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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4473/">https://reviewboard.asterisk.org/r/4473/</a>
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<a href="https://reviewboard.asterisk.org/r/4473/diff/4/?file=72697#file72697line418" style="color: black; font-weight: bold; text-decoration: underline;">/branches/13/include/asterisk/res_pjsip.h</a>
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(Diff revision 4)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">struct ast_sip_endpoint_id_configuration {</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="cm">/*! Do we send messages for connected line updates for unanswered incoming calls immediately to this endpoint? */</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="kt">unsigned</span> <span class="kt">int</span> <span class="n">rpid_immediate</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;">Is this an ABI issue? Maybe this member should be last in the structure for v13 to avoid changing the offset of send_diversion and refresh_method. Seems like it's in the right place for trunk (keeping unsigned int's together). Not sure if it even matters for res_pjsip to provide stable ABI.</pre>
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<p>- Corey Farrell</p>
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<p>On March 20th, 2015, 4:50 p.m. EDT, rmudgett wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By rmudgett.</div>
<p style="color: grey;"><i>Updated March 20, 2015, 4:50 p.m.</i></p>
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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-24781">ASTERISK-24781</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<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;">Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
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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;">* Ran the tests/channels/pjsip testsuite tests. They still pass.
* Made a call chain as follows: 100 -> * -> * -> * -> 200. With the patch
there are no unnecessary messages. Without the patch there were several
"180 Ringing" messages sent back to the caller.
* https://reviewboard.asterisk.org/r/4518/ testsuite test passes.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/13/res/res_pjsip/pjsip_configuration.c <span style="color: grey">(433198)</span></li>
<li>/branches/13/res/res_pjsip.c <span style="color: grey">(433198)</span></li>
<li>/branches/13/include/asterisk/res_pjsip.h <span style="color: grey">(433198)</span></li>
<li>/branches/13/contrib/ast-db-manage/config/versions/23530d604b96_add_rpid_immediate.py <span style="color: grey">(PRE-CREATION)</span></li>
<li>/branches/13/configs/samples/pjsip.conf.sample <span style="color: grey">(433198)</span></li>
<li>/branches/13/channels/chan_pjsip.c <span style="color: grey">(433198)</span></li>
<li>/branches/13/CHANGES <span style="color: grey">(433198)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4473/diff/" style="margin-left: 3em;">View Diff</a></p>
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