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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2305/">https://reviewboard.asterisk.org/r/2305/</a>
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<p style="margin-top: 0;">On February 11th, 2013, 8:11 a.m., <b>Joshua Colp</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/2305/diff/2/?file=33297#file33297line720" style="color: black; font-weight: bold; text-decoration: underline;">/team/group/pimp_my_sip/res/res_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; ">static int unload_module(void)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="cm">/* </span><span class="cs">XXX</span><span class="cm"> There is an issue right now that on Asterisk shutdown, res_sip</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * is unloaded before other SIP modules. When those other SIP modules attempt</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * to unregister their services, they rely on the threadpool to be present.</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * So for now, don't shut down the threadpool. Once module dependency stuff is</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * worked out, then this can be uncommented so the threadpool shuts down.</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> */</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="cm">/* ast_threadpool_shutdown(sip_threadpool); */</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;">I improved module reference count stuff so this may actually work now.</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;">Yep, seems to be fine now.</pre>
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<p>- Mark</p>
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<p>On February 6th, 2013, 5:13 p.m., Mark Michelson wrote:</p>
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<div>Review request for Asterisk Developers, Matt Jordan and Joshua Colp.</div>
<div>By Mark Michelson.</div>
<p style="color: grey;"><i>Updated Feb. 6, 2013, 5:13 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;">This changeset introduces threadpool usage into the new SIP work. Function calls that originate from the Asterisk core (such as channel callbacks) and from PJSIP's event handling thread (such as incoming request/response callbacks) now push their work into the SIP threadpool.
The benefits to this changeset are:
1) Frees up PJSIP event handling threads so that they can process more incoming SIP messages at a time
2) Places similar SIP tasks into their own task queue so that they can be completed in sequence, thus limiting resource contention. So far, session tasks are the only ones that use this mechanism
I added a new API call called ast_sip_push_task_synchronous(), that allows for you to push a task to a servant and block until the task completes. This is used in several cases where Asterisk threads need to make use of a PJSIP feature but need to have the result of the pushed task before returning. A good example of this is gulp_request() in chan_gulp. We need to call some PJSIP routines to set up a UAC and a dialog and such. But we also need to return a new ast_channel before gulp_request() returns.
Please have a look over this to see if I've made any obvious mistakes (e.g. memory leaks, tasks being not being executed in the threadpool when they should be)</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've run the OPTIONS test in the testsuite and can confirm that Asterisk shutdown does not crash Asterisk any more. I also ran incoming and outgoing calls and ensured that they completed correctly with no issue.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/team/group/pimp_my_sip/channels/chan_gulp.c <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/include/asterisk/res_sip.h <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/include/asterisk/res_sip_session.h <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/include/asterisk/threadpool.h <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/main/taskprocessor.c <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/main/threadpool.c <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip.c <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip.exports.in <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip/config_transport.c <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip/sip_options.c <span style="color: grey">(381009)</span></li>
<li>/team/group/pimp_my_sip/res/res_sip_session.c <span style="color: grey">(381009)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2305/diff/" style="margin-left: 3em;">View Diff</a></p>
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