[asterisk-dev] [Code Review] 4473: chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.
rmudgett
reviewboard at asterisk.org
Mon Mar 23 15:08:53 CDT 2015
> On March 20, 2015, 11:50 p.m., Corey Farrell wrote:
> > /branches/13/include/asterisk/res_pjsip.h, lines 418-419
> > <https://reviewboard.asterisk.org/r/4473/diff/4/?file=72697#file72697line418>
> >
> > 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.
Ugh. Yes it is an ABI change. Even worse, the new value cannot go at the end of the struct because the struct itself is contained within a key struct (ast_sip_endpoint). This is going to put a damper on new options in res_pjsip for v13.
- rmudgett
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On March 20, 2015, 3:50 p.m., rmudgett wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4473/
> -----------------------------------------------------------
>
> (Updated March 20, 2015, 3:50 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-24781
> https://issues.asterisk.org/jira/browse/ASTERISK-24781
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> 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.
>
>
> Diffs
> -----
>
> /branches/13/res/res_pjsip/pjsip_configuration.c 433198
> /branches/13/res/res_pjsip.c 433198
> /branches/13/include/asterisk/res_pjsip.h 433198
> /branches/13/contrib/ast-db-manage/config/versions/23530d604b96_add_rpid_immediate.py PRE-CREATION
> /branches/13/configs/samples/pjsip.conf.sample 433198
> /branches/13/channels/chan_pjsip.c 433198
> /branches/13/CHANGES 433198
>
> Diff: https://reviewboard.asterisk.org/r/4473/diff/
>
>
> Testing
> -------
>
> * 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.
>
>
> Thanks,
>
> rmudgett
>
>
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