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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 style="margin-top: 0;">On May 18th, 2012, 2:52 p.m., <b>Mark Michelson</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1924/diff/2/?file=28132#file28132line30387" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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(Diff revision 2)
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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 void sip_get_anyrtp_peer_multi(struct ast_channel *chan1, struct ast_channel *chan2, enum ast_rtp_glue_result *glue1, enum ast_rtp_glue_result *glue2, 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="o">*</span><span class="n">glue1</span> <span class="o">=</span> <span class="n">AST_RTP_GLUE_RESULT_REMOTE</span><span class="p">;</span></pre></td>
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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 the function is currently written, you may end up overriding a previous AST_RTP_GLUE_RESULT_LOCAL with AST_RTP_GLUE_RESULT_REMOTE here. My suggestion at the top of the review will help with this.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">That's a good catch. Turns out this feature also prevented all use local bridging with permit/deny used as well.</pre>
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<p>- jrose</p>
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<p>On May 18th, 2012, 12:21 p.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 18, 2012, 12:21 p.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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