[asterisk-dev] [Code Review] Call Completion: Asterisk Component

Mark Michelson mmichelson at digium.com
Mon Feb 22 18:06:53 CST 2010 


> On 2010-02-22 17:47:34, David Vossel wrote:
> > This is a result of a quick scan through the chan_sip changes.

Thanks for the comments, David. In the future, could you try to highlight more lines of code in your comments. The last one, in particular, was hard to tell where the comment was.


> On 2010-02-22 17:47:34, David Vossel wrote:
> > /trunk/channels/chan_sip.c, line 934
> > <https://reviewboard.asterisk.org/r/523/diff/2/?file=8236#file8236line934>
> >
> >     The ast_cc_service_type enum starts with AST_CC_NONE.  Would not having this in this map cause issues?  What would happen if sip_cc_service_map[0] was accessed for some reason later and this wasn't present?  The service string would not be initialized.

The way the code is set up currently, it is not possible to access sip_cc_service_map[0], but you make a good point that future code might try to do that. What you are incorrect about though is that the service string will be uninitialized. It will actually be empty, which really isn't a problem.


> On 2010-02-22 17:47:34, David Vossel wrote:
> > /trunk/channels/chan_sip.c, line 5085
> > <https://reviewboard.asterisk.org/r/523/diff/2/?file=8236#file8236line5085>
> >
> >     What if recall_monitor is NULL here?!  you check to verify it isn't in the block above this which indicates that it could be possible.

Definitely right on this one. The ao2_t_ref call should be inside the if block here.


> On 2010-02-22 17:47:34, David Vossel wrote:
> > /trunk/channels/chan_sip.c, line 21923
> > <https://reviewboard.asterisk.org/r/523/diff/2/?file=8236#file8236line21923>
> >
> >     Just a code style comment here: This function has a few things to keep up with on return. unref the agent, close pidf_doc, and free_text.  This looks like a good candidate to use a cleanup_failure label and a goto cleanup_failure instead of return -1.  Everything looks good here though.

Sounds like a good idea.


- Mark


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On 2010-02-22 15:16:37, Mark Michelson wrote:
> 
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> 
> (Updated 2010-02-22 15:16:37)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This is it folks. CCSS is at a point where a review will be beneficial.
> 
> CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
> overview of the architecture can be found in the file doc/CCSS_architecture.pdf
> in the CCSS branch. Off the top of my head, the big differences between what is
> implemented and what is in the document are as follows:
> 
> 1. We did not end up modifying the Hangup application at all.
> 2. The document states that a single call completion monitor may be used across
>    multiple calls to the same device. This proved to not be such a good idea
>    when implementing protocol-specific monitors, and so we ended up using one
>    monitor per-device per-call.
> 3. There are some configuration options which were conceived after the document
>    was written. These are documented in the ccss.conf.sample that is on this
>    review request.
>    
> For some basic understanding of terminology used throughout this code, see the
> ccss.tex document that is on this review.
> 
> This implements CCBS and CCNR in several flavors.
> 
> First up is a "generic" implementation, which can work over any channel technology
> provided that the channel technology can accurately report device state. Call
> completion is requested using the dialplan application CallCompletionRequest and can
> be canceled using CallCompletionCancel. Device state subscriptions are used in order
> to monitor the state of called parties.
> 
> Next, there is a SIP-specific implementation of call completion. This method uses the
> methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
> using SIP signaling. There are a few things to note here:
> 
> * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
>   what is defined in the referenced draft.
> 
> * Implementation of the draft required support for SIP PUBLISH. I attempted to write
>   this in a generic-enough fashion such that if someone were to want to write PUBLISH
>   support for other event packages, such as dialog-state or presence, most of the effort
>   would be in writing callbacks specific to the event package.
> 
> * A subportion of supporting PUBLISH reception was that we had to implement a PIDF
>   parser. The PIDF support added is a bit minimal. I first wrote a validation
>   routine to ensure that the PIDF document is formatted properly. The rest of the
>   PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
>   code. In other words, while there is PIDF support here, it is not in any state
>   where it could easily be applied to other event packages as is.
> 
> Finally, there are a variety of ISDN-related call completion protocols supported. These
> were written by Richard Mudgett, and as such I can't really say much about their
> implementation. I'm leaving it to Richard to add any comments he wants about this
> matter. The libpri component of call completion support is posted as review 522
> on ReviewBoard.
> 
> The code added here is somewhat massive, so questions are welcome as much as critiques.
> Happy reviewing!
> 
> 
> Diffs
> -----
> 
>   /trunk/CHANGES 248346 
>   /trunk/apps/app_dial.c 248346 
>   /trunk/channels/chan_dahdi.c 248346 
>   /trunk/channels/chan_local.c 248346 
>   /trunk/channels/chan_sip.c 248347 
>   /trunk/channels/sig_analog.h 248346 
>   /trunk/channels/sig_analog.c 248346 
>   /trunk/channels/sig_pri.h 248346 
>   /trunk/channels/sig_pri.c 248346 
>   /trunk/channels/sip/include/sip.h 248346 
>   /trunk/configs/ccss.conf.sample PRE-CREATION 
>   /trunk/configs/chan_dahdi.conf.sample 248346 
>   /trunk/configure.ac 248346 
>   /trunk/doc/tex/asterisk.tex 248346 
>   /trunk/doc/tex/ccss.tex PRE-CREATION 
>   /trunk/funcs/func_callcompletion.c PRE-CREATION 
>   /trunk/include/asterisk/ccss.h PRE-CREATION 
>   /trunk/include/asterisk/channel.h 248346 
>   /trunk/include/asterisk/channelstate.h PRE-CREATION 
>   /trunk/include/asterisk/devicestate.h 248346 
>   /trunk/include/asterisk/frame.h 248346 
>   /trunk/include/asterisk/manager.h 248346 
>   /trunk/include/asterisk/rtp_engine.h 248346 
>   /trunk/include/asterisk/xml.h 248346 
>   /trunk/main/asterisk.c 248346 
>   /trunk/main/ccss.c PRE-CREATION 
>   /trunk/main/channel.c 248346 
>   /trunk/main/xml.c 248346 
>   /trunk/tests/test_amihooks.c 248346 
>   /trunk/tests/test_utils.c 248346 
> 
> Diff: https://reviewboard.asterisk.org/r/523/diff
> 
> 
> Testing
> -------
> 
> Too much to list :)
> 
> For a decent look at the tests executed, see the "Testing Done" section of review request 410.
> Those tests have all been run using combinations of generic, SIP, and ISDN agents and monitors.
> 
> 
> Thanks,
> 
> Mark
> 
>

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