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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1946/">https://reviewboard.asterisk.org/r/1946/</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;">With the minor changes below, this looks good to me.</pre>
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<a href="https://reviewboard.asterisk.org/r/1946/diff/2/?file=28284#file28284line28622" 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="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only)</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> <span class="k">else</span> <span class="k">if</span> <span class="p">(</span><span class="o">!</span><span class="n">strcasecmp</span><span class="p">(</span><span class="n">v</span><span class="o">-></span><span class="n">name</span><span class="p">,</span> <span class="s">"directmedia_outgoing"</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">ast_set2_flag</span><span class="p">(</span><span class="o">&</span><span class="n">peer</span><span class="o">-></span><span class="n">flags</span><span class="p">[</span><span class="mi">2</span><span class="p">],</span> <span class="n">ast_true</span><span class="p">(</span><span class="n">v</span><span class="o">-></span><span class="n">value</span><span class="p">),</span> <span class="n">SIP_PAGE3_DIRECT_MEDIA_OUTGOING</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;">This shouldn't be necessary now.</pre>
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<a href="https://reviewboard.asterisk.org/r/1946/diff/2/?file=28286#file28286line939" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/configs/sip.conf.sample</a>
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(Diff revision 2)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">;directmedia=outgoing ; Only send directmedia reinvites for outgoing calls. This is useful</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> ; to help prevent reinvite glare scenarios (i.e. two SIP servers sending</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                                                                simultaneous reinvites to one another).</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;">Add "It implies 'yes'." like the 'update' option has.</pre>
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<p>- Kevin</p>
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<p>On May 25th, 2012, 2:12 p.m., Mark Michelson wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Mark Michelson.</div>
<p style="color: grey;"><i>Updated May 25, 2012, 2:12 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;">There are times where multiple Asterisk servers are peered together over SIP. In such situations, it is possible for both Asterisk servers to attempt to send direct media reinvites to each other simultaneously. This results in a glare situation in which each of the Asterisk servers sends a 491 to the other. After a waiting period, the reinvites are re-attempted. This waiting period can potentially be distracting since it can cause the media to take multiple seconds to finalize, especially if more than 2 Asterisk servers are involved.
This patch introduces a new SIP peer option called "directmedia_outgoing". If enabled, then when communicating with the peer, Asterisk will only attempt to send reinvites if the call direction is outgoing. The assumption is that the peer Asterisk server will also have this setting enabled. This way, when the two Asterisk servers communicate, they will never attempt to send direct media reinvites to each other. Instead, it will always be the peer that placed the call that will send the direct media reinvite.</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;">I have tested this by running two Asterisk servers and ensuring that the option was honored and that the media streams were still set up properly.</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_sip.c <span style="color: grey">(367639)</span></li>
<li>/trunk/channels/sip/include/sip.h <span style="color: grey">(367639)</span></li>
<li>/trunk/configs/sip.conf.sample <span style="color: grey">(367639)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1946/diff/" style="margin-left: 3em;">View Diff</a></p>
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