net/pjsip: Update to 2.12.1 and drop maintainership

net/pjsip: Update to 2.12

multimedia/libvpx: update 1.11.0

Changes:	https://chromium.googlesource.com/webm/libvpx/+log/v1.10.0..v1.11.0
Reported by:	portscout, Repology

net/pjsip: Add CPE information

Approved by:	portmgr (blanket)

net/pjsip: Update to 2.11.1

Security:	92ad12b8-ec09-11eb-aef1-0897988a1c07

Clean up some things

- Remove duplicate variables
- Remove nop variables
- Sort categories
- Remove redundant option descriptions that match the default ones

Reported by:	portscan

One more small cleanup, forgotten yesterday.
Reported by:	lwhsu

all: Remove all other $FreeBSD keywords.

Remove # $FreeBSD$ from Makefiles.

Update pjsip to 2.11

multimedia/openh264: update to 2.1.1

Changes:	https://github.com/cisco/openh264/releases/tag/v2.1.1
ABI:		https://abi-laboratory.pro/tracker/timeline/openh264/
Reported by:	GitHub (watch releases)

Add patches fixing race condition in build system. This allows
removing MAKE_JOBS_UNSAFE.

PR:		246300
Submitted by:	Dmitry Wagin <dmitry.wagin@ya.ru>

Import patch to fix build on !x86 !arm.

PR:		246014
Submitted by:	pkubaj

- Update pjsip t 2.10
- Get distfile from github
- Add OPUS and VPX options, enabled by default
- Make VIDEO and WEBRTC options enabled by default
- Fix typo in WEBRTC option description
- Fix pkgconfig patch to respect LOCALBASE

PR:		245607
Submitted by:	yuri

net/pjsip: Unbreak IPV6 option

- Do not set IPV6_CFLAGS twice
- Turn off IPv6 support when the option is off

Differential Revision:	https://reviews.freebsd.org/D22693

Add USES={gl,gnome,mate,sdl,xorg} missed in the last round

Update patches imported from asterisk project to latest version.

Obtained
from:	https://gerrit.asterisk.org/plugins/gitiles/asterisk/+/164e5372a213bd3264870d614f006df2be043938

Import patches from asterisk project.

MFH:		2019Q4

multimedia/ffmpeg: update to 4.2

Changes:	https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n4.2:/Changelog
ABI:		https://abi-laboratory.pro/tracker/timeline/ffmpeg/

- Update pjsip to 2.9
- Remove patches already applied upstream
- Bump PORTREVISION on asterisk ports to avoid crashes

multimedia/openh264: update to 2.0.0

- Add symlinks for cliqz and thunderbird

Changes:	https://github.com/cisco/openh264/releases/tag/v2.0.0
ABI:		https://abi-laboratory.pro/tracker/timeline/openh264/
Reported by:	GitHub (watch releases)

- Add to pjsip a customized config_site.h file with values suggested
  by the asterisk project. This allows WebRTC to work correctly in
  asterisk out of the box [1]
- Also import some patches to pjsip from the asterisk project. These
  patches have already been integrated in upstream pjsip development
  sources [2]

Reported by:	zhenya1993lzn@yandex.ru [1]
Obtained
from:	https://github.com/asterisk/asterisk/tree/16.3/third-party/pjproject/patches
[2]
MFH:		2019Q2

- Add LICENSE_FILE
- Update WWW

Approved by:	portmgr blanket

- Update asterisk13 to 13.24.0
- Update asterisk15 to 15.7.0
- Update asterisk16 to 16.1.0
- Update pjsip to 2.8
- Remove upstreamed patches from pjsip port
- Add USES=gnome to asterisk ports
- Silence portlint warnings

multimedia/ffmpeg: update to 4.1

Changes:	https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n4.1:/Changelog
ABI:		https://abi-laboratory.pro/tracker/timeline/ffmpeg/

Bump PORTREVISION to chase update of multimedia/v4l_compat and libv4l

Make asterisk and pjsip ports use the new net/libsrtp2 port as a dependency.

Reported by:	tijl (thanks!)

multimedia/ffmpeg: update to 4.0

- FFSERVER support was removed upstream
- libressl now uses libtls backend instead of patching openssl one
- Clang i386 no longer uses 16-byte aligned stack

Changes:	https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n4.0:/Changelog
ABI:		https://abi-laboratory.pro/tracker/timeline/ffmpeg/
PR:		227726
Exp-run by:	antoine
Differential Revision:	https://reviews.freebsd.org/D15175

Reword option description to include all asterisk versions.

- Update net/pjsip to 2.7.2
- Sort options and option helpers
- Remove stray YUV_DESC variable

- Update pjsip to 2.7.1
- Import patches from the asterisk project
- Regenrate patches

Security:	19b052c9-c533-11e7-8da5-001999f8d30b
MFH:		2017Q4

multimedia/ffmpeg: update to 3.4

Notable changes:
- i386/amd64 now depend on NASM per
https://git.ffmpeg.org/gitweb/ffmpeg.git/commitdiff/4f9297ac3b39
- NETCDF is now MYSOFA but the dependency doesn't exist in ports yet
- SCHROEDINGER is gone per
https://git.ffmpeg.org/gitweb/ffmpeg.git/commitdiff/220b24c7c9
- DRM, LIBRSVG2, LIBXML2 are new options
- ABI isn't completely compatible: some structs have changed

Minor cleanup:
- Don't pass --disable-{in,out}dev when it's already blocked by disabled
dependency
- Drop redundant "Enable" from option descriptions
- Switch CDIO_DESC to use Mk/bsd.options.desc.mk

Changes:	https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n3.4:/Changelog
ABI:		https://abi-laboratory.pro/tracker/timeline/ffmpeg/
PR:		223057
Exp-run by:	antoine

Update pjsip patch to work with libressl after the latest update
to 2.7, which broke it.

PR:		216898
Submitted by:	OlivierW <olivierw1+bugzilla-freebsd@hotmail.com>

Correctly handle option caused conflict.

PR:		222845
Submitted by:	O. Hartmann <ohartmann@walstatt.org>

Fix CONFLICTS to not overwrite slave port setting.

PR:		222804
Submitted by:	vvd@unislabs.com

- Update net/pjsip to 2.7
- Bump PORTVERSION on asterisk13 port to avoid ABI problems
- While here, sort the plist

Register conflict with libsrtp.

Reported by:	rodrigo
> PR:                       If and which Problem Report is related.
> Submitted by:             If someone else sent in the change.
> Reported by:              If someone else reported the issue.
> Reviewed by:              If someone else reviewed your modification.
> Approved by:              If you needed approval for this commit.
> Obtained from:            If the change is from a third party.
> MFC after:                N [day[s]|week[s]|month[s]].  Request a reminder
email.
> MFH:                      Ports tree branch name.  Request approval for merge.
> Relnotes:                 Set to 'yes' for mention in release notes.
> Security:                 Vulnerability reference (one per line) or
description.
> Sponsored by:             If the change was sponsored by an organization.
> Differential Revision:    https://reviews.freebsd.org/D### (*full* phabric URL
needed).
> Empty fields above will be automatically removed.

M    pjsip/Makefile

multimedia/openh264: update to 1.7.0

Changes:	https://github.com/cisco/openh264/releases/tag/v1.6.0
Changes:	https://github.com/cisco/openh264/releases/tag/v1.7.0
ABI:		https://abi-laboratory.pro/tracker/timeline/openh264/

- Update net/asterisk13 to 13.15.1
- Import patches to fix security issues in net/pjsip [1]
- While here regenerate net/pjsip patches

Obtained from:	Asterisk 13.15.1 distribution file [1]
MFH:		2017Q2
Security:	0537afa3-3ce0-11e7-bf9d-001999f8d30b
Security:	fab87bff-3ce5-11e7-bf9d-001999f8d30b

Only strip shared libraries if we are actually install them.

This fixes the build when the SHARED option is off.

Reported by:	Dmitry Arkhireev <che at teftel.ru>

Fix net/pjsip build with libressl.

PR:		216898
Submitted by:	OlivierW <olivierw1+bugzilla-freebsd@hotmail.com>

- Update pjsip to 2.6
- Add new WEBRTC option, disabled by default
- Make audio/speexdsp a dependency of the SPEEX option, reported
  by poudriere
- Regenerate some patches
- Bump net/asterisk13 PORTREVISION, I observed crashed when updating
  the pjsip libraries "below" it

multimedia/ffmpeg: update to 3.2.2

Changes:	https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n3.2.2:/Changelog
PR:		207547
Submitted by:	riggs, ebirth@b0ss.net (libressl fix)
Exp-run by:	antoine (3 tries)

- Add PJSUA option (on by default) to install the pjsua client binary
- Strip libraries
- Reflowed OPTIONS_DEFINE

PR:		211533
Submitted by:	Joseph Mingrone <jrm at ftfl.ca>

Update to 2.5.5.

PR:		211533
Submitted by:	Joseph Mingrone <jrm at ftfl.ca>

multimedia/openh264: update to 1.6.0

API/ABI has changed, so bump PORTREVISION in consumers. Some of those
are marked BROKEN that may need a fix similar to libav.

https://git.libav.org/?p=libav.git;a=commitdiff;h=d825b1a53065

Changes:	https://github.com/cisco/openh264/releases/tag/v1.6.0

- Update to 2.5.1
- Remove duplicate IPV6 opption in OPTIONS_DEFINE

- Update net/pjsip to 2.5 [1]
- Disable unneeded ALSA support in pjsip [1]
- Replace custom patch with USES=pathfix
- Fix pjsip build system to allow building while previous version
  is installed in PREFIX/LOCALBASE
- Bump dependent port asterisk13 PORTREVISION to avoid runtime crash
  (seen while testing)

PR:		209477 [1]
Submitted by:	yuri at rawbw.com

- Update net/asterisk13 to 13.8.1
- Change PJ_IOQUEUE_MAX_HANDLES build time limit in pjsip as suggested
  by asterisk project [1] to mitigate potential DoS [2]
- Add DEBUG and IPV6 options to pjsip port

Obtained
from:	https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject
[1]
Security:	ee50726e-0319-11e6-aa86-001999f8d30b
		e21474c6-031a-11e6-aa86-001999f8d30b [2]
MFH:		2016Q2

Remove ${PORTSDIR}/ from dependencies, categories m, n, o, and p.

With hat:	portmgr
Sponsored by:	Absolight

- Update pjsip to 2.4.5
- Convert to USES=localbase
- Bump asterisk PORTREVISION, it needs to be rebuild after this update.

Update portaudio to v19/Remove portaudio2 [1]
Chase portaudio change
Add patches from debian for games/cultivation
Add patches from upsteam for audio/rezound
Mark py-fastaudio as broken

Approved by:	maintainer

Add a slave port to net/pjsip to force installing pjsip with external
SRTP library.

Make the www/asterisk13 depend on this slave port when both SRTP
and PJSIP options in it are enabled, this allows enabling SRTP
support in asterisk13 without the need to manually reconfigure other
ports.

Reported by:	mat@ and a few others

Modify asterisk UPDATING entry to reflect the present situation.

While here also modify misleading and stale comment in the net/pjsip
port EXTSRTP option.

Thanks to mat@ for making me notice these.

Due to recent changes in OpenSSL, disable SRTP support for asterisk13
by default and avoid pjsip pulling in libsrtp, otherwise a not
working package would be generated.

Add note to UPDATING to keep users informed.

- Make pjsip use ports provided libsrtp instead of boudled one
- Fix asterisk13 SRTP support
- Fix asterisk13 SPEEX_LIB_DEPENDS
- While here make SRTP option default for asterisk13 since it does
  not add dependencies

PJSIP is a free and open source multimedia communication library
written in C language implementing standard based protocols such
as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol
(SIP) with rich multimedia framework and NAT traversal functionality
into high level API that is portable and suitable for almost any
type of systems ranging from desktops, embedded systems, to mobile
handsets.

WWW: http://www.pjsip.org/

Please note that default options are tailored for use by the upcoming
asterisk13 port.