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Import baresip-0.4.11 as wip/baresip. Baresip is a portable and modular SIP User-Agent with audio and video support. Features * Audio codecs: AMR, BV32, Celt, G.711, G.722, G.722.1, G.726, GSM, iLBC, iSAC, L16, OPUS, Silk, Speex * Video codecs: H.263, H.264, MPEG4, VP8 * Audio drivers: Alsa, Coreaudio, Gstreamer, Symbian MDA, OpenSLES, OSS, Portaudio, Windows wave * Video sources: FFmpeg avformat, MacOSX qtcapture, MacOSX quicktime, Video4Linux and Video4Linux2, X11 Grabber * Video output modules: OpenGL, SDL/SDL2, X11, DirectFB * NAT Traversal modules: STUN, TURN, ICE, NAT-PMP * Media encryption modules: SRTP, DTLS-SRTP, ZRTP TODO: Builds and packages, but doesn't run: baresip v0.4.11 Copyright (C) 2010 - 2014 Alfred E. Heggestad et al. config: creating config template /home/wiz/.baresip/config /usr/pkg/lib/baresip/modules: detected 21 modules _res is not supported for multi-threaded programs. zsh: abort (core dumped) baresip
2014-08-10 00:26:01 +00:00
$NetBSD: distinfo,v 1.1 2014/08/10 00:26:01 thomasklausner Exp $
baresip: update to baresip-0.6.2 2019-04-19 Alfred E. Heggestad * Version 0.6.2 Alfred E. Heggestad (124): daala: remove module remove USE_VIDEO compile flag (#658) config: remove sip_trans_bsize option contact: fix bug in contact prev/next cmd: remove unused complete flag log: add command to toggle loglevel ('v') debug_cmd: fix warning Remove natbd module (#659) message: make listen/unlisten more robust (ref #650) update doxygen comments srtp: fix warnings update README stream: define port 9 as PORT_DISCARD add offerer flag to video and stream stream: check for multiplexed RTCP packets on RTP port stream: change logic for rtcp-mux attribute omx: update doxygen comment bv32: add doxygen header mk: modules in alphabetical order mk: modules in alphabetical order mk: modules in alphabetical order h265: use avcodec API for the encoder h265: fixes for Debian 8 h265: make it work with old encoder api h265: init time_base manually fix av_packet_free allocate avpacket h265: cleanup cleanup cleanup h265: add configurable decoder cleanup use pkg-config for libs h265: update documentation mk: enable h265.so if avcodec installed h265: include avutil mem.h travis: use ubuntu 16.04 h265: add wrapper for av_frame_alloc h265: add wrapper for avcodec_free_context h265: fix config fix crash with ffmpeg 2.8 add wrapper for av_packet_free fix warning use av_free_packet cleanup deprecate v4l.so -- use v4l2.so instead h265: tested with YUV444P pixel format h265: check pixel format on changes Merge remote-tracking branch 'origin/master' into h265_use_avcodec_encoder h265: fix avcodec_free_context wrapper H265 use avcodec encoder (#668) debian: add source format 1.0 aufile: add sample config (ref #663) v4l: remove module, use v4l2.so instead video: remote orient parameter video: remove video_set_orient aubridge: remove audio resampler aubridge: fix warning aubridge: add support for multiple sample formats aubridge: fix warnings auloop: remove usage of audio codec jack: add support for FLOAT sample format Avcodec remove libx264 (#671) config: add avcodec.so sample config stream: set rtcp-mux attribute if enabled sdl: remove module stream: send a dummy RTCP packet to open NAT pinhole avcodec: use pkg-config for linker flags avformat: use pkg-config for linker flags config: remove usage of USE_AVCODEC coreaudio: remove ios specific code h265: one file per line avformat: move AVCodec from struct to stack avformat: remove codec_id check avformat: minor cleanup debian: remove usage of shlibs:Depends from dev package debug_cmd: fix warning avformat: minor cleanup avformat: minor cleanup vidloop: rename intra to keyframe vidloop: print keyframes only if codec is enabled avcodec: move destructor to the top of the file plc: check input arguments auloop: rename ab to aubuf Opus multistream (#678) mqtt: minor updates mqtt: use re_snprintf mqtt: update documentation (fixes #669) h264: fix h264_is_keyframe, IDR_SLICE is keyframe avcodec: check input arguments vidloop: show video display pixel-format in summary avcodec: clean up decoding code avcodec: add color range MPEG h265: add color range and GOP size ffmpeg: check avutil version for color range avcodec: set slice-max-size in H264 packetization-mode 0 avcodec: add sdp.c avcodec: move h264_fmtp_cmp to sdp.c avcodec: add support for H264 packetization mode 1 av1: update comment avcodec: handle H264 STAP-A packets test: fix ua register test-cases (ref #680) ua: add delayed_close flag (ref #680) menu: call uag_current() directly video: save pixel format of outgoing stream aulevel: add support for sample format FLOAT Vidfilt add param (#682) test: copy uri_cmp source from libre sdl2: handle window closed event (SDL_QUIT) vidloop: stop loop if window was closed sdl2: add support for quit key avcodec: fixes for packetization_mode 1 bump version to 0.6.2 vidfilt: update doxygen comments rtcpsummary: fix warnings about unused variables mqtt: fix warnings about unused variables gst_video1: use GST_BUFFER_PTS mk: add echo module to list of basic modules core: remove some unused values menu: call parameter is used mwi: minor formatting changes audio: align debug text Update README.md ua: add ua_destroy() -- ref #686 Andreas Hansson (1): Added debug cmd to print UUID (#674) Juha Heinanen (2): do not add basic modules if BASIC_MODULES has value 'no' exclude more modules if BASIC_MODULES=no Nicolas Tizon (1): vidloop: update vidsrc (#662) Roger Sandholm (1): Readme and config example correction, httpd comments (#666) Timmo Verlaan (1): menu: add uadel to delete a uac by aor (#680) juha-h (1): srtp: added sending of MENC_EVENT_SECURE event (#660) premultiply (1): Add 48 khz sampling rate support (#685) weili-jiang (1): Command not found returns error (#664)
2019-05-16 20:39:12 +00:00
SHA1 (baresip-0.6.2.tar.gz) = 0d3c459d48fb6a5e6d4cc6f350c4f9617fb732b3
RMD160 (baresip-0.6.2.tar.gz) = cf8ec653af0060b464ae51d0760c08369d8c0967
SHA512 (baresip-0.6.2.tar.gz) = b905bb88114a8a4a091d101db80a0b6c2e25f589e7f501916a324078c21c286d121a7247bbee7d2d904baeeb792051a499498605107ddb31160aaf8cc361b963
Size (baresip-0.6.2.tar.gz) = 600664 bytes
baresip: update to baresip-0.5.10 Chanelog: 2018-07-04 Alfred E. Heggestad * Version 0.5.10 * GIT URL: https://github.com/alfredh/baresip.git * GIT tag: v0.5.10 * NOTE: Requires libre v0.5.7 or later Requires librem v0.5.3 or later * build: - Updated MSVS project (thanks Encamy) * baresip-core: - account: add more accessor functions (thanks Juha Heinanen) - audio: add audio_set_hold - aufilt: add struct audio parameter - mediaenc: add menc_event handler (thanks Juha Heinanen) - net: add support for IP-address in 'net_interface' (thanks @Encamy) - stream: add stream_call - stream: check SDP_SENDONLY flag - stream: correct flags in stream_send (thanks Andreas Hansson) - stream: set RTP socket buffersize to 65536 (ref #415) - ua: add events for VU level (thanks Ola Palm) - ua: add ua_update_account - ua: don't append domain if uri is IP address (thanks Ali Shirvani) - ui: add ui_input_long_command - videnc: add timestamp parameter - video: add video_calc_rtp_timestamp_fix - video: lock when setting encoder (ref #418) (#441) * Modules: * aufile: add slow cpu detection * auloop: add samplerate and channels argument to command * av1: add timestamp parameter to encode function * avcodec: add timestamp parameter to encode function set baseline profile on ffmpeg H.264 encoder remove checks for old versions of libx264 * dshow: fix build for VC and mingw (thanks @Encamy) add picture vertical flipping (thanks Nicolas Tizon) * dtls_srtp: add usage of medienc event handler * gst_video: add timestamp parameter to encode function * gst_video1: add timestamp parameter to encode function * h265: add timestamp parameter to encode function * httpd: no echoing of long commands * menu: add video switch command /vidsrc (thanks Ali Shirvani) * opensles: check state before calling Destroy * sdl2: print renderer info * vidloop: refactoring of timestamp routines * vp8: add timestamp parameter to encode function * vp9: add timestamp parameter to encode function * vumeter: add periodic events (thanks Ola Palm) * zrtp: add usage of medienc event handler (thanks Juha Heinanen)
2018-07-04 18:15:17 +00:00
SHA1 (patch-mk_modules.mk) = 70284955c9f5fc42a3f86680eff26a0a0ba514e7
baresip: update to baresip-0.4.20 Pkgsrc changes: * change the build procedure a bit to accommodate modules * use the options framework top optionally build a number of modules * install example configuration files Notes: On my NetBSD machine the "oss" and "portaudio" modules do not work correctly. This is probably due to the fact that the module opens and initializes the device twice (once for reading and once for writing) but I was unable to diagnose the exact problem. A workaround is to use the jack plugin (or pulseaudio) by enabling the "jack.so" module in ~/.baresip/config. The jack server can start automatically: echo "/usr/pkg/bin/jackd -T -r -d sun" > ~/.jackdrc export JACK_START_SERVER=yes The sndiod module currently does not build. Changelog: 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)
2016-09-06 08:13:31 +00:00
SHA1 (patch-modules_ilbc_ilbc.c) = b19d181f41d84ad5cdc2a6e93c5004dab25e6c32
SHA1 (patch-modules_v4l2_v4l2.c) = 71ba2d1e5c8ba61eb011bd2b6b9e0d9cdaec5797