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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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<p style="margin-top: 0;">On March 13th, 2015, 5:19 p.m. CDT, <b>gareth</b> wrote:</p>
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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;">Ship It!</pre>
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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;">Well, if we're okay with the patch, then my initial comments still stand :-)
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1. For this to go into Asterisk 13, tests will need to be provided that cover the new parameter. (Really, those tests should be written regardless)
2. The CHANGES file will need to get updated with the new option.
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You'll need to update the alembic scripts for the new configuration option.
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<p>- Matt</p>
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<p>On March 13th, 2015, 12:13 p.m. CDT, 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 13, 2015, 12:13 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.
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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">(432895)</span></li>
<li>/branches/13/res/res_pjsip.c <span style="color: grey">(432895)</span></li>
<li>/branches/13/include/asterisk/res_pjsip.h <span style="color: grey">(432895)</span></li>
<li>/branches/13/configs/samples/pjsip.conf.sample <span style="color: grey">(432895)</span></li>
<li>/branches/13/channels/chan_pjsip.c <span style="color: grey">(432895)</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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