[asterisk-dev] [Code Review] 4473: chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.
rmudgett
reviewboard at asterisk.org
Fri Mar 13 12:25:54 CDT 2015
> On March 12, 2015, 3:41 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.
>
> gareth wrote:
> 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?
OK, that makes sense but the code that rpid_immediate controls doesn't. As described in the review description, that code immediately sends redundant "180 Ringing" or "183 Progress" messages at best and at worst lies that the other end is ringing with inaccurate connected line information.
- rmudgett
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On March 13, 2015, 12:13 p.m., rmudgett wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4473/
> -----------------------------------------------------------
>
> (Updated March 13, 2015, 12:13 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 432895
> /branches/13/res/res_pjsip.c 432895
> /branches/13/include/asterisk/res_pjsip.h 432895
> /branches/13/configs/samples/pjsip.conf.sample 432895
> /branches/13/channels/chan_pjsip.c 432895
>
> 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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