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2017-05-14 Alfred E. Heggestad <alfred.heggestad@gmail.com>

	* Version 0.5.3

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.5.3
	* NOTE: Requires libre v0.5.3 or later
	        Requires librem v0.5.0 or later

	* config:
	  - (no changes)

	* build:
	  - detect jack module (thanks Tony Langley)
	  - Updated MSVS projects to vs2015 (thanks Mikhail Barg)

	* baresip-core:
	  - aulevel: add aulevel_calc_dbov()
	  - audio: Set correct clock rate for telephone events
		   (thanks Jan Hoffmann)
	  - play: Add gapless repeat for tone playback (thanks Jan Hoffmann)

	* selftest:
	  - add tests for aulevel
	  - add tests for audio player
	  - add mock aucodec/auplay

	* Modules:

	* gst_video1: Tune x264enc for low latency (thanks Jonathan Sieber)

	* httpd: fix a crash

	* ice: update to latest libre ICE-api

	* omx: Fixed some problems on OMX/RaspberryPi (thanks Jonathan Sieber)

	* srtp: fix SRTP for early-media (thanks Jan Hoffmann)

	* vumeter: use aulevel_calc_dbov to calculate signal energy

	* zrtp: update to latest libzrtp from freeswitch (thanks Juha Heinanen)


2017-04-07 Alfred E. Heggestad <alfred.heggestad@gmail.com>

	* Version 0.5.2

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.5.2
	* NOTE: Requires libre v0.5.0 or later
	        Requires librem v0.5.0 or later

	* new modules:
	  - omx    OpenMAX IL video display module (thanks Jonathan Sieber)

	* config:
	  - (no changes)

	* baresip-core:
	  - aucodec: make the API re-entrant
	  - aufilt: make the API re-entrant
	  - auplay: make the API re-entrant
	  - ausrc: make the API re-entrant
	  - video: using a video-source is now optional

	* Modules:

	* avformat: add pixelformat AV_PIX_FMT_YUVJ420P (Thanks Gary Metalle)

	* cairo: print picture info, use grey background

	* dtmfio: check fd before calling fclose (thanks Richard Perez)

	* h265: enable YUV444P pixelformat

	* oss: fix build for Solaris 11

	* speex: mark the module as deprecated, see speex.org


2017-03-04 Alfred E. Heggestad <alfred.heggestad@gmail.com>

	* Version 0.5.1

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.5.1
	* NOTE: Requires libre v0.5.0 or later
	        Requires librem v0.5.0 or later

	* new modules:

	* config:
	  - stunuser		STUN username for STUN/TURN/ICE
	  - stunpass		STUN password for STUN/TURN/ICE
	  - snd_path		Path to sndfile audio dump files

	* baresip-core:
	  - account: add more accessor functions
	  - account: add 'stunuser' and 'stunpass'
	  - commands: make the struct commands opaque
	  - message: make the API re-entrant, multiple listeners
	  - menc: make the API re-entrant
	  - mnat: make the API re-entrant

	* selftest:
	  - add tests for account
	  - add tests for message

	* Modules:

	* amr: use MOD-CFLAGS instead of global CFLAGS

	* avcodec: added optional config 'avcodec_h264dec' to specify hardware
		   accellerated FFmpeg decoder (thanks Harald Gutmann)

	* avformat: remove blocking sleep, use packet timestamp to
		    pace video stream (thanks Harald Gutmann)

	* debug_cmd: add OpenSSL version to systems info

	* gtk: fix build where USE_NOTIFICATIONS is not defined
	       get rid of system header warnings by using -isystem

	* httpd: add support for un-escaping of URL parameters
		 (thanks to elektm93)

	* menu: add new command 'ausrc' to switch audio source
		add new command 'auplay' to switch audio player

	* sdl2: add more pixelformats (ref #202)
		(thanks Harald Gutmann)

	* sndfile: add config to specify path for dump files (thanks Elektm93)
		   add test for sndfile on *BSD. (#194) (thanks jungle-boogie)

	* swscale: get dst-size from config (ref #203)

	* v4l2_codec: Video device selection bug (#218)
		      (thanks Richard Perez)


2016-12-23 Alfred E. Heggestad <alfred.heggestad@gmail.com>

	* Version 0.5.0

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.5.0
	* NOTE: Requires libre v0.5.0 or later
	        Requires librem v0.5.0 or later

	* new modules:
	  - av1		Experimental AV1 video codec
	  - debug_cmd	Debug commands for advanced users
	  - pcp		Port Control Protocol (PCP) for NAT traversal
	  - swscale	Video scaling using FFmpeg's libswscale

	* config:
	  - call_max_calls	Maximum number of calls per account

	* baresip-core:
	  - call: add multiple lines
	  - call: start video on reinvite (thanks Gary Metalle)
	  - cmd: add support for long commands
	  - cmd: make it re-entrant
	  - config: add some modules to template (thanks Dmitrij D. Czarkoff)
	  - contact: make it re-entrant
	  - play: make it re-entrant
	  - vidcodec: add a intraframe-flag to api
	  - video: resend FIR until Intra frame received

	* selftest:
	  - add test for DTMF in call
	  - add test for contacts
	  - add test for long commands
	  - add test for maximum calls
	  - add test for multiple calls
	  - add test for video call
	  - add audio-source mock
	  - add video-codec mock
	  - add video-display mock
	  - add video-source mock

	* Modules:

	* aufile: convert samples from little-endian to host-endian

	* auloop: use long commands /auloop and /auloop_stop

	* av1: new module for Experimental AV1 video codec

	* avcodec: add config option 'avcodec_h264enc' to set encoder name
		   (thanks to @hargut)

	* avformat: fix init and warnings (thanks Maciej Koman)

	* b2bua: use long command /b2bua

	* contact: use long commands

	* debug_cmd: new module for advanced debug commands

	* g7221: expose spandsp api (thanks to Steve Underwood)

	* gtk: use long command /gtk

	* h265: add 'profile-id=1' to SDP

	* menu: add long commands
		add command 'line' or '@' to set current call

	* opengl: fix deprecated warnings on OSX 10.12

	* opensles: add support for stereo
		    (thanks to Juha Heinanen and Vijay Pratap Singh)

	* opus: add support for SDP parameter mirroring
		(thanks to Sveriges Radio)

	* pcp: new module for Port Control Protocol (PCP) NAT traversal
	       requires librew (https://github.com/alfredh/rew)

	* plc: expose spandsp api (thanks to Steve Underwood)

	* presence: add long commands /presence_{on,off}line

	* snapshot: use long commands (thanks Dmitrij D. Czarkoff)

	* sndio: use driver-suggested buffer size (thanks Dmitrij D. Czarkoff)

	* swscale: new module for video filter using libswscale

	* v4l2: pick up VID_FMT_NV12 and VID_FMT_NV21 formats as well (#176)
		don't check for native/emulated format (#179)
		(thanks Dmitrij D. Czarkoff)

	* vidloop: use long commands

	* vp8: add 'intra' parameter to decoder api
	       fix building with old versions of libvpx

	* wincons: graceful closing of thread (fixes #151)
		   (thanks to @GGGO)

	* zrtp: use long command


2016-07-22 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.20

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.4.20
	* NOTE: Requires libre v0.4.17 or later
	        Requires librem v0.4.7 or later

	* new modules:
	  - pulse      Pulseaudio driver
	  - vp9        VP9 video codec

	* config:
	  - audio_path          Path to audio files
	  - call_local_timeout  Timeout for incoming calls
	  - redial_attempts     Number of redial attempts
	  - redial_delay        Redial delay in seconds

	* baresip-core:
	  - baresip: added a global baresip instance (WIP)
	  - call: add RTP timeout (thanks to Sveriges Radio)
	  - config: added call_local_timeout for incoming call timeout
	  - config: added compile-time configureable CONFIG_PATH
	  - config: added 'audio_path' config variable (thanks Juha Heinanen)
	  - net: made it re-entrant with struct network
	  - ua: added uag_set_exit_handler
	  - ua: fix bug with reg_uri limited to 64-chars
	  - video: vidfilters should not modify decoded image

	* selftest:
	  - add test for network
	  - add test for sending SIP OPTIONS
	  - add test for RTP timeout

	* Modules:

	* avcodec: fix usage of deprecated API

	* avformat: remove support for scaling
		    fix usage of deprecated API

	* cons: relay log-messages to active UDP/TCP connections
		https://github.com/alfredh/baresip/issues/144

	* h265: fix usage of deprecated API

	* menu: added support for re-dial on failure
		(thanks to Sveriges Radio)

	* mpa: Bug with reinit of codec structs (thanks Christian Hoene)

	* natpmp: added support for RTCP

	* presence: use correct struct in deref handler

	* pulse: new module for Pulseaudio driver
		 (thanks to Matthias Apitz for testing)

	* vidloop: vidfilters should not modify decoded image

	* vp8: module renamed from vpx.so to vp8.so

	* vp9: new module implementing VP9 video codec

	* wincons: use ReadConsoleInput, thanks to GGGO (fixes #139)
		   https://github.com/alfredh/baresip/issues/139


2016-05-20 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.19

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.4.19
	* NOTE: Requires libre v0.4.14 or later
	        Requires librem v0.4.7 or later

	* new modules:
	  - mpa        MPA Speech and Audio Codec (thanks Christian Hoene)

	* baresip-core:
	  - audio: remove is_g722 exception
		   use aucodec's rtp clockrate for calculating RTP timestamp
		   plc: make sure sampc is exactly one ptime frame
	  - aucodec: split srate into DSP srate and RTP clockrate
		     (these are different for e.g. G.722 and MDA)
	  - mos: add mos_calculate() (thanks Lorenzo Mangani)
	  - net: use configured dns servers only, if specified
	  - ua: fix potential NULL-pointer crash for uag.cfg

	* selftest:
	  - add test for SIP registration with DNS
	  - add test for SIP registration with authentication
	  - add test for MOS calculations
	  - added a mock DNS Server
	  - added a mock SIP Server

	* Modules:

	* aucodec: add support for NV12 and YUVJ420P pixel formats

	* daala: update to libdaala version 0.0-1564-g79787c7

	* gtk: fix autodetection of libgtk+ 2.0 (thanks Charles Lehner)

	* h265: remove call to x265_cleanup, caused crash on OpenBSD

	* mpa: new module that implements MPA Speech and Audio Codec
	       (this module was contributed by Christian Hoene)

	* opus: added new configuration parameters:
		opus_cbr        {yes,no}   # Constant Bitrate (inverse of VBR)
		opus_inbandfec  {yes,no}   # Enable inband FEC
		opus_dtx        {yes,no}   # Enable DTX

	* presence: improved interoperability, allow white space before
		    xml element closing tags (thanks Juha Heinanen)

	* x11: added borderless window (thanks Doug Blewett)


2016-03-12 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.18

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.4.18
	* NOTE: Requires libre v0.4.14 or later
	        Requires librem v0.4.7 or later

	* baresip-core:
	  - call: fix SIP INFO with dtmf-relay (thanks Gary Metalle)
	  - ua: add event UA_EVENT_CALL_CLOSED for ua_hangup()

	* selftest:
	  - add tests for answer a call and hangup

	* Modules:

	* alsa: fix potential crash (thanks Gary Metalle)

	* audiounit: fix compilation for iOS (issue #91)

	* avcodec: fix compilation for FFmpeg 3.0

	* avformat: fix compilation for FFmpeg 3.0

	* gtk: always handle incoming calls (thanks Charles Lehner)

	* h265: fix compilation for FFmpeg 3.0

	* menu: add config 'menu_bell  off/on' to enable Bell alert
		add command 'A' for switch audio device (thanks AlexMarlo)

	* v4l2_codec: add list of encoders (fixes #99)


2016-01-17 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.17

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT tag: v0.4.17
	* NOTE: Requires libre v0.4.14 or later
	        Requires librem v0.4.7 or later

	* new modules:
	  - echo        Echo server module
	  - jack        JACK Audio Connection Kit audio-driver

	* baresip-core:
	  - config: keep config object in memory
	  - ua: moved playing of ringtones out of core, to "menu" module
		(let's keep the core nice and slim..)
	  - ui: added ui_password_prompt()

	* selftest:
	  - silence debug/info log by default, only print warnings
	    (use -v to see verbose logging)

	* Modules:

	* alsa: added config option to specify the sample format
		"alsa_sample_format    {s16,float,s24_3le}"
		thanks to Ola Palm for valuable feedback

	* audiounit: fix recording on OSX (thanks Sebastian Reimers)
		     print hardware samplerate in debug mode

	* auloop: add support for 44100 Hz samplerate

	* daala: update to latest libdaala API (thanks Dmitrij D. Czarkoff)

	* echo: new module which implements a simple Echo-server, to
		be used in combination with the aubridge.so module.
		contributed by Sebastian Reimers

	* gtk: fixes to support C89 compiler (thanks Dmitrij D. Czarkoff)

	* jack: new module which implements audio-driver for JACK

	* menu: playing of ringtones moved here, from ua.c

	* sndio: fix crash when device open fails (thanks Dmitrij D. Czarkoff)


2015-12-01 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.16

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT commit bed2241da3261e472f09b21958f0cc1324a94f27
	* GIT tag: v0.4.16
	* NOTE: Requires libre v0.4.14 or later

	* new modules:
	  - v4l2_codec  Video4Linux2 video codec (H264 hardware encoding)
	  - vidinfo     Video info overlay module

	* baresip-core:
	  - audio: add audio_set_source() and audio_set_player()
	  - audio: flush tx-buffer for all modes (thanks Thibault Gueslin)
	  - call: add call_is_outgoing()
	  - call: check address-family of incoming SDP offer (thanks Olle)
	  - h264: move H.264 packetization code to core
	  - main: add -u option to append extra global UA parameters
	  - main: pre-load modules after all arguments are parsed
	  - ua: add events UA_EVENT_SHUTDOWN,UA_EXIT
	  - ua: add ua_hold_answer()
	  - ua: add ua_set_media_af()
	  - ua: delay mod-unloading if mods has a ref to struct ua

	* build:
	  - add verbose build with V=1 (thanks Dmitrij D. Czarkoff)
	  - add pkg-config file (thanks William King)
	  - add travis.yml file for Github build-system

	* Modules:

	* alsa: fix memory leaks

	* avcodec: move common H.264 packetization code to core

	* cairo: use pkg-config in makefile

	* daala: update to latest libdaala (thanks Dmitrij D. Czarkoff)

	* gst_video: use H.264 packetization API from core

	* gst_video1: use H.264 packetization API from core

	* gtk: fix segmentation fault on window close

	* mwi: add 500ms delay after closing subscription

	* oss: use pthread for ausrc instead of fd_listen (fixes FreeBSD)

	* presence: use sipevent_sock instance from UA core
		    add 500ms delay after closing subscription

	* v4l2_codec: new module

	* vidinfo: new module

	* zrtp: fix ZRTP over TURN by moving helper to layer 10
		fix ZID verification (thanks Ingo Feinerer)


2015-09-26 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.15

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT commit 86262a6fc17e19e2be82eb8a2a05ec0f884d3d38
	* GIT tag: v0.4.15
	* NOTE: Requires libre v0.4.13 or later

	* added selftest binary

	* baresip-core:
	  - audio: fix televent when pt != 101 (reported by AndyJRobinson)
	  - magic: use __func__ for C99 or later
	  - sip: make sip_req_send() public
	  - ua: add UA_EVENT_CALL_DTMF_START/END, thanks Gary Metalle

	* Modules:

	* alsa: added extra logging

	* gtk: add support for libnotify (thanks Charles Lehner)

	* video: fix potential null deref (thanks Tomasz Ostrowski)

	* zrtp: added 36-bytes preamble for TURN-header


2015-08-08 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.14

	* GIT URL: https://github.com/alfredh/baresip.git
	* GIT commit ebac23b0692de71ee4c3a436f0372013150c937f
	* GIT tag: v0.4.14
	* NOTE: Requires libre v0.4.13 or later

	* new modules:
	  - gtk		GTK+ 2.0 UI (thanks Charles E. Lehner)
	  - gst1	Gstreamer 1.0 audio module
	  - gst_video1	Gstreamer 1.0 video module (thanks Thomas Strobel)
	  - daala	Experimental video-codec using Daala

	* baresip-core:
	  - baresip: added -m argument to pre-load modules
	  - config: add kqueue to sample config (thanks Dmitrij D. Czarkoff)
	  - log: make code C89 compliant (thanks Victor Sergienko)
	  - module: added module_preload()
	  - ua: add CALL_EVENT_TRANSFER_FAILED
	  - ua: skip initial white space from uri (thanks Juha Heinanen)
	  - ua: ua_prev_call()
	  - videnc: move videnc_packet_h to update-handler

	* build:
	  - added optional $(MOD)_CFLAGS for local module CFLAGS
	  - added project file for Visual C++ Express 2010
	  - freebsd: add include path to $(SYSROOT)/local/include
	    (thanks Hellmuth Michaelis)

	* Modules:

	* avcodec: make code C89 compliant (thanks Victor Sergienko)

	* cons: make code C89 compliant (thanks Victor Sergienko)

	* daala: new module

	* dshow: updates for VC2010 (thanks Victor Sergienko)

	* gst1: new module

	* gst_video1: new module

	* gtk: new module

	* menu: fix crash when 0 UAs (thanks Hans Petter Selasky)
		added command 'H' to hold previous call (thanks xanm)

	* wincons: make code C89 compliant (thanks ggcoding)


2015-06-20 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.13

	* GIT commit 2e3e825ef5532dfde5a8b52de9ebaac51aa20a9c
	* NOTE: Requires libre v0.4.12 or later

	* new modules:
	  - aufile      Audio module for using a WAV-file as audio input
	  - b2bua       Back-to-Back User-Agent (B2BUA) module
	  - codec2      CODEC2 audio codec
	  - gst_video   Gstreamer video codec
	  - h265        H.265 (HEVC) video codec

	* baresip-core:
	  - contact: add support for access-control (thanks Doug Blewett)
	  - ausrc: change base-class to a const pointer
	  - auplay: change base-class to a const pointer
	  - vidsrc: change base-class to a const pointer
	  - vidisp: change base-class to a const pointer
	  - video: smooth sending of video packets


	* Modules:

	* amr: added support for octet-align mode (thanks to Stefan Sayer)

	* aubridge: copy audio-samples if resampler not needed

	* aufile: new module for using a WAV-file as audio source

	* avcapture: only register 1 video source

	* avformat: fix segfault on recent versions of libav

	* b2bua: new experimental module

	* codec2: new module for CODEC2 audio codec

	* dtls_srtp: uppercase fingerprint, interop (thanks Juha Heinanen)
		     alternative SDP protocols for interop

	* dtmfio: unregister event handler on close (thanks Hellmuth Michaelis)

	* gst_video: new module using Gstreamer as a video codec
		     (Thanks to Victor Sergienko and Fadeev Alexander)

	* h265: new module for H.265 video codec

	* httpd: added raw mode (thanks Lorenzo Mangani)

	* menu: create user-agent with a command 'R' (thanks Lorenzo Mangani)

	* opus: add configuration of "opus_bitrate"
		(thanks to Juha Heinanen)

	* speex: add configuration of "speex_mode_nb" and "speex_mode_wb"
		 (thanks to Dmitrij D. Czarkoff and Juha Heinanen)

	* vidloop: add VIDLOOP_INTERNAL_FMT and split encoder/decoder

	* x11: catch Window delete (thanks to Doug Blewett)

	* zrtp: initialize remote_zid (thanks to Ingo Feinerer)


2014-12-24 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.12

	* GIT commit 67993e35d980375458348b264c4a35a944bb5180
	* NOTE: Requires libre v0.4.11 or later

	* baresip:
	  - account: add regint and pubint
	  - audio: fix checking of sample-rate range
	  - config: remove the "input" block
	  - config: added support for quoted device parameters
	  - config: fix conversion of bandwidth to kbit/s
	  - config: generate more relevant config for FreeBSD and OpenBSD
		    (thanks Dmitrij D. Czarkoff)
	  - reg: add support for extracting GRUU parameter
	  - main: add -p option to set path to audio files
	  - sipreq: make response-handler optional
	  - ua: add support for GRUU (RFC 5627)
	    (many thanks to Juha Heinanen for starting this work and
	     helping out with the testing)
	  - ua: moved presence-status to each struct ua instance
	  - ua: add presence status to each User-Agent instance
	  - ua: use public-GRUU if set, otherwise local cuser
	  - ui: make UI single instance
	  - video: add VIDENC_INTERNAL_FMT (suggested by Victor Sergienko)

	* docs: added sample configuration files

	* account: added pubint for Publishing Interval

	* avcodec: upgrade to recent ffmpeg/libav APIs
		   either FFmpeg or libav can be used

	* celt: deleted module (replaced by opus)

	* cons: update usage of struct ui, added output handler
		added config: cons_listen    0.0.0.0:5555

	* evdev: update usage of struct ui, added output handler
		 added config: evdev_device    /dev/input/event0

	* httpd: added ui output handler

	* menu: added command 'o' for sending OPTION request
		(thanks to Juha Heinanen)

		added command 'D' for accepting incoming calls

	* mwi: subscribe to MWI after Registration succeeded
	       (thanks to Juha Heinanen)

	* opensles: add double-buffering and some tuning
		    (thanks to Francesco Bradascio)

	* opus: added config "opus_bitrate" (thanks to Sebastian Reimers)

	* presence: added support for PUBLISH (thanks to Juha Heinanen)
		    interop fixes and tuning

	* stdio: update usage of struct ui, added output handler

	* uuid: use internal version of generating UUID

	* v4l2: use memory mapped mode only

	* vumeter: dont call tmr_start from non-RE thread

	* wincons: update usage of struct ui, added output handler

	* winwave: fix bug when closing player device
		   (thanks to Tomasz Ostrowski)
		   add support for mapping device name to index

	* zrtp: add support for verify SAS (thanks to Ingo Feinerer)


2014-06-21 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.11

	* GIT commit 7a465f2eb92f4e32740093e5ad4970d528908c51

	* baresip:
	  - audio: added audio_ismuted() to get audio mute status
	  - audio: fix timestamp generation for stereo-streams
	  - audio: send outgoing audio-packets as soon as possible
	  - audio: upgrade to sample-based ausrc/auplay API
	  - auplay: change API to use samples instead of 8-bit buffer
	  - auplay: remove option to specify sample format (always S16LE)
	  - ausrc: change API to use samples instead of 8-bit buffer
	  - ausrc: remove option to specify sample format (always S16LE)
	  - call: added support for X-RTP-Stat header (thanks Lorenzo Mangani)
	  - call: check for common audio-codecs (thanks Juha Heinanen)
	  - logging: use info() instead of DEBUG_INFO();
	  - logging: use warning() instead of DEBUG_WARNING()
	  - play: convert WAV-file from little-endian to native-endian
	  - removed support for Symbian OS

	* debian: upgrade debian files

	* avcapture: also build for MacOSX

	* alsa: fix sample-endianess with SND_PCM_FORMAT_S16
		upgrade to sample-based ausrc/auplay API

	* audiounit: upgrade to sample-based ausrc/auplay API

	* auloop: upgrade to sample-based ausrc/auplay API

	* coreaudio: upgrade to sample-based ausrc/auplay API

	* dtls_srtp: use DTLS code from libre (needs libre v0.4.9 or later)
		     use SRTP code from libre (needs libre v0.4.9 or later)

	* dtmfio: new module to send DTMF-events via FIFO file
		  (contributed by Aaron Herting)

	* fakevideo: new module for fake video input/output driver

	* gst: upgrade to sample-based ausrc/auplay API

	* ice: set default candidates for ICE-lite

	* libsrtp: module 'srtp.so' renamed to 'libsrtp.so'

	* mda: Symbian MDA audio driver was deleted

	* menu: fix issue with audio-mute on multiple calls

	* opensles: upgrade to sample-based ausrc/auplay API

	* oss: upgrade to sample-based ausrc/auplay API

	* portaudio: upgrade to sample-based ausrc/auplay API

	* rst: upgrade to sample-based ausrc/auplay API

	* selftest: new module for testing the baresip core api

	* sndio: new module for OpenBSD audio driver
                 (It was contributed by Dmitrij D. Czarkoff, thank you!)

	* srtp: module is now using SRTP-stack from libre (v0.4.9 or later)

	* syslog: use logging framework to get messages

	* v4l2: add format negotiation and OpenBSD support
                (contributed by Dmitrij D. Czarkoff)

	* winwave: upgrade to sample-based ausrc/auplay API


2014-01-23 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.10

	* baresip:
	  - account: add account_set_display_name() -- thanks Dimitris
	  - audio: use both srate/channels to check if resampler is needed
	  - aufilt: change from frame_size to ptime
	  - auplay: change from frame_size to ptime
	  - ausrc: change from frame_size to ptime
	  - config: add optional ausrc_channels and auplay_channels
	  - config: create config dir with mode 0700 (suggested by Jann Horn)
	  - play: update auplay usage with ptime

	* alsa: update to new ausrc/auplay API with ptime
		fix bug when snd_pcm_readi() returns -EPIPE (thanks Remik)
		open device from main thread instead of alsa-thread (thanks EL)
		(caused problems with Sennheiser Century SC 660 + USB adapter)
	
	* auloop: minor cleanups and improvements

	* coreaudio: update to new ausrc/auplay API with ptime

	* gst: update to new ausrc/auplay API with ptime

	* l16: fix a bug with sample count

	* opus: fix a memory corruption error in opus_decode_pkloss()

	* oss: update to new ausrc/auplay API with ptime

	* plc: update to new aufilt API with ptime

	* portaudio: update to new ausrc/auplay API with ptime
		     fix bugs when using channels=2 (stereo)
		     configure device index using "device" parameter

	* rst: update to new ausrc/auplay API with ptime

	* speex_aec: update to new aufilt API with ptime

	* speex_pp: update to new aufilt API with ptime

	* winwave: update to new ausrc/auplay API with ptime

	* zrtp:	update to use libzrtp from Travis Cross' github
		use config dir to store ZRTP cache-file (thanks Juha Heinanen)
	
	
2014-01-06 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.9

	* new modules:
	  - zrtp  Media Path Key Agreement for Unicast Secure RTP

	* build:
	  - added support for LLVM clang compiler

	* baresip:
	  - account: add account_laddr()
	  - audio: upgrade to new librem auresamp API
	  - config: use oss,/dev/dsp as default device for FreeBSD
	  - log: added new logging framework
	  - main: added new verbose debug argument (-v)
	  - net: added sanity check for HAVE_INET6 build flag
	  - play: added play_set_path() -- thanks to Dimitris P.
	  - ua: added uag_find_param()
	  - ua: fix param-bug in ua_connect() -- thanks to Juha Heinanen

	* aubridge: upgrade to new librem auresamp API

	* avcodec: use new av_frame_alloc() api

	* celt: deprecate CELT-module, use OPUS instead

	* opengles: fix warnings (thanks to Dimitris P.)

	* opensles: fix bugs in player and recorder

	* opus: encode/decode sdp parameters as of I-D

	* speex_resamp: module removed, replaced by librem's resampler

	* zrtp: new module for ZRTP media encryption (use ;mediaenc=zrtp)


2013-12-06 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.8

	* new modules:
	  - dtls_srtp  DTLS-SRTP media encryption module (RFC 5763,5764)
	  - aubridge   Audio Bridge to connect auplay->ausrc
	  - vidbridge  Video Bridge module to connect vidisp->vidsrc

	* baresip:
	  - added RFC 5576  Source-Specific Media Attributes in SDP
	  - audio: set SDP bandwidth only if "rtp_bandwidth" config set
	  - play: do not store a copy of global config
	  - stream: save RTCP statistics from Sender-reports
	  - stream: add SDP ssrc attribute
	  - stream: added metrics for packets/bytes transmit/receive
	  - ua: added uag_current()/_set() to get/set current User-Agent
	  - video: set maximum RTP packet-size to 1024 bytes

	* config:
	  - added "video_display  module,device" for Video Display
	  - added "rtp_stats      {off,on}" for RTP Statistics after Call
	  - default RTP bandwidth is now 0-0

	* contact: dynamic command description for "Message" handling
		   dial from current UA (thanks to Simon Liebold)

	* isac: upgrade to draft-ietf-avt-rtp-isac-04

	* srtp: added auto-negotiation of RTP-profile for incoming calls
		(RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF)

	* vidloop: fix memory leak


2013-11-12 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.7

	* new modules:
	  - httpd   HTTP webserver UI module

	* baresip:
	  - added RFC 5506 Support for Reduced-Size RTCP
	  - audio: minor cleanups
	  - cmd: ignore RELEASE key in editor mode
	  - conf: add conf_get_sa()
	  - mnat: add address family (af) to session handler
	  - realtime: fixes for iOS (thanks Dimitris)
	  - ua: make ua_register() public
	  - ua: add ua_calls() to get list of calls
	  - ua: only create register client if regint > 0

	* debian: update dependencies (thanks Juha Heinanen)

	* rpm: added RPM package spec file

	* alsa: open device from thread to avoid blocking re-main loop

	* avcodec: build fixes for Debian Testing

	* avformat: use sys_msleep()

	* contact: improve matching logic (thanks EJC Lindner)

	* dshow: initialize variables (found with cppcheck)

	* evdev: fix formatted printing (found with cppcheck)

	* ice: use address family (AF) from call

	* ilbc: update to separate encoder/decoder states (thanks Dimitris)
	
	* snapshot: initialize variables (found with cppcheck)

	* stun: use address family (AF) from call

	* turn: use address family (AF) from call

	* uuid: fix usage of strncat()


2013-10-11 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.6

	* new modules:
	  - directfb   DirectFB video display module (thanks Andreas Shimokawa)
	  - dshow      Windows DirectShow vidsrc (thanks Dusan Stevanovic)
	  - wincons    Console input driver for Windows

	* baresip:
	  - audio: print audio-pipelines in console/debug
	  - aufilt: split into separate encoder+decoder states
	  - call: add local uri/name, dtmf-handler
	  - call: fix decoding of DTMF/SIP-INFO for '*' and '#'
	  - export CALL_EVENT_* in public API
	  - fix various clang warnings
	  - sipreq: use outbound proxy if specified (thanks EJC Lindner)
	  - ua: add possibility to specify 'struct call' for hangup/answer
	  - ua: move SIP extensions into a dynamic vector container
	  - ua: move playing of tones from call.c to ua.c
	  - vidfilt: split into separate encoder+decoder states
	  - vidisp: remove input handler

	* menu: improve call-transfer handling

	* plc: update to separate encoder/decoder states

	* selfview: update to separate encoder/decoder states

	* snapshot: remove state which was not needed

	* sndfile: update to separate encoder/decoder states
                   print unique timestamp to saved files

	* speex_aec: update to separate encoder/decoder states

	* speex_pp: update to separate encoder/decoder states

	* vidloop: update to separate encoder/decoder vidfilt states

	* vumeter: update to separate encoder/decoder states

	* wincons: new module for Console input on Win32


2013-08-31 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.5

	* new modules:
	  - account      Account loader module
	  - natpmp	 NAT-PMP client (RFC 6886)
	  - sdl2         Video display using libSDL2
	
	* baresip:
	  - account: added SIP account parser and container
	  - config: split conf.c into conf.c and config.c
	  - config: move enum audio_mode to struct config
	  - config: move uuid to struct config
	  - more usage of the #ifdef USE_VIDEO macro
	  - message: add handling of SIP MESSAGE send/recv
	  - mediaenc: added rtp_sock parameter to media-handler
	  - ua: cleanup public struct ua API
	  - vidisp api: remove unused 'parent' parameter
	  - call: handle incoming DTMF in SIP INFO (application/dtmf-relay)
	  - sdp: added sdp_decode_multipart()
	  - net: fix bug on IP-refresh when 'net_interface' is used
	  - video: minor cleanups
		   handle incoming RTCP_RTPFB_GNACK
	
	* isac: fix encode_update() signature

	* menu: move dialbuffer here from ua.c
		added command 'g' to print current config

	* mwi: multiple MWIs for multiple UAs

	* presence: include supported methods in SIP messages

	* srtp: improved interop and debugging
		handle incoming RTP/RTCP-demultiplexing

	* uuid: write loaded UUID directly to struct config

	* vidloop: added video-filters


2013-05-18 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.4

	* new modules:
	  - g726      G.726 audio codec
	  - mwi       Message Waiting Indication
	  - snapshot  Save video-stream as PNG images

	* config:
	  - added 'sip_certificate' to use a Certificate for SIP/TLS
	  - added 'ausrc_srate' and 'auplay_srate' to force DSP samplerate

	* baresip:
	  - added a simple BFCP client
	  - aufilt: improved API
	  - mediaenc: improved API with session state
	  - ua: added event handler framework
	  - aucodec: improved API with separate encode/decode state
	  - vidcodec: improved API with separate encode/decode state
	  - sdp.c: added SDP helper functions
	  - ua: move registration client to reg.c
	  - audio: added internal resampler

	* auloop: added config option 'auloop_codec' for setting codec

	* ice: remove old 'ice_interface' config option

	* menu: move handling of status-mode here

	* selfview: added config option 'selfview_size'

	* vp8: upgrade to draft-ietf-payload-vp8-08

	* winwave: cleanup and minor fixes


2013-01-01 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.3

	* new modules:
	  - selfview    Video selfview as video-filter module
	  - vumeter	Audio-filter module to display recording/playback level

	* config:
	  - added 'net_interface" to bind to a specific network interface
	  - added accounts 'regq' parameter for SIP Register client

	* baresip:
	  - added video-filter plugin API (vidfilt)
	  - audio.c: cleanups, split into transmit/receive part
	  - ua: added SIP Allow-header (thanks Juha Heinanen)
	  - ua: added Register q-value (thanks Juha Heinanen)
	  - ua: fix DTMF end event bug

	* avcodec: fix x264 fps bug (thanks Trevor Jim)

	* ice: only include ufrag/pwd in session SDP (thanks Juha Heinanen)

	
2012-09-09 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.2

	* new modules:
	  - auloop    Audio-loop test module
	  - contact   Contacts module
	  - isac      iSAC audio codec
	  - menu      Interactive menu
	  - opengles  OpenGLES video output
	  - presence  Presence module
	  - syslog    Syslog module
	  - vidloop   Video-loop test module

	* baresip:
	  - added support for call transfer
	  - added support for call waiting
	  - added multiple calls per user-agent
	  - added multiple registrations per user-agent
	  - cmd: added new command interface
	  - ua:  handle SIP Require header for incoming calls
	  - ui:  cleanup, use dynamic interactive menu
	
	* config:
	  - added 'audio_alert' for ringtones etc.
	  - added 'outboundX=proxy' for multiple outbound proxies
	  - added 'module_tmp' for temporary module loading
	  - added 'module_app' for application modules
	
	* avcodec: upgrade to latest FFmpeg and fix pts bug

	* natbd: register command 'z' for status

	* srtp: fix memleak on close

	* uuid: added UUID loader


2012-04-21 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.1

	* baresip: do not include rem.h from baresip.h
		   rename struct conf to struct config
		   vidsrc API: move size to alloc handler
		   aucodec API: change fmtp type to 'const char *'
				add SDP fmtp compare handler
		   vidcodec API: added enqueue and packetizer handlers
				 remove size from vidcodec_prm
				 remove decoder parameters from alloc
				 change fmtp type to 'const char *'
				 add SDP fmtp compare handler
		   remove aufile.c, use librem instead
		   audio: fix Telev timestamp (thanks Paulo Vicentini)
			  configurable order of playback/source start
		   ua_find: match AOR for interop (thanks Tomasz Ostrowski)
		   ua: more robust parsing for incoming MESSAGE
		   ua: password prompt (thanks to Juha Heinanen)
	
	* build: detect amr, cairo, rst, silk modules

	* config: split 'audio_dev' parameter into 'audio_player/audio_source'
		  order of audio_player/audio_source decide opening order
		  rename 'video_dev' parameter to 'video_source'
		  added optional 'auth_user=NAME' account parameter
		  (idea was suggested by Juha Heinanen)
	
	* alsa: play: no need to call snd_pcm_start(), explictly started when
		writing data to the device. (thanks to Christof Meerwald)

	* amr: 	more portable AMR codec
	
	* avcodec: automatic size from encoded frames
		   detect packetization-mode from SDP format
		   use enqueue handler
	
	* avformat: update to latest versions of ffmpeg
	
	* cairo: new experimental video source module

	* cons: added support for TCP

	* evdev: added KEY_KPx (thanks to ccwufu on OpenWRT forum)

	* g7221: use bitrate from decoded SDP format
		 added optional G722_PCM_SHIFT for 14-bit compat
	
	* rst: thread-based video source
	
	* silk: fix crash, init encoder, bitrate=64000 and complexity=2
	        (reported by Juha Heinanen)
	
	* srtp: decode SDES lifetime and MKI

	* v4l, v4l2: better module detection for FreeBSD 9
		     do not include malloc.h
		     (thanks to Matthias Apitz)

	* vpx: auto init of encoder
	
	* winwave: fix memory leak (thanks to Tomasz Ostrowski)

	* x11: add support for 16-bit graphics
	

2011-12-25 Alfred E. Heggestad <aeh@db.org>

	* Version 0.4.0

	* updated doxygen comments (thanks to Olle E. Johansson)

	* docs: added modules description

	* baresip: add ua_set_aumode(), configurable audio-tx mode
		   vidsrc API: added media_ctx shared with ausrc
		   ausrc API: add media_ctx shared with vidsrc
		   audio_encoder_set() - stop audio source first
		   audio_decoder_set() - include SDP format parameters
		   aufile: add PREFIX to share path (thanks to Juha Heinanen)
		   natbd.c: move code to a new module 'natbd'
		   get_login_name: check both LOGNAME and USER
		   ua.c: unique contact-user with address of struct ua
		   ua.c: find correct UA for incoming SIP Requests
		   ua_connect: param is optional (thanks to Juha Heinanen)
		   video: add video_set_source()
	
	* amr: minor improvements

	* audiounit: new module for MacOSX/iOS audio driver

	* avcapture: new module for iOS video source

	* avcodec: fixes for newer versions of libavcodec

	* gsm: handle packet-loss

	* natbd: move to separate module from core
	
	* opengl: fix building on MacOSX 10.7
		  (thanks to David Jedda and Atle Samuelsen)

	* opus: upgrade to opus v0.9.8

	* rst: use media_ctx for shared audio/video stream

	* sndfile: fix stereo mode
	

2011-09-07 Alfred E. Heggestad <aeh@db.org>

	* Version 0.3.0

	* baresip: use librem for media processing
		   added support for video selfview
		   aubuf, autone, vutil: moved to librem
		   ua: improved API
		   conf: use internal parser instead of fscanf()
		   vidloop: cleanup, use librem for processing

	* config: add video_selfview={pip,window} parameter	

	* amr: new module for AMR and AMR-WB audio codecs (RFC 4867)

	* avcodec, avformat: update to latest version of FFmpeg

	* coreaudio: fix building on MacOSX 10.5 (thanks David Jedda)

	* ice: fix building on MacOSX 10.5 (thanks David Jedda)

	* opengl: remove deps to libswscale

	* opensles: new module OpenSLES audio driver

	* opus: new module for OPUS audio codec

	* qtcapture: remove deps to libswscale

	* rst: new module for mp3 audio streaming

	* silk: new module for SILK audio codec

	* v4l, v4l2: remove deps to libswscale

	* x11: remove deps to libswscale, use librem vidconv instead

	* x11grab: remove deps to libswscale


2011-05-20 Alfred E. Heggestad <aeh@db.org>

	* Version 0.2.0

	* baresip: Added support for SIP Outbound (RFC 5626)
		   The SDP Content Attribute (RFC 4796)
		   RTP/RTCP Multiplexing (RFC 5761)
		   RTP Keepalive (draft-ietf-avt-app-rtp-keepalive-09)

	* config: add 'outbound' to sipnat parameter (remove stun, turn)
		  add rtpkeep={zero,stun,dyna,rtcp} parameter
		  audio_codecs parameter can now specify samplerate
		  add rtcp_mux for RTP/RTCP multiplexing on/off

	* alsa: set buffersize and fix samplesize (thanks to Luigi Rizzo)

	* avcodec: added support for MPEG4 video codec (RFC 3016)
		   wait for keyframe before decoding

	* celt: upgrade libcelt version and cleanups

	* coreaudio: fix buffering in recorder

	* ice: several improvements and fixes
	       added new config options

	* ilbc: handle asymmetric modes

	* opengl: enable vertical sync

	* sdl: upgrade to latest version of libSDL from mercurial

	* vpx: added support for draft-westin-payload-vp8-02

	* x11: handle remote display with optional shared memory

	* x11grab: new video-source module (thanks to Luigi Rizzo)

	* docs: updated doxygen comments