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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1377/">https://reviewboard.asterisk.org/r/1377/</a>
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<p style="margin-top: 0;">On August 23rd, 2011, 12:36 p.m., <b>rmudgett</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1377/diff/1/?file=18571#file18571line25672" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/chan_sip.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="kt">int</span> <span class="n">timeout</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="kt">int</span> <span class="n">holdtimeout</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="kt">int</span> <span class="n">keepalive</span> <span class="o">=</span> <span class="mi">0</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;">Initialization not necessary since they are initialized later.
Also you might wish to rename holdtimeout to hold_timeout to be consistent with the function name.</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;">Although initialization is not needed, I always do this as a best practice thing.
As for the name, "holdtimeout" is the name used for the value everywhere else in chan_sip, including the sip.conf variable name. "hold_timeout" only appears in the rtp instance code.</pre>
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<p>- Rob</p>
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<p>On August 23rd, 2011, 11:15 a.m., Rob Gagnon wrote:</p>
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<div>Review request for Asterisk Developers and Rob Gagnon.</div>
<div>By Rob Gagnon.</div>
<p style="color: grey;"><i>Updated Aug. 23, 2011, 11:15 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;">Copying from original post on Jira:
While reviewing how "rtptimeout" and "rtpholdtimeout" operate in code, I found that the check_rtp_timeout() function could be optimized a little to speed up asterisk performance.
Currently, three integers are fetched from the rtp instance multiple times apiece (causing of course, multiple stack operations)
In the worst-case scenario:
- ast_rtp_instance_get_timeout() is called 4 times.
- ast_rtp_instance_get_hold_timeout() is called 4 times.
- ast_rtp_instance_get_keepalive() is called 3 times.
With this patch, each function will be called only once, thus removing up to 9 stack push/pops during runtime
Description posted on jira would be the same
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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;">Compiled code with patch, ran on test system without any problems. Turned logging way up, and verifiied that calls that were silent/disconnected by accident were still being hung-up on.</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/ASTERISK-18319">ASTERISK-18319</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">(333000)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1377/diff/" style="margin-left: 3em;">View Diff</a></p>
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