CVS log for pkgsrc/devel/opal/Attic/buildlink3.mk

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Default branch: MAIN

*: Remove Ekiga and related packages.

No OpenSSL 1.1 support and upstream development is effectively stalled.

Also various other dead dependencies: gstreamer0.10, sdl12, esound,
GNOME 2 libgnomeui...

Bump PKGREVISION for security/openssl ABI bump.

Recursive revbump following MesaLib update, categories a through f.

Recursive PKGREVISION bump for OpenSSL API version bump.

PKGREVISION bumps for the security/openssl 1.0.1d update.

recursive bump from cyrus-sasl libsasl2 shlib major bump.

Set BUILDLINK_ABI_DEPENDS correctly (with +=, not ?=)
It turns out there were a lot of these.

Recursive PKGREVISION bump for jpeg update to 8.

Needs to include speex here as well.

Simply and speed up buildlink3.mk files and processing.
This changes the buildlink3.mk files to use an include guard for the
recursive include. The use of BUILDLINK_DEPTH, BUILDLINK_DEPENDS,
BUILDLINK_PACKAGES and BUILDLINK_ORDER is handled by a single new
variable BUILDLINK_TREE. Each buildlink3.mk file adds a pair of
enter/exit marker, which can be used to reconstruct the tree and
to determine first level includes. Avoiding := for large variables
(BUILDLINK_ORDER) speeds up parse time as += has linear complexity.
The include guard reduces system time by avoiding reading files over and
over again. For complex packages this reduces both %user and %sys time to
half of the former time.

Update opal to 3.6.1. Again no changelog in sight.

Main features on this new major release branch are:

    * Instant Messaging. Using several different ?standards?: RFC4975, MSRP, T.140, SIP-IM etc etc.
    * Much improved support for SIP event handling (SUBSCRIBE/NOTIFY/PUBLISH) using factories so more event packages can be added easily.
    * Support for SIP ?dialog? event package.
    * Support for Shared Line Appearance (aka Multiple Line Appearance, Bridged Line Appearance or Busy Lamp Field)
    * Significant improvements in IPv6 support (thanks Yuri Kiryanov!)
    * Major enhancements to the video rate controller.
    * Removal of the old H.263 ffmpeg plug in and enhancement of the H.263-1998 plug it to take its place.
    * H.224 support (H.323 only).
    * SBC audio codec plug in.
    * G.722 audio codec plug in.
    * CELT audio codec plug in.
    * SWIG support for use in other languages, Java first.
    * Much improved multi-platform sample GUI client, OpenPhone. Now standard test environment replacing simpleOPAL.

Update to 3.4.2:

* Update version number for beta v3.4.2
* Fix system speex detection.
* Simplify setting of LIBS and CFLAGS.
* Fix H.261 on solaris.
  Thanks to Elaine Xiong from SUN.
* Fixed line endings
* Fixed inclusion of RFC2833 (DTMF) capability in H.323 calls,
  thanks Hannes Friederich
* Fixed bug in "C" API when message callback function is used
  instead of the normal message queue. Thanks Yuriy Cherniavsky
* Fixed SIP INVITE loop detection, forked INVITE request arriving
  over multiple paths.
* Applied patch:
  [ 2119085 ] "userinput" disables "fax" for H.323
  Thanks Vyacheslav Frolov
* Note some parts of H.323 still use the numerical form as it is
  fundamental to the protocol so H323Capability has acquired the
  DefaultAudioSessionID etc enumeration.
* Added IVR sample application.
* Fixed IVR media format list being altered by
  OpalCall::OpenSourceMediaStreams() due to container class
  referencing single instance.
* Fixed release of the call when audio channel ends. IVR cannot
  continue when no more audio being sent.
* Fixed local party URL so does not use a username (meaningless),
  but uses the VXML reference as the part after the "ivr:".
* Added parameter to ivropal set the SIP or H.323 listening
  interfaces
* Fix static x264 compilation.
* Fix H.264 helper filename on windows.
* Mechanism for forking (over multiple interfaces) the register was
  incorrect, The specification is typically vague on the subject as
  it only goes in to detail on forking with respect to INVITE.
  However the new technique is believed to be compliant by me at
  least!
* When registering, should include all possible interfaces.
  Especially UDP and TCP should both be registered.
* When registering should also do the appropriate STUN handling,
  depending on if registrar is "inside" or "outside".
* Parsing of a URI that is quoted by <> but has no "" based display
  name was incorrect and would the URI as display name.
* Parsing of sips URI when quoted by <> would fail completely.
* Creation of a sips URI from username and transport address was
  broken as well. When fixing this also prevented default transport
  from being included in URI.
* Added mutex when writing simultaneous transactions to same
  transport, could sometimes cause packet to go out incorrect
  interface.
* Fixed incorrect LID caller line ID field being used for display
  name.
* Fixed crash if H.323 OLC opens codec that fails to be createad
  late in the process.
* Adapt to libavcodec's new interface 52.
* Also trigger the presence callback when the NOTIFY body is empty.
  It is desirable for the client to know something happened.
* Update wxWidgets version in OpenPhone ReadMe.txt instructions.
* Fixed interoperability issues with Cisco call manager, working
  around an annoying bug with the Cisco.
* Killed the ununsed .major and .major.minor symlinks.
  The SONAME has always referred to the .major.minor.build lib
* Fixed OpenPhone about box.
* Fix H.264 flags to request I-frames thanks to Mauro Davi.
* Improved LID logging so indicates reason why line was not
  selected.
* Added some H.261 and H.263 annexes to H.263
* Added command line argument to codec tester so can adjust any
  media option on the encoder.
* Added OpenPhone menu and dialog to do transmit video control, in
  particular, changing the target bit rate.
* Reversed stupid change
* Fixed issue with propagating changes to media options to
  transcoders and streams in a patch.
* Added video frame rate to OpenPhone statistics.
* Fixed phantom "busy" detect in LID.
* Fixed incorrect line type (pots/pstn) selection.
* Fixed using any line if cannot find specificity requested line,
  should return "busy" if the line is not available.
* Changed ring detect so increments count on "rising edge" of ring
  cadence. Reduces delay after caller ID is received and routing
  call.
* Added function to get at OpalLine from OpalLineConnection.
* Fixed DevStudio compile/link issues with some options removed, in
  particular IAX2 and H.323. Have gone through every header file
  and made sure that they are all subject to the #defines in
  opal/buildops.h. Reported by Giorgio Alfarano.
* Also made sure they all have consistent #defines for multiple
  inclusion, reported by Stefano Sabatini.
* Fixed numerous retries using the same authentication credentials.
  If the first time doesn't work due to bad username or password
  and nothing changes then is highly unlikely to work one of the
  other 9 times it retries!
* Added priority field when have multiple interfaces being
  registered. Always prioritise the interface the register as being
  talked to on.
* Fixed listening interfaces port when mapped by STUN being
  included as first interface in the listener list.
* Fixed problem in forked INVITE where remote can be seen my
  multiple interfaces, a race exists between CANCEL of second
  dialog and it being accepted by the remote server with a 200 OK.
* Fixed numerous retries using the same authentication credentials.
  If the first time doesn't work due to bad username or password
  and nothing changes then is highly unlikely to work one of the
  other 9 times it retries!
* Change error for a SIP BYE on a call already destroyed from "405
  Method not allowed" to more reasonable "481 Transaction/call does
  not exist"
* Windows Mobile build fixes
* Added user defined tone in LIDs.
* Backport from TRUNK (Deal with endpoints that use spaces and ; as
  seperators in the same FMTP string. Sheesh!).
* Fixed some Linux weirdness where the first attempt to do TCP
  listen exclusive prevents the second attempt to listen with
  REUSEADDR flag set. The reason the code tries one and then the
  other is lost in the mists of time, so lets just make it do one
  OR the other and see what gives.
* Also, fixed being able to specify socket re-use address mode (a
  trailing '+' in address string) without a port. Now something
  like "*+" will work, listening on all interfaces in non-exclusive
  mode.
* Fix second case where simpleopal tries to use SDL for both output
  and preview video
  Backport from trunk
* Fixed race condition where if reply from SIP server is so fast
  that it gets in before the second PDU of a forked query, the
  second PDU is sent via the incorrect interface.
* Fixed printing of new LID tone.
* Fixed missing party B address when originating PSTN call.

Import opal-3.4.1 stable release to replace devel/opal-devel

OPAL is the "next generation" of OpenH323 that has a new architecture.
Not only does it support H.323 and SIP, but new VoIP protocols or
devices can be added very easily.
BTW: OPAL is an acronym for Open Phone Abstraction Library, but
that is not really important :)

Initial revision