path: root/docs/ChangeLog

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