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

[Mark Michelson]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/523/
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(Updated 2010-02-22 15:16:37.571082)


Review request for Asterisk Developers.


Changes
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Addressed my comments.

For some reason, the test_utils and test_amihooks files really like to delete themselves when I merge in the CCSS changes. I'll have to investigate that a bit more. I assure you that I will not actually make those changes when the time comes to commit :)


Summary
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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 An admittedly out-of-date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 (updated)
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  /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
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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