">;</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                <span class="p">}</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">        <span class="p">}</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">        <span class="n">sip_pvt_unlock</span><span class="p">(</span><span class="n">p2</span><span class="p">);</span></pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">You don't have to do this, but you can skip over these lines entirely if res is 0.</pre>
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<p>- Mark</p>
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<p>On May 21st, 2012, 9:29 a.m., jrose wrote:</p>
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<div>Review request for Asterisk Developers, Mark Michelson and Matt Jordan.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated May 21, 2012, 9:29 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;">Related to https://reviewboard.asterisk.org/r/1899/ which has some locking changes that need to be made still...
This patch takes Mark's suggested approach for adding callbacks for the rtp_bridge function to be able to supply two channels for determining rtp glue stuff in cases where a channel driver needs data about both peers in order to determine what types of bridging are permissible.</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;">Similar to the testing done for the review 1899 version. Calls were tested with and without directmediapermit/deny on both sides of a call with calls being started from both directions. Specific things tested include whether or not the host address lists were copied properly (because that was a rather substantial problem earlier on) and the results of ast_apply_ha in each case.</pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/AST-876">AST-876</a>
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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_sip.c <span style="color: grey">(366775)</span></li>
<li>/trunk/include/asterisk/rtp_engine.h <span style="color: grey">(366775)</span></li>
<li>/trunk/main/rtp_engine.c <span style="color: grey">(366775)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1924/diff/" style="margin-left: 3em;">View Diff</a></p>
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