[asterisk-dev] [Code Review] 4473: chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.

[gareth]

> On March 12, 2015, 8:41 p.m., Matt Jordan wrote:
> > 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.
> 
> rmudgett wrote:
>     Actually I'd prefer that the rpid_immediate option not exist at all and the code it controls to just be removed.  Sending connected line updates back to the caller _before_ getting connected doesn't really make sense.  This is what the REDIRECTING information is supposed to be doing.

I disagree, providing connected line updates pre-answer makes perfect sense. Why would I want to know the connected-line name only after the call has been answered?

That assumes the call even is answered, sending the connected-line information immediately allows the phone to include the name in it's call history.

If no call-forwarding and/or re-addressing has taken place why would REDIRECTING be used?


- gareth


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On March 10, 2015, 11:48 p.m., rmudgett wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4473/
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> 
> (Updated March 10, 2015, 11:48 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24781
>     https://issues.asterisk.org/jira/browse/ASTERISK-24781
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch builds on https://reviewboard.asterisk.org/r/4472/
> 
> When that patch is committed this patch reduces to simply adding the
> rpid_immediate option to skip generating INVITE response messages as a
> result of connected line updates.
> 
> 
> 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_caller_id.c 432761 
>   /branches/13/res/res_pjsip/pjsip_configuration.c 432761 
>   /branches/13/res/res_pjsip.c 432761 
>   /branches/13/include/asterisk/res_pjsip.h 432761 
>   /branches/13/configs/samples/pjsip.conf.sample 432761 
>   /branches/13/channels/chan_pjsip.c 432761 
> 
> 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.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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