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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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<a href="https://reviewboard.asterisk.org/r/1946/diff/2/?file=28284#file28284line30604" 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 int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active)</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">p</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">SIP_PAGE3_DIRECT_MEDIA_OUTGOING</span><span class="p">)</span> <span class="o">&&</span> <span class="o">!</span><span class="n">p</span><span class="o">-></span><span class="n">outgoing_call</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_log</span><span class="p">(</span><span class="n">LOG_NOTICE</span><span class="p">,</span> <span class="s">"Can't send reinvite because it is an incoming call!</span><span class="se">\n</span><span class="s">"</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">sip_pvt_unlock</span><span class="p">(</span><span class="n">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_channel_unlock</span><span class="p">(</span><span class="n">chan</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;">I decided to do another round of testing before committing and noticed that despite having directmedia=outgoing set, I would see reinvite glares occasionally. I then noticed this block of code. I had meant to be returning here instead of continuing with an unlocked sip_pvt and channel. Adding the return results in one-way audio in my tests, so I need to figure out why that is happening and how to fix it.</pre>
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<p>- Mark</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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