All notable changes to baresip will be documented in this file.
The format is based on Keep a Changelog, and this project adheres to Semantic Versioning.
- ci: add coverage workflow by @maximilianfridrich in baresip#2550
- cmake: fix win32 dbghelp by @sreimers in baresip#2552
- cmake/gtk3: make sure gtk3 libs are found by @landryb in baresip#2554
- pipewire: add pipewire module by @cspiel1 in baresip#2439
- audio: count TX underruns correctly (#2535) by @cspiel1 in baresip#2553
- variadic function fixes by @maximilianfridrich in baresip#2523
- ua: unescape incoming Refer-To header by @maximilianfridrich in baresip#2541
- pipewire/cmake: declare include dirs as system by @sreimers in baresip#2556
- sndio: re-add sndio module for OpenBSD by @landryb in baresip#2555
- Client cert renegotiation in http by @fAuernigg in baresip#2461
- aufile: joind already terminated thread frees stack by @cspiel1 in baresip#2557
- ccheck: c11 err handling exclude mutex_alloc by @sreimers in baresip#2560
- alsa: use atomic for run flag by @cspiel1 in baresip#2558
- Add instruction to build the doxygen docs by @gibix in baresip#2561
- fakevideo: use atomic for run flag by @cspiel1 in baresip#2565
- cmake/findOpus: fix Could NOT find OPUS (missing: OPUS_INCLUDE_DIR) by @jobo-zt in baresip#2568
- cmake/findsdl: fix Could NOT find SDL (missing: SDL_INCLUDE_DIR) by @jobo-zt in baresip#2571
- avformat, g722: add macro UNISTD switch support for Windows by @jobo-zt in baresip#2574
- v4l2: use atomic for run flag by @cspiel1 in baresip#2567
- mc: use atomic for run flag by @cspiel1 in baresip#2566
- avformat: use atomic for run flag by @cspiel1 in baresip#2564
- ausine: use atomic for run flag by @cspiel1 in baresip#2563
- aubridge: use atomic for run flag by @cspiel1 in baresip#2562
- config: add different options for audio and video jitter buffer by @sreimers in baresip#2569
- Revert "ua: unescape incoming Refer-To header (#2541)" by @maximilianfridrich in baresip#2577
- stream: log last RTP packet debug after 100ms by @sreimers in baresip#2587
- ci/sanitizers: exit on first undefined behavior by @sreimers in baresip#2589
- @landryb made their first contribution in baresip#2554
- @gibix made their first contribution in baresip#2561
- @jobo-zt made their first contribution in baresip#2568
Full Changelog: https://github.com/baresip/baresip/compare/v3.1.0...v3.2.0
- config: add net_af config setting by @juha-h in baresip#2490
- gzrtp: RX thread - safe stop by @cspiel1 in baresip#2492
- ci: avoid hardcoded OpenSSL path on macOS by @robert-scheck in baresip#2505
- fix cmake modules by @sreimers in baresip#2507
- cmake/mqtt: fix MOSQUITTO_LIBRARY by @sreimers in baresip#2508
- mc: send module event whenever receiver is stopped by @cspiel1 in baresip#2509
- menu: limit early audio TX streams by @cspiel1 in baresip#2503
- call: check if SIP UPDATE is allowed, but always update local media by @cspiel1 in baresip#2504
- account: increase line handler size to 1024 characters by @juha-h in baresip#2511
- cmake: avoid include of /usr/local/include by @cspiel1 in baresip#2506
- call,audio: respect SDP media dir on audio start similar to video by @cspiel1 in baresip#2501
- video: refactor paced and burst sending by @sreimers in baresip#2482
- ctrl_dbus,ice,png_vf: Fix format string usage by @maximilianfridrich in baresip#2517
- menu limit early video by @cspiel1 in baresip#2514
- play: flush of the aubuf directly before the replay starts by @cspiel1 in baresip#2512
- stream: fix setting of RTP tos for IPv6 by @cspiel1 in baresip#2527
- call: only flush audio stream when stream starts by @cspiel1 in baresip#2526
- menu: use busy tone when call declined (scode 603) by @cspiel1 in baresip#2529
- ua: incoming DTMF key=0 should be reported as DTMF end by @cspiel1 in baresip#2528
- video: fix possible 32bit overflow by @sreimers in baresip#2534
- ua: deref call on reset_transp fail by @maximilianfridrich in baresip#2532
- uag: avoid transport reset if local address has not changed by @juha-h in baresip#2537
- ci: add gcc-12 for Ubuntu 22.04 (ubuntu-latest) by @robert-scheck in baresip#2542
- docs: remove librem from README files by @robert-scheck in baresip#2543
Full Changelog: https://github.com/baresip/baresip/compare/v3.0.0...v3.1.0
- ua: allow custom headers in sip REGISTER request by @Koshub in baresip#2452
- merge rem into re by @alfredh in baresip#2442
- main: fix async init order (after config load) by @sreimers in baresip#2457
- ci: install pkg-config on mac-os by @cspiel1 in baresip#2459
- ci: remove rem in sanitizers and valgrind yml by @cspiel1 in baresip#2458
- video: fix vidqueue_poll list_move by @sreimers in baresip#2465
- Dshow fixes by @tomek-o in baresip#2467
- Moved adding of custom headers from ua_connect_dir to ua_call_alloc by @juha-h in baresip#2470
- Include also params to MESSAGE URI by @juha-h in baresip#2469
- video: remove unused qent->dst by @sreimers in baresip#2474
- call: Fix delayed (auto) answer if awaiting PRACK by @maximilianfridrich in baresip#2473
- video: add TX thread by @sreimers in baresip#2460
- ccheck: add check_list_unlink check by @sreimers in baresip#2471
- stream: add stream_enable_tx() api by @sreimers in baresip#2479
- audio: align Audio TX thread name by @sreimers in baresip#2480
- Send event when dump file is opened by @juha-h in baresip#2486
- video: add NULL pointer check for vidisp by @cspiel1 in baresip#2483
- ua: Fix calls of ua_event() by @maximilianfridrich in baresip#2495
- call: Fix calls of call_event_handler by @maximilianfridrich in baresip#2496
- @Koshub made their first contribution in baresip#2452
- @tomek-o made their first contribution in baresip#2467
Full Changelog: https://github.com/baresip/baresip/compare/v2.12.0...v3.0.0
- call: default status code for rejecting incoming calls by @cspiel1 in baresip#2409
- dtls_srtp: enable single DTLS connection mode by @alfredh in baresip#2411
- ci: try to fix flaky azure mirrors by @sreimers in baresip#2413
- cmake/pulse: Remove pulse-simple library lookup by @robert-scheck in baresip#2414
- webrtc_aecm: use C11 mutex by @juha-h in baresip#2415
- pulse: replace obsolete string pulse_async (makes baresip PipeWire compatible) by @cspiel1 in baresip#2412
- vidpacket: add keyframe flag by @alfredh in baresip#2416
- av1: use keyframe instead of new-flag by @alfredh in baresip#2418
- av1: fix warnings by @alfredh in baresip#2419
- make rtcp interval configureable by @sreimers in baresip#2420
- sndio: remove deprecated module by @alfredh in baresip#2422
- PRACK refactoring by @maximilianfridrich in baresip#2401
- ci: merge build and cmake by @alfredh in baresip#2425
- menu: ringback/early audio handling for parallel calls by @cspiel1 in baresip#2403
- magic: use C99 func macro by @alfredh in baresip#2427
- stream: remove stream_decode from internal API by @cspiel1 in baresip#2430
- use RE_ARRAY_SIZE() macro by @alfredh in baresip#2429
- cmake: link RESOLV_LIBRARY by @sreimers in baresip#2432
- ci/build: fix Ubuntu 22.04 workaround by @sreimers in baresip#2435
- avcapture: use RE_ARRAY_SIZE macro by @alfredh in baresip#2434
- pulse: remove obsolete doxygen note to be experimental by @cspiel1 in baresip#2436
- gtk: return NULL on mtx_init() != thrd_success by @robert-scheck in baresip#2440
- ci: add libgtk-3-dev to build GTK+ 3 module by @robert-scheck in baresip#2441
- event: missing class name case for RTPESTAB event by @cspiel1 in baresip#2447
- ci: add sanitizers by @sreimers in baresip#2449
- bump version numbers to 2.12.0 by @alfredh in baresip#2453
Full Changelog: https://github.com/baresip/baresip/compare/v2.11.0...v2.12.0
- uag,call: do not override status code and reason by @cspiel1 in baresip#2345
- stream: send RTP NAT pinhole opener until RTP is received by @cspiel1 in baresip#2346
- mediatrack: add audio and video getters by @sreimers in baresip#2347
- Added rtcp_mux related API functions by @juha-h in baresip#2352
- make: remove deprecated makefile by @alfredh in baresip#2354
- Removed rtcp_mux config variable by @juha-h in baresip#2353
- Use bool instead of "yes"/"no" in account API functions by @juha-h in baresip#2355
- aubuf: add AUBUF_FILE mode by @cspiel1 in baresip#2363
- play: flush aubuf before restart by @cspiel1 in baresip#2364
- call: avoid unwanted re-invites on ESTABLISHED event by @cspiel1 in baresip#2362
- avcodec: constrain bitrate by @sreimers in baresip#2365
- pulse: rename to pulse_simple.so by @alfredh in baresip#2371
- module: remove module_tmp by @alfredh in baresip#2373
- audio: remove unused last_sampc by @alfredh in baresip#2372
- audio: add rtpext_find() (refactoring) by @alfredh in baresip#2375
- multicast: remove ref to pthread by @alfredh in baresip#2379
- video: add rtcp-fb Generic NACK by @sreimers in baresip#2380
- call: set media dir also for MNAT case by @cspiel1 in baresip#2382
- pulse: rename pulse_async.so to pulse.so (default) by @alfredh in baresip#2381
- RTP Resend by @sreimers in baresip#2378
- make: remove unused srcs.mk by @alfredh in baresip#2387
- TLS server support SNI based certificate selection by @cspiel1 in baresip#2330
- audiounit: use C11 mutex by @alfredh in baresip#2386
- webrtc_aec: use C11 mutex by @alfredh in baresip#2384
- coreaudio: use C11 mutex by @alfredh in baresip#2388
- gtk: use C11 threads by @alfredh in baresip#2391
- remove pulse_simple.so -- use pulse.so by @alfredh in baresip#2383
- ci: rename ccheck.yml to lint.yml by @alfredh in baresip#2394
- fritzbox2baresip: use open with explicitly specifying an encoding by @robert-scheck in baresip#2396
- test: remove mock_aufilt (unused) by @alfredh in baresip#2392
- Ci pylint by @alfredh in baresip#2395
- gzrtp: use C11 mutex by @alfredh in baresip#2393
- C11 mutex by @mbattista in baresip#2397
- tls multiple server certs by @cspiel1 in baresip#2399
- call: return EINVAL if answer not possible by @maximilianfridrich in baresip#2405
- ccheck: fix some pylint warnings by @alfredh in baresip#2398
- Fixed account debug of mwi and call_transfer by @juha-h in baresip#2406
- avformat: fix printf format for samplerate and channels by @alfredh in baresip#2407
- cmake: increase minimum required version by @cspiel1 in baresip#2408
Full Changelog: https://github.com/baresip/baresip/compare/v2.10.0...v2.11.0
- sdl: small improvements by @sreimers in baresip#2285
- vidinfo: allow all pixel formats by @alfredh in baresip#2291
- vid: add support for YUV422P pixel format by @alfredh in baresip#2280
- avformat: fix hwaccel vaapi by @alfredh in baresip#2299
- mk: add deprecate notice by @alfredh in baresip#2302
- mingw: upgrade to OpenSSL 3.0.7 by @alfredh in baresip#2303
- dshow: fix some warnings by @alfredh in baresip#2305
- dshow: fix pragma warning by @alfredh in baresip#2306
- ci: install libsdl2 development package by @alfredh in baresip#2307
- sdl: work in progress fixes for multi-threading by @alfredh in baresip#2300
- Stop segfaulting when no URI is passed to dial command by @SimonHyde-BBC in baresip#2311
- ice: local candidate policy config by @sreimers in baresip#2312
- auresamp: check handler arguments by @alfredh in baresip#2313
- fixes 2315 and GTK errors on quit by @mbattista in baresip#2316
- auresamp: avoid division by zero (#2293) by @cspiel1 in baresip#2317
- cmake: check for XShm.h (#2318) by @cspiel1 in baresip#2319
- pulse_async: avoid integer overrun for timestamps in recorder by @cspiel1 in baresip#2321
- ua: use sdp connection data instead origin by @sreimers in baresip#2298
- rtpext: move from baresip to re by @alfredh in baresip#2322
- acc,stream: add rtcp_mux account param by @sreimers in baresip#2320
- video: video_update cleanup by @sreimers in baresip#2324
- aufile/src: add auframe support by @sreimers in baresip#2325
- ice/tmr_async_handler: fix possible segfault by @sreimers in baresip#2326
- webrtc: fix browser offer handling by @sreimers in baresip#2327
- Space at the beginning of sip: creates errors by @mbattista in baresip#2329
- opus_multistream: update mimetype to ad-hoc standard by @alfredh in baresip#2328
- webrtc: add offerer and recvonly options by @sreimers in baresip#2331
- test: replace RSA cert with EC cert by @alfredh in baresip#2332
- Add OPTIONS handling for webrtc demo by @RenSym in baresip#2333
- mk: remove rtpext.c from srcs.mk by @cspiel1 in baresip#2336
- ua: change refer log to info() by @alfredh in baresip#2338
- @SimonHyde-BBC made their first contribution in baresip#2311
- @RenSym made their first contribution in baresip#2333
Full Changelog: https://github.com/baresip/baresip/compare/v2.9.0...v2.10.0
- sndfile Module - filename includes strm->cname (i.e. call->local_uri)~ by @ninp0 in baresip#2165
- log: optional timestamps by @cspiel1 in baresip#2169
- avcodec: remove H263 codec by @alfredh in baresip#2182
- mk: bump PROJECT_NUMBER in Doxyfile by @cspiel1 in baresip#2201
- stream: correct Doxygen for peer field by @cspiel1 in baresip#2202
- cmake: add pre version handling by @sreimers in baresip#2203
- cmake,debian: use dh-cmake by @sreimers in baresip#2204
- cmake: add pkgconfig by @robert-scheck in baresip#2205
- Avoid webrtc_aecm module C++20 extension warnings by @juha-h in baresip#2215
- cmake/ctrld_dbus: ninja and subdirectory fixes by @sreimers in baresip#2221
- cmake: link CMAKE_CURRENT_BINARY_DIR modules by @sreimers in baresip#2223
- cmake,debian: fix libbaresip dependency by @sreimers in baresip#2224
- cmake: set C only flags by @sreimers in baresip#2226
- FindPNG needs to find also include directory by @juha-h in baresip#2230
- FindVPX needs to find also include directory by @juha-h in baresip#2231
- Multicast send events on mcreg enable commands by @cHuberCoffee in baresip#2219
- call, menu: support display name for outgoing calls by @cspiel1 in baresip#2220
- call: hangup call on transp reset if necessary by @maximilianfridrich in baresip#2229
- portaudio: add mediadev_add with mediadev driver fields by @sreimers in baresip#2233
- call: fix mnat call_streams_alloc by @sreimers in baresip#2242
- jack: fix CodeQL uninitialized local variable by @sreimers in baresip#2244
- Avoid snapshot compiler warnings by @juha-h in baresip#2239
- avformat: remove old call to avcodec_register_all() by @alfredh in baresip#2246
- avformat: remove LIBAVUTIL_VERSION_MAJOR check by @alfredh in baresip#2247
- ua: wording for warning in ua_refer_send() by @cspiel1 in baresip#2249
- ua: use mbuf functions for ua_connect_dir by @cspiel1 in baresip#2250
- ci: use actions/checkout@v3 by @sreimers in baresip#2254
- avcodec: remove av_packet_free() wrapper by @alfredh in baresip#2255
- selfview: create window in encode_update by @alfredh in baresip#2253
- alsa: use C11 threads by @alfredh in baresip#2256
- config: fix template for avcodec_xxx by @alfredh in baresip#2258
- avformat: use C11 threads by @alfredh in baresip#2259
- v4l2: use C11 threads by @alfredh in baresip#2261
- avcodec: remove LIBAVUTIL_VERSION_MAJOR check by @alfredh in baresip#2260
- multicast: use C11 threads by @alfredh in baresip#2262
- menu fix display name by @cspiel1 in baresip#2251
- account: do not complete dial URI if scheme is included by @cspiel1 in baresip#2267
- menu: simplify URI complete by @cspiel1 in baresip#2268
- gtk: use new function account_uri_complete_strdup() by @cspiel1 in baresip#2273
- Removed module avformat dependency on libpostproc by @juha-h in baresip#2274
- make: detect and add swscale module in modules.mk by @agorgl in baresip#2281
- cmake: add APP_MODULES symlinks by @sreimers in baresip#2286
- cmake: use CMAKE_SHARED_MODULE_SUFFIX by @sreimers in baresip#2292
- @ninp0 made their first contribution in baresip#2165
- @agorgl made their first contribution in baresip#2281
Full Changelog: https://github.com/baresip/baresip/compare/v2.8.1...v2.9.0
- baresip.h: bump BARESIP_VERSION by @cspiel1 in baresip#2196
- opensles cmake by @juha-h in baresip#2108
- test/call: Add test_call_change_videodir by @maximilianfridrich in baresip#2080
- cmake: bump min version to 3.10 by @alfredh in baresip#2112
- zrtp: remove module, use gzrtp instead by @alfredh in baresip#2109
- Avoid gzrtp compile warnings by @juha-h in baresip#2110
- Update video in menu when UA_EVENT_CALL_REMOTE_SDP is recieved by @juha-h in baresip#2113
- http/https requests with large body by @fAuernigg in baresip#2100
- call: send reinvite after established handlers by @maximilianfridrich in baresip#2117
- refer out of dialog by @cspiel1 in baresip#2115
- remove unused functions in baresip.h by @cspiel1 in baresip#2122
- webrtc/demo: make https optional by @sreimers in baresip#2120
- Restored original working behavior in uag request_handler by @juha-h in baresip#2124
- uag: out-of-dialog REFER handler checks to.tag by @cspiel1 in baresip#2125
- Update media fixes by @cspiel1 in baresip#2116
- account: set 100rel default to no by @cspiel1 in baresip#2128
- avcodec: remove usage of old FFmpeg api (before 4.1.9) by @alfredh in baresip#2126
- rtp: Improve media synchronization by @sreimers in baresip#2129
- avformat: remove usage of old FFmpeg api by @alfredh in baresip#2130
- i2s: remove deprecated module by @alfredh in baresip#2131
- ci: migrate to CMake by @alfredh in baresip#2132
- menu: during early media switch on/off ringback by @cspiel1 in baresip#2133
- call, event, audio: send DTMF via hidden call by @cspiel1 in baresip#2134
- ua,reg,serreg: fix serial registration mode by @cspiel1 in baresip#2139
- CodeQL fixes by @sreimers in baresip#2143
- cmake: set atomic-implicit-seq-cst only for C language by @sreimers in baresip#2145
- cmake: define -Wshorten-64-to-32 C only by @sreimers in baresip#2146
- Cmake of webrtc_aec module plus remove of unused aec.cpp var by @juha-h in baresip#2144
- cmake: make include dir public by @sreimers in baresip#2147
- cmake: add APP_MODULES and APP_MODULES_DIR by @sreimers in baresip#2148
- Added cmake of gst module by @juha-h in baresip#2149
- Improved call closed message by @juha-h in baresip#2151
- gtk & menu: Fix potential memory leaks by @maximilianfridrich in baresip#2153
- call: allocate streams after peer_uri was set by @cspiel1 in baresip#2154
- dshow/cmake: fix stdc++ linking with MSVC by @sreimers in baresip#2156
- cmake: fix MSVC library output name by @sreimers in baresip#2157
- webrtc: add HAVE_GETOPT by @sreimers in baresip#2158
- config: ignore dirent.h on win32 by @sreimers in baresip#2159
- ua: do not duplicate request URI parameters by @maximilianfridrich in baresip#2152
- cmake: add netroam module by @robert-scheck in baresip#2170
- cmake: add portaudio module by @robert-scheck in baresip#2173
- cmake: add jack module by @robert-scheck in baresip#2172
- avcodec,config: add setting for keyframe interval by @cspiel1 in baresip#2171
- cmake: add sdl module by @robert-scheck in baresip#2174
- call: set peer URI early for incoming calls by @cspiel1 in baresip#2168
- cmake: Add options -DDEFAULT_CAFILE="…" and -DDEFAULT_AUDIO_DEVICE="…" by @robert-scheck in baresip#2179
- cmake: add gtk module by @robert-scheck in baresip#2176
- cmake: add opus_multistream module by @robert-scheck in baresip#2175
- cmake: synchronize behaviour of -DSHARE_PATH="…" with GNU Makefiles by @robert-scheck in baresip#2180
- cmake: synchronize behaviour of -DMOD_PATH="…" with GNU Makefiles by @robert-scheck in baresip#2181
- Move docs/COPYING to LICENSE and update content to match with re/rem by @robert-scheck in baresip#2188
- cmake: add ABI (soname) versioning by @robert-scheck in baresip#2187
- misc: Use example domains and IPs by @robert-scheck in baresip#2186
- cmake: symlink modules by @sreimers in baresip#2190
- cmake: add mpa module by @robert-scheck in baresip#2191
2.7.0 - 2022-09-01
- menu: fix menu_ua_carg data preference by @sreimers in baresip#2045
- call: remember media dir for established state by @cspiel1 in baresip#2046
- avformat: fix ffmpeg channel_layout deprecation by @sreimers in baresip#2048
- cmake: add multicast module by @cHuberCoffee in baresip#2049
- play: ring tone fixes if file_ausrc is set by @cspiel1 in baresip#2050
- add peerconnection and mediatrack by @alfredh in baresip#2054
- main,test: close re async before tmr_debug by @sreimers in baresip#2055
- http: new file for HTTP functions by @alfredh in baresip#2056
- http: add http_reply_json() by @alfredh in baresip#2057
- play: tmr_polling has to check if ausrc is used by @cspiel1 in baresip#2061
- cmake: use object instead of static for modules by @sreimers in baresip#2064
- [WIP] import baresip-webrtc by @alfredh in baresip#2059
- FindAMR.cmake fixes/improvements by @juha-h in baresip#2066
- cmake: fix modules install path and install share files by @sreimers in baresip#2068
- hook up webrtc to main cmake file by @alfredh in baresip#2067
- avformat: check shared for both audio+video by @alfredh in baresip#2069
- cmake: add V4L2 module by @alfredh in baresip#2071
- Omx remove by @alfredh in baresip#2070
- cmake: add directfb module by @alfredh in baresip#2072
- main,webrtc/main: add re_thread_async_init by @sreimers in baresip#2076
- cmake: add wincons and winwave modules by @alfredh in baresip#2077
- cmake: add sndfile module by @alfredh in baresip#2078
- Mention actual GTK+ 3 usage (instead of 2) in README.md by @robert-scheck in baresip#2079
- ctrl_tcp: change unsafe operations on an mbuf to the safe mbuf interface by @cHuberCoffee in baresip#2082
- gzrtp: Call event hander when SAS needs to be verified by @juha-h in baresip#2081
- Generate also MENC_EVENT_PEER_VERIFIED event by @juha-h in baresip#2084
- gzrtp: Generate only one MENC_EVENT_PEER_VERIFIED event when all streams are verified by @juha-h in baresip#2086
- config,net: add use_getaddrinfo/dns_getaddrinfo option by @sreimers in baresip#2087
- cmake: add_compile_options and use re config by @sreimers in baresip#2089
- cmake/modules: build syslog only if available by @sreimers in baresip#2090
- cmake: add selftest by @sreimers in baresip#2093
- cmake: add win32 linklibs by @sreimers in baresip#2091
- cmake: add mqtt by @sreimers in baresip#2094
- Improve C11 cchecks by @sreimers in baresip#2096
- Added cmake of gzrtp module by @juha-h in baresip#2095
- Gzrtp cmake by @juha-h in baresip#2102
- Added cmake of webrtc_aecm module by @juha-h in baresip#2104
- Suppressed unused var warnings in webrtc_aecm module by @juha-h in baresip#2105
- call: do not set call state to answered, if session progress (PRACK) by @RobertMi21 in baresip#2106
2.6.0 - 2022-08-01
- conf: check input arguments by @alfredh in baresip#1932
- dtls_srtp: print TLS cipher name by @alfredh in baresip#1933
- cmake: add AAC module by @alfredh in baresip#1935
- call, menu: make selective early media RFC-3261 conform by @cspiel1 in baresip#1929
- config: add flag to enable/disable linklocal by @alfredh in baresip#1934
- audio: update filters if codec changes by @cspiel1 in baresip#1937
- Fix CMake fails when OpenSSL is not present by @widgetii in baresip#1939
- sip: add RFC 3262 support by @maximilianfridrich in baresip#1930
- Add CMake target to install baresip executable, library and modules by @widgetii in baresip#1940
- audio: fix SEGV if stream_alloc() fails by @cspiel1 in baresip#1942
- gst_video: remove deprecated module by @alfredh in baresip#1943
- ci: test cmake by @alfredh in baresip#1944
- cmake: add aptx module by @alfredh in baresip#1945
- avcodec: remove avcodec_free_context wrapper by @alfredh in baresip#1947
- avcodec: remove old call to avcodec_init() by @alfredh in baresip#1948
- cmake: add ffmpeg modules by @alfredh in baresip#1949
- cmake: add codec2 module by @alfredh in baresip#1950
- thread: thrd_error fixes by @sreimers in baresip#1955
- Revert PR #1922 by @juha-h in baresip#1964
- sip: add RFC 3311 support by @maximilianfridrich in baresip#1941
- cmake: add amr module by @alfredh in baresip#1962
- ci/misc: bump [email protected] by @sreimers in baresip#1968
- ci: add cmake/macos by @alfredh in baresip#1961
- Feature: add user data to call by @copiltembel in baresip#1951
- cmake: add audiounit module by @alfredh in baresip#1969
- cmake: add avcapture module by @alfredh in baresip#1970
- cmake: add coreaudio module by @alfredh in baresip#1972
- audio: remove unused aubuf for decoding by @cspiel1 in baresip#1974
- Modules cmake by @viordash in baresip#1975
- Modules cmake by @viordash in baresip#1978
- audio: always start reading in TX thread by @cspiel1 in baresip#1979
- audio: always start reading in TX poll mode by @cspiel1 in baresip#1980
- multicast: always start reading of TX aubuf by @cspiel1 in baresip#1981
- pulse_async: reduce number of reconnect attempts by @RobertMi21 in baresip#1977
- ci/build: replace deprecated macos-10.15 by @sreimers in baresip#1984
- ci/build/macos: disable dbus by @sreimers in baresip#1985
- Improve RFC 3262 by @maximilianfridrich in baresip#1973
- call: do not stop streams on session progress by @cspiel1 in baresip#1986
- audio: revert some TX commits and fix TX poll mode by @cspiel1 in baresip#1987
- call: fix heap-buffer-overflow in prack_handler by @sreimers in baresip#1988
- Improve re_atomic handling by @sreimers in baresip#1982
- mk/ctrl_dbus: fix atomic implicit warnings by @sreimers in baresip#1991
- cmake: add mixminus module by @sreimers in baresip#1994
- cmake: add dtls_srtp module by @alfredh in baresip#1993
- Stun uri cred by @viordash in baresip#1960
- event: fix wrong place of the err check by @copiltembel in baresip#1992
- Added mwi module cmake build by @juha-h in baresip#1995
- call: disable prack_handler temporarily by @sreimers in baresip#1998
- Fix prack handling by @maximilianfridrich in baresip#1999
- ci: test re/rem with cmake by @alfredh in baresip#1997
- Added cmake of zrtp module by @juha-h in baresip#2005
- Added cmake of zrtp module by @juha-h in baresip#2006
- Added cmake of uuid module by @juha-h in baresip#2007
- cmake: fix openssl linking by @sreimers in baresip#2008
- Load also pulse-simple library if exists by @juha-h in baresip#2010
- Added cmake of presence module by @juha-h in baresip#2011
- cmake: add more libs, stable branch and static build by @sreimers in baresip#2012
- Added cmake of selfview module by @juha-h in baresip#2014
- Added cmake of vp8 and vp9 modules by @juha-h in baresip#2016
- Added cmake of g722 module by @juha-h in baresip#2015
- Added cmake of srtp module by @juha-h in baresip#2017
- cmake: add module override option by @sreimers in baresip#2020
- cmake: add EXTRA_MODULES option by @sreimers in baresip#2021
- ci/cmake: add brew packages by @sreimers in baresip#2023
- Added cmake of g726 module by @juha-h in baresip#2022
- cmake: add ctrl_dbus module by @cspiel1 in baresip#2000
- cmake: refactor module prefix by @sreimers in baresip#2024
- Added cmake of snapshot module by @juha-h in baresip#2026
- Cmake add dshow by @alfredh in baresip#2031
- fakevideo: use C11 threads by @alfredh in baresip#2032
- cmake: add evdev module by @alfredh in baresip#2033
- aubridge: use C11 threads by @alfredh in baresip#2035
- ausine: use C11 threads by @alfredh in baresip#2036
- cmake: check for HAVE_UNISTD_H by @alfredh in baresip#2039
- cmake,mk: prepare main version for release by @sreimers in baresip#2040
- gsm: remove deprecated module by @alfredh in baresip#2034
- cmake: add g7221 module by @alfredh in baresip#2041
- @widgetii made their first contribution in baresip#1939
- @maximilianfridrich made their first contribution in baresip#1930
- @copiltembel made their first contribution in baresip#1951
2.5.0 - 2022-07-01
- audio: add optional decoding buffer by @cspiel1 in baresip#1842
- audio: RX filter thread needs separate sampv buffer by @cspiel1 in baresip#1879
- aufile: fix possible data race warning by @cspiel1 in baresip#1880
- audiounit,coreaudio: fix kAudioObjectPropertyElementMaster deprecation by @sreimers in baresip#1881
- av1: explicitly check for supported OBU types by @alfredh in baresip#1882
- audiounit/coreaudio: fix kAudioObjectPropertyElementMain by @sreimers in baresip#1885
- ci/build: bump macos min. sdk to 10.12 by @sreimers in baresip#1883
- ci: run only for pull requests and main branch by @sreimers in baresip#1887
- multicast: C11 mutex by @alfredh in baresip#1892
- dtls_srtp: enable ECC by default, remove RSA by @alfredh in baresip#1891
- ci/build: add ubuntu 22.04 by @sreimers in baresip#1890
- test: add check for memory leaks by @sreimers in baresip#1896
- stream,metric: RX real-time - make metric thread-safe by @cspiel1 in baresip#1895
- Cmake findre by @alfredh in baresip#1893
- test: wait for both audio and video to be established by @alfredh in baresip#1903
- docs: remove old TODO file by @alfredh in baresip#1902
- audio: fixed check for aubuf started flag by @cspiel1 in baresip#1904
- use new mutex interface by @cspiel1 in baresip#1905
- audio: make rx.filtl thread-safe by @cspiel1 in baresip#1897
- audio: allocate correct buffer size for static auplay srate by @cspiel1 in baresip#1906
- Pulseaudio Async Interface Module by @cHuberCoffee in baresip#1907
- Do not destroy register client when it is unregistered by @juha-h in baresip#1908
- Two spaces are required after email address by @juha-h in baresip#1909
- cmake: add alsa module by @alfredh in baresip#1910
- cmake: fix static openssl and thread linking by @sreimers in baresip#1911
- In start_registering, create register clients if reg list is empty by @juha-h in baresip#1913
- ctrl_dbus: use new thread and mtx interface by @cspiel1 in baresip#1916
- cmake: add pulse and pulse_async module by @cHuberCoffee in baresip#1919
- Un-subscribe mwi at un-register by @juha-h in baresip#1918
- call: update media on session progress. by @RobertMi21 in baresip#1922
- ctrl_dbus send event in main thread by @cspiel1 in baresip#1921
- uag: add timestamps to SIP trace by @cspiel1 in baresip#1914
- main: fix open timers check by @sreimers in baresip#1925
- cmake: add account module by @alfredh in baresip#1926
2.4.0 - 2022-06-01
- mulitcast unmute bad quality by @cspiel1 in baresip#1821
- menu ringback for parallel call by @cspiel1 in baresip#1827
- multicast: support error code EAGAIN of jbuf_get() by @cspiel1 in baresip#1832
- use RTP clock rate for timestamp calculation by @cspiel1 in baresip#1834
- av1 obu by @alfredh in baresip#1835
- av1 packetizer by @alfredh in baresip#1836
- av1: depacketizer by @alfredh in baresip#1837
- Disabled debug statement by @juha-h in baresip#1838
- h264: move from rem to re by @sreimers in baresip#1839
- ua: send new event UA_EVENT_CREATE at successful ua allocation by @cHuberCoffee in baresip#1840
- evdev: fix wrong ioctl size by @sreimers in baresip#1843
- aufile: ausrc_prm has to be copied when source is allocated by @cspiel1 in baresip#1844
- conf: missing pointer initialization found by clang analyzer by @cspiel1 in baresip#1845
- mk/modules: fix omx RPI detection by @sreimers in baresip#1847
- auconv: add auconv_to_float (fixes #1833) by @alfredh in baresip#1849
- avfilter: migrate to C11 mutex by @alfredh in baresip#1850
- avformat: C11 mutex by @alfredh in baresip#1851
- selfview: C11 mutex by @alfredh in baresip#1852
- audio: C11 mutex by @alfredh in baresip#1853
- metric: C11 mutex by @alfredh in baresip#1854
- play: C11 mutex by @alfredh in baresip#1855
- dns: add query cache by @sreimers in baresip#1848
- video: C11 mutex by @alfredh in baresip#1856
- aufile: C11 threads by @alfredh in baresip#1858
- audio: add more locking by @alfredh in baresip#1857
- aufile/play: fix run data race by @sreimers in baresip#1859
- mc: multicast receiver enable state fix by @cHuberCoffee in baresip#1861
- audio: C11 thread by @alfredh in baresip#1860
- av1: add packetize handler by @alfredh in baresip#1865
- net/net_debug: add default route hint by @sreimers in baresip#1864
- ice: fix local prio calculation by @sreimers in baresip#1863
- avformat: open codec if not passthrough by @alfredh in baresip#1866
- dtls_srtp: Minor whitespace fix by @robert-scheck in baresip#1870
- vp8: add packetize handler by @alfredh in baresip#1868
- vp9: add packetizer by @alfredh in baresip#1871
- debug_cmd: support absolute path for command aufileinfo by @cspiel1 in baresip#1875
- event: add diverter URI to UA event by @cspiel1 in baresip#1876
- aufileinfo with synchronous response by @cspiel1 in baresip#1877
2.3.0 - 2022-05-01
- mc: multicast mute function by @cHuberCoffee in baresip#1805
- mc: reject incoming call if high prio multicast is received by @cHuberCoffee in baresip#1804
- mc: mcplayer stream fade-out and fade-in by @cHuberCoffee in baresip#1802
- clean_number now will remove all non-digit chars by @mbattista in baresip#1806
- Workflows cmakelint by @alfredh in baresip#1808
- ccheck: check all CMakeLists.txt files by @sreimers in baresip#1810
- mk: remove win32 MSVC project files by @alfredh in baresip#1811
- cmake: add modules by @sreimers in baresip#1812
- ajb,aubuf: timestamp is given in [us] by @cspiel1 in baresip#1809
- call: allow optional leading space in SIP INFO for dtmf-relay by @thomas-karl in baresip#1814
- conf: add fs_file_extension() by @alfredh in baresip#1816
- Updated debian version by @juha-h in baresip#1817
- pulse: fix timestamp integer overrun for arm by @cspiel1 in baresip#1818
- fix audio multicast artefacts by @cspiel1 in baresip#1819
- audio: flush aubuf if ssrc changes by @cspiel1 in baresip#1822
- Debian control dependency update by @juha-h in baresip#1823
- pulse: support restart of pulseaudio during stream by @cspiel1 in baresip#1824
- version 2.3.0 by @alfredh in baresip#1826
- @thomas-karl made their first contribution in baresip#1814
2.0.2 - 2022-04-09
- Added API function call_diverteruri by @juha-h in baresip#1780
- Avoid undeclared 'CLOCK_REALTIME' on RHEL/CentOS 7 (fixes #1781) by @robert-scheck in baresip#1782
- audio: add lock in audio_send_digit by @GGGO in baresip#1786
- vumeter: use new auframe_level() by @sreimers in baresip#1788
- reg.c: use already declared acc by @GGGO in baresip#1789
- aubuf adaptive jitter buffer by @cspiel1 in baresip#1784
- multicast set aubuf silence by @cspiel1 in baresip#1791
- ccheck: fix line number in error print by @cspiel1 in baresip#1793
- test: check the correct stream in UA_EVENT_CALL_MENC by @alfredh in baresip#1794
- audio: missing lock around stream_send by @GGGO in baresip#1796
- docs: remove obsolete jitter_buffer_wish from config example by @cspiel1 in baresip#1798
- Multicast jbuf and aubuf changes by @cHuberCoffee in baresip#1797
- uag: uag_hold_resume() should not return err if there is no call to hold by @cspiel1 in baresip#1799
- stream: remove mbuf_get_left check in rtp_handler by @GGGO in baresip#1801
- cmake: preliminary support by @alfredh in baresip#1800
- @GGGO made their first contribution in baresip#1786
2.0.1 - 2022-03-27
- audio: fix rx_thread (adaptive jitter buffer) by @sreimers in baresip#1769
- test: init fixture by @alfredh in baresip#1772
- test: refactoring of test_account_uri_complete by @alfredh in baresip#1773
- mk: check also if extensions/XShm.h is present by @cspiel1 in baresip#1774
- menu: support custom SIP headers by @cspiel1 in baresip#1775
- menu: use new sdp_dir_decode by @cspiel1 in baresip#1776
- menu: avoid multiple hash entries with same key by @cspiel1 in baresip#1777
- menu: support audio file config value "none" by @cspiel1 in baresip#1778
- intercom: add video preview call by @cspiel1 in baresip#1779
2.0.0 - 2022-03-11
- debug_cmd: use module_event() for aufileinfo events by @cspiel1 in baresip#1345
- multicast: use module_event() for sending events by @cspiel1 in baresip#1346
- ctrl_dbus: use module_event() to send exported event by @cspiel1 in baresip#1347
- ua,call: add CALL_EVENT_OUTGOING by @cspiel1 in baresip#1348
- GTK caller history by @mbattista in baresip#1350
- Convert FRITZ!Box XML phone book into Baresip contacts by @robert-scheck in baresip#1382
- menu: play ringtone on audio_alert device by @cspiel1 in baresip#1396
- menu: use str_isset() for command parameter by @cspiel1 in baresip#1397
- dtls_srtp: use elliptic curve cryptography by @cHuberCoffee in baresip#1385
- Support for s16 playback in jack. Needed for play tones by @srperens in baresip#1399
- Check that account ;sipnat param has valid value by @juha-h in baresip#1401
- Tls sipcert per acc by @cHuberCoffee in baresip#1376
- Vidsrc add packet handler by @alfredh in baresip#1402
- ToS for video and sip by @cspiel1 in baresip#1393
- account: add accounts parameter to force media address family by @cspiel1 in baresip#1395
- Selective early media by @cspiel1 in baresip#1398
- ua,uag: split ua.c and uag.c by @cspiel1 in baresip#1349
- Account media af template by @cspiel1 in baresip#1406
- account: add missing client certificate parameter to template by @cHuberCoffee in baresip#1408
- account: update answermode values in template by @cspiel1 in baresip#1405
- menu: command uafind raises UA to head by @cspiel1 in baresip#1407
- ctrl_dbus: fix possible memleak on failed initialization by @cspiel1 in baresip#1410
- video passthrough by @alfredh in baresip#1418
- menu: enable auto answer calls also for command dialdir by @cspiel1 in baresip#1412
- menu: add command for settings media local direction by @cspiel1 in baresip#1413
- Accounts addr params by @cspiel1 in baresip#1414
- Accounts example cleanup by @cspiel1 in baresip#1415
- menu,call: fix hangup for outgoing call by @cspiel1 in baresip#1417
- multicast: add source and player API calls by @cHuberCoffee in baresip#1403
- menu: add command /uareg by @alfredh in baresip#1421
- menu: return complete URI for commands dial,dialdir by @cspiel1 in baresip#1424
- menu: in command dialdir call uag_find_requri() with uri by @cspiel1 in baresip#1425
- gst: replace variable length array (buf) with mem_zalloc by @sreimers in baresip#1426
- menu: avoid possible memleaks for dial/dialdir commands by @cspiel1 in baresip#1430
- uag: use local cuser for selecting user-agent (#1433) by @cspiel1 in baresip#1434
- Work on Intercom module by @cspiel1 in baresip#1432
- Attended Transfer on GTK by @mbattista in baresip#1435
- Update README.md with configuration suggestion by @webstean in baresip#1438
- README fixes by @juha-h in baresip#1440
- Accounts examples and template by @cspiel1 in baresip#1441
- serreg: use a timer for registration restart by @cspiel1 in baresip#1445
- gst: audio playback not correct for some WAV files. by @RobertMi21 in baresip#1442
- Working on intercom (ringtone override) by @cspiel1 in baresip#1436
- Use line number 0 if user did not provide any line number by @negbie in baresip#1451
- AMR Bandwidth Efficient mode support by @srperens in baresip#1423
- Working on Intercom (menu: allow other modules to reject a call) by @cspiel1 in baresip#1437
- auframe: add samplerate and channels by @sreimers in baresip#1452
- account: comment out very basic example in template by @cspiel1 in baresip#1458
- call answer media dir by @cspiel1 in baresip#1449
- Account auto answer beep by @cspiel1 in baresip#1461
- serreg: unregister correct User-Agents on registration failure by @cspiel1 in baresip#1462
- mk: enable auto-detect of av1 module by @alfredh in baresip#1463
- ctrl dbus makefile depends by @cspiel1 in baresip#1457
- stream: check if media is present before enabling the RTP timeout by @cspiel1 in baresip#1465
- ctrl_dbus: generate dbus code and documentation in makefile by @cspiel1 in baresip#1456
- auframe: always set srate and ch by @janh in baresip#1468
- auto answer beep per alert info URI by @cspiel1 in baresip#1466
- auframe: move to rem by @sreimers in baresip#1470
- mixminus: add conference feature by @sreimers in baresip#1411
- vidbridge: check vidbridge_disp_display args fixes segfault by @sreimers in baresip#1471
- gst: fixed some memory leaks by @RobertMi21 in baresip#1476
- ua, menu: move auto answer delay handling to menu (#1474) by @cspiel1 in baresip#1475
- ua,menu: move handling of ANSWERMODE_AUTO to menu (#1474) by @cspiel1 in baresip#1478
- ausine: support for multiple samplerates by @alfredh in baresip#1479
- account: fix IPv6 only URI for account_uri_complete() by @cspiel1 in baresip#1472
- ilbc: remove deprecated module by @alfredh in baresip#1483
- aubridge/device: remove unused sampv_out (old resample code) by @sreimers in baresip#1484
- pkg-config version check by @sreimers in baresip#1481
- mk: support more locations for libre.pc and librem.pc by @cspiel1 in baresip#1486
- net: remove unused domain by @alfredh in baresip#1489
- audio: fix aufilt_setup update handling by @sreimers in baresip#1498
- SIP redirect callbackfunction by @cHuberCoffee in baresip#1495
- add secure websocket tls context by @sreimers in baresip#1499
- test: add stunuri by @alfredh in baresip#1503
- turn: refactoring, add compv by @alfredh in baresip#1505
- fmt: add string to bool function by @cspiel1 in baresip#1501
- mk: check glib-2.0 at least like in ubuntu 18.04 by @cspiel1 in baresip#1507
- registration fixes by @cspiel1 in baresip#1510
- uag,menu: add commands to enable/disable UDP/TCP/TLS by @cspiel1 in baresip#1502
- config,audio: add setting audio.telev_pt by @cspiel1 in baresip#1509
- stream: fix telephone event (#1494) by @cspiel1 in baresip#1506
- Fix I2S compile error, use auframe by @andreaswatch in baresip#1512
- ci/tools: fix pylint by @sreimers in baresip#1515
- config: not all audio config was printed by @cspiel1 in baresip#1516
- net: replace network_if_getname with net_if_getname by @sreimers in baresip#1518
- account: add setting audio payload type for telephone-event by @cspiel1 in baresip#1517
- uag,menu: simplify transport enable/disable and support also ws/wss by @cspiel1 in baresip#1514
- rst: remove deprecated module by @alfredh in baresip#1519
- turn: add TCP and TLS transports by @alfredh in baresip#1520
- speex_pp: remove deprecated module by @alfredh in baresip#1521
- call: allow video calls by only rejecting a call without any common codecs by @cHuberCoffee in baresip#1523
- multicast: add missing join for multicast addresses by @cHuberCoffee in baresip#1524
- confg,uag: rework on sip_transports setting by @cspiel1 in baresip#1525
- ua: check if peer is capable of video for early video by @cHuberCoffee in baresip#1526
- mqtt/subscribe: replace fixed command buf and increase response size by @sreimers in baresip#1527
- mqtt: add reconnect handling (lost broker connection) by @sreimers in baresip#1528
- event: increase module_event buffer size by @sreimers in baresip#1532
- mqtt/subscribe: use safe odict_string to prevent crashes by @sreimers in baresip#1534
- stream: add stream_set_label by @alfredh in baresip#1537
- Makefile dependency check improvements by @sreimers in baresip#1531
- account: add enable/disable flag for video by @cspiel1 in baresip#1536
- audio: use account specific audio telev pt correctly by @cspiel1 in baresip#1542
- net: add missing HAVE_INET6 by @cspiel1 in baresip#1543
- account: remove unused API function for video enable by @cspiel1 in baresip#1544
- gst: changed log level for end of file message by @RobertMi21 in baresip#1548
- multicast: add new configurable multicast TTL config parameter by @cHuberCoffee in baresip#1545
- call: fix early video capability check (wrong SDP direction checked) by @cHuberCoffee in baresip#1549
- audio: catch end of file message in ausrc error handler (#1539) by @RobertMi21 in baresip#1550
- menu: added stopringing command by @RobertMi21 in baresip#1551
- stream: remove obsolete rx.jbuf_started by @cspiel1 in baresip#1552
- ua: downgrade level of message "ua: using best effort AF" by @viordash in baresip#1553
- outgoing calls early callid by @cspiel1 in baresip#1547
- audio: changed log level for ausrc error handler messages by @RobertMi21 in baresip#1554
- SIP default protocol by @cspiel1 in baresip#1538
- serreg: fix server selection in case all server were unavailable by @cHuberCoffee in baresip#1557
- multicast: fix missing unlock by @alfredh in baresip#1559
- config: replace strcpy by saver re_snprintf (#1558) by @cspiel1 in baresip#1560
- multicast: fix coverity scan by @alfredh in baresip#1561
- odict: hide struct odict_entry by @sreimers in baresip#1562
- ctrl_dbus: use mqueue to trigger processing of command in remain thread by @cspiel1 in baresip#1565
- multicast,config: add separate jitter buffer configuration by @cspiel1 in baresip#1566
- ua: emit CALL_CLOSED event when user agent is deleted by @cspiel1 in baresip#1564
- core: move stream_enable_rtp_timeout to api by @sreimers in baresip#1569
- stream: add mid sdp attribute by @alfredh in baresip#1570
- rtpext: change length type to size_t by @alfredh in baresip#1573
- avcodec: remove old backwards compat wrapper by @alfredh in baresip#1575
- main: Added option (-a) to set the ua agent string. by @RobertMi21 in baresip#1576
- menu fix tones for parallel outgoing calls by @cspiel1 in baresip#1577
- Fix win32 by @viordash in baresip#1579
- Fix static analyzer warnings by @viordash in baresip#1580
- call: added auto dtmf mode by @RobertMi21 in baresip#1583
- RTP inbound telephone events should not lead to packet loss by @cspiel1 in baresip#1581
- Running tests in a win32 project by @viordash in baresip#1585
- stream: wrong media direction after setting stream to hold by @RobertMi21 in baresip#1587
- move network check to module by @cspiel1 in baresip#1584
- serreg: do not ignore returned errors of ua_register() by @cspiel1 in baresip#1589
- Bundle media mux by @alfredh in baresip#1588
- mixausrc: no warnings flood when sampc changes by @cspiel1 in baresip#1595
- ua: select laddr with route to SDP offer address by @cspiel1 in baresip#1590
- net,uag: allow incoming peer-to-peer calls with user@domain by @cspiel1 in baresip#1591
- uag: in uag_reset_transp() select laddr with route to SDP raddr by @cspiel1 in baresip#1592
- uag: exit if transport could not be added by @cspiel1 in baresip#1593
- avcodec: use const AVCodec by @alfredh in baresip#1602
- module: deprecate module_tmp by @alfredh in baresip#1600
- test: use ausine as audio source by @alfredh in baresip#1601
- Selftest fakevideo by @alfredh in baresip#1604
- When adding local address, check that it has not been added already by @juha-h in baresip#1606
- start without network by @cspiel1 in baresip#1607
- config: add netroam module by @sreimers in baresip#1608
- multicast: allow any port number for sender and receiver by @cHuberCoffee in baresip#1609
- netroam: add netlink immediate network change detection by @cspiel1 in baresip#1612
- remove uag transp rm (#1611) by @cspiel1 in baresip#1616
- net dns srv get by @cspiel1 in baresip#1615
- move calls to stream_start_rtcp to call.c by @alfredh in baresip#1617
- video: null pointer check for the display handler by @cspiel1 in baresip#1621
- audio: add lock by @alfredh in baresip#1619
- ua: select proper af and laddr for outgoing IP calls by @cspiel1 in baresip#1618
- audio: lock stream by @alfredh in baresip#1622
- test: replace mock ausrc with ausine by @alfredh in baresip#1623
- menu ringback session progress by @cspiel1 in baresip#1625
- New module providing webrtc aec mobile mode filter by @juha-h in baresip#1626
- uag: respect setting sip_listen (#1627) by @cspiel1 in baresip#1628
- select laddr for SDP with respect to net_interface by @cspiel1 in baresip#1630
- stream: do not start audio during early-video by @cspiel1 in baresip#1629
- remove struct media_ctx by @alfredh in baresip#1632
- ci: add libwebrtc-audio-processing-dev (module webrtc_aec) by @sreimers in baresip#1635
- auconv: new module for audio format conversion by @alfredh in baresip#1634
- Support for IPv6 link local address for streams by @cspiel1 in baresip#1624
- call: check if address family is valid also for video stream by @cspiel1 in baresip#1636
- audio: pass pointer to tx->ausrc_prm instead of local variable by @cspiel1 in baresip#1637
- menu: add an event for call transfer by @cspiel1 in baresip#1641
- netroam: error handling for reset transport by @cspiel1 in baresip#1642
- mk: use CC_TEST for auto detect modules by @sreimers in baresip#1647
- test: use dtls_srtp.so module instead of mock by @alfredh in baresip#1646
- stream: create jbuf only if use_rtp is set by @cspiel1 in baresip#1648
- multicast: fix memleak in player destructor by @cspiel1 in baresip#1653
- stream: split up sender/receiver by @alfredh in baresip#1654
- set sdp laddr to SIP src address by @cspiel1 in baresip#1645
- serreg fix fallback accounts by @cspiel1 in baresip#1660
- ctrl_dbus: print command with the warning by @cspiel1 in baresip#1662
- call: new transfer call state to handle transfered calls correctly by @cHuberCoffee in baresip#1658
- serreg: prevent fast register retries if offline by @cspiel1 in baresip#1663
- av1: update packetization code by @alfredh in baresip#1657
- call: magic check in sipsess_desc_handler() by @cspiel1 in baresip#1664
- alsa: use snd_pcm_drop instead of snd_pcm_drain by @sreimers in baresip#1669
- Increased debian compat level to 10 by @juha-h in baresip#1667
- conf: fix conf_configure_buf() config parse by @sreimers in baresip#1666
- stream flush rtp socket by @cspiel1 in baresip#1671
- Transfer like rfc5589 by @cHuberCoffee in baresip#1678
- GTK: mem_derefer call earlier by @mbattista in baresip#1682
- netroam: add fail counter and event by @cspiel1 in baresip#1685
- Added API functions stream_metric_get_(tx|rx)_bitrate by @juha-h in baresip#1686
- Multicast new functions by @cHuberCoffee in baresip#1687
- avcodec: Enable pass-through for more codecs by @abrodkin in baresip#1692
- menu: filter for the correct call state in menu_selcall by @cHuberCoffee in baresip#1693
- test: fix warning on mingw32 by @alfredh in baresip#1696
- menu: Play ringback in play device by @myrkr in baresip#1698
- sip: add optional TCP source port by @cspiel1 in baresip#1695
- rtpext: change id unsigned -> uint8_t by @alfredh in baresip#1701
- ci: add mingw build test by @sreimers in baresip#1700
- test: use mediaenc srtp instead of mock by @alfredh in baresip#1702
- test: remove mock mediaenc by @alfredh in baresip#1704
- descr: add session_description by @alfredh in baresip#1706
- use fs_isfile() by @alfredh in baresip#1709
- stream: only call rtp_clear for audio by @alfredh in baresip#1710
- checks if call is available before calling call, closes #1708 by @mbattista in baresip#1712
- conf: add conf_loadfile by @alfredh in baresip#1713
- ice: remove ice_mode by @sreimers in baresip#1714
- audio: use auframe in encode_rtp_send, ref #1699 by @alfredh in baresip#1715
- Increased account's max video codec count from four to eight by @juha-h in baresip#1717
- gtk: Avoid duplicate call_timer registration by @myrkr in baresip#1719
- Attended call transfer by @cHuberCoffee in baresip#1718
- menu: exclude given call when searching for active call by @cspiel1 in baresip#1721
- menu: play call waiting tone on audio_player device by @cspiel1 in baresip#1722
- ci/build/macos: link ffmpeg@4 by @sreimers in baresip#1725
- module auresamp by @cspiel1 in baresip#1705
- test: remove h264 testcode, already in retest by @alfredh in baresip#1726
- h265: move from avcodec to rem by @alfredh in baresip#1728
- mc: send more details at receiver - timeout event by @cHuberCoffee in baresip#1731
- h265: move packetizer from avcodec to rem by @alfredh in baresip#1732
- FFmpeg 5 by @sreimers in baresip#1734
- Fixing clang ThreadSanitizer warnings by @sreimers in baresip#1730
- auresamp: replace anonymous union for pre C11 compilers by @cspiel1 in baresip#1738
- aufile: align naming of alloc handlers by @sreimers in baresip#1739
- auresamp fixes by @cspiel1 in baresip#1741
- mc: new priority handling with multicast state by @cHuberCoffee in baresip#1740
- remove support for Solaris platform by @alfredh in baresip#1745
- Allow hanging up call that has not been ACKed yet by @juha-h in baresip#1747
- Multicast identical condition and fmt string fix by @cHuberCoffee in baresip#1751
- audio: allocate aubuf before ausrc_alloc (fixes data race) by @sreimers in baresip#1748
- call: send supported header for 200 answering/ok by @cHuberCoffee in baresip#1752
- event: check if media line is present for encoding audio/video dir by @cspiel1 in baresip#1754
- Removed unused variable in modules/webrtc_aec/aec.cpp by @juha-h in baresip#1756
- audio use module auconv by @cspiel1 in baresip#1742
- test: use aufile module by @alfredh in baresip#1757
- x11grab: remove module, use avformat.so instead by @alfredh in baresip#1758
- audio: declare iterator inside for-loop (C99) by @alfredh in baresip#1759
- aufile: set run=true before write thread starts (#1727) by @cspiel1 in baresip#1762
- Added new API function call_supported() and used it in menu module by @juha-h in baresip#1761
- aufile: separate aufile_src.c from aufile.c by @cspiel1 in baresip#1765
- ctrl_dbus: fix possible data race (#1727) by @cspiel1 in baresip#1764
- menu select other call on hangup by @cspiel1 in baresip#1763
- event: encode also combined media direction by @cspiel1 in baresip#1766
- @srperens made their first contribution in baresip#1399
- @negbie made their first contribution in baresip#1451
- @andreaswatch made their first contribution in baresip#1512
- @viordash made their first contribution in baresip#1553
- @abrodkin made their first contribution in baresip#1692
- @myrkr made their first contribution in baresip#1698
1.1.0 - 2021-04-24
- cons: emulate key-release -- ref #1329
- Correct reverse domain name notation (#1342) #1342
- gtk with account_uri_complete (#1339) #1339
- bump version to 1.1.0 -- ref #1333
- ui: fix leaking of cmd_ctx (#1338) #1338
- DTMF tones for A B C D (#1340) #1340
- account: use a fixed username for the template
- contact: update contacts template
- config: disable ctrl_dbus in config template
- Module event (#1335) #1335
- add event UA_EVENT_MODULE to tell to app when snapshot has been written (#1330) #1330
- ringtone: generated busy and ringback tone (#1332) #1332
- audio: prevent restart of rx_thread on call termination (#1331) #1331
- modules: update auplay/ausrc modules
- Auplay remove inheritance (#1328) #1328
- h264: add doxygen comment
- vidloop: add VIDEO_SRATE
- vidloop: check error
- vidloop: add vidframe_clear
- vidloop: split enable_codec into encoder/decoder
- Ausrc remove inheritance (#1326) #1326
- ua: remove prev call (#1323) #1323
- sndfile: get number of bytes from auframe
- plc: check format of struct auframe
- speex_pp: check format of struct auframe
- webrtc_aec: use format from struct auframe
- README: update codecs and RFCs
- menu: use uri complete for command dialdir (#1321) #1321
- video: check for video display before calling handler
- Changed name and made public (#1319) #1319
- menu: return call-id for dial and dialdir (#1320) #1320
- Fixes for account uri complete (#1318) #1318
- Avoid compiler warnings:
- Avoid compiler warnings (I haven't found anything wrong with the code)
- vidfilt: fix warning
- vidfilt: split parameters into encode/decode
- snapshot: fix warnings
- video: group functions from vidutil.c
- avfilter: fix warnings
- vumeter: use format from audio frame
- replaced ua_uri_complete with account_uri_complete (#1317) #1317
- aulevel: move to librem
- omx: fix warning
- vidisp: remove inheritance (#1316) #1316
- docs: change video settings to match the default values (#1315) #1315
- menu: select call in cmd_find_call() (#1314) #1314
- menu: use menu_stop_play() (#1311) #1311
- main: unload app modules in signal handler (#1310) #1310
- avformat: replace const double with double
- avformat: clean up ifdefs (#1313) #1313
- ci: drop ubuntu 16.04 support - end of life
- avformat: proper code formatting
- avcodec: add avcodec prefix to log messages
- avcodec: check length of H265 packet
- x11grab: remove vidsrc inheritance
- v4l2: remove vs inheritance
- vidsrc: remove concept of baseclass/inheritance
- ua,menu: remove uag_find_call_state (#1304) #1304
- Updated homepage
- sdl: correct aspect-ratio in fullscreen mode
- vidloop: add vidisp parameters
- auloop: use auframe_size
- audio: use auframe_size
- Auplay use auframe (#1305) #1305
- Docs examples config (#1302) #1302
- Serreg fixes (#1301) #1301
- Update config.c #1303
- contact: use uag_find_requri()
- ua: use new tls function to set cafile and path #1300
- config: add sip_capath config line
- Call event answered fixes alsa issue (#1299) #1299
- ctrl_dbus: send DBUS signal when dbus interface is ready (#1296) #1296
- Multicast call priority (#1291) #1291
- Menu fixes for play tones2 (#1294) #1294
- gst: add missing include unistd.h #1297
- multicast: cleanup function description and fix doxygen warning (#1292) #1292
- menu: remove call resume for command hangup (#1289) #1289
- ua: add a generic filter API for calls (#1293) #1293
- Merge pull request #1288 from cspiel1/remove_call_resume_on_termination #1288
- menu: remove call resume on termination
- multicast: fix build error when using HAVE_PTHREAD=
- alsa_play.c add suggestion to use dmix (#1283) #1283
- readme.md: added multicast module (#1282) #1282
- audiounit: fix typo
- update copyright year (#1287) #1287
- config cleanup (#1286) #1286
- update copyright year (#1285) #1285
- conf: add call_hold_other_calls config option (#1280) #1280
- config.c: added rtmp to config template (#1284) #1284
- main.c: update year #1281
- The avformat_decoder should be optional (#1277) #1277
- src/audio: set started false with audio_stop (#1278) #1278
- readme: update baresip fork links
- ausine: mono support and stereo_left/right option #1274
- menu: fix incoming calls are not selected on call termination (#1271) #1271
- test: remove mock_aucodec, using g711 instead
- opengl: remove deprecated module (#1268) #1268
- Added account_dtmfmode and account_set_dtmfmode API functions (#1269) #1269
- avcodec: remove support for MPEG4 codec
- call: start streams asynchronously (issue #1261) (#1267) #1267
- audio: remove special handling of Comfort Noise
- multicast: fix one doxygen warning
- menu: update doxygen comment
- menu: correct hangupall command for parallel call feature (#1264) #1264
- menu: on call termination select another active call (#1260) #1260
- ua: correct doxygen of uag_hold_resume() #1262
- menu: simplify cmd_hangupall() (#1259) #1259
- support for sending of DTMF INFO (#1258) #1258
- Menu optional call parameter (#1254) #1254
- cleanup tabs and spaces #1256
- ua: correct doxygen for uag_hold_others()
- ua: add doxygen for call find functions
- menu: add doxygen to cmd_hangup(), cmd_hold(), cmd_resume()
- menu: command accept searches all User-Agents for an incoming call
- ua: add function uag_find_call_state()
- menu: print correct warning for hangup, accept, hold, resume
- menu: add optional parameter call-id to cmd_call_resume()
- menu: add optional parameter call-id to cmd_call_hold()
- menu: add optional parameter call-id to cmd_hangup()
- menu: add optional parameter call-id to cmd_answerdir()
- menu: add utility function that decodes complex command parameters
- menu: use SDP_SENDRECV for cmd_answerdir() as fallback
- menu: add optional parameter call-id to cmd_answer()
- ua: add call find per call-id function
- call: call_info() prints also the call-id
- ua: in ua_print_calls() print User-Agent info in header
- menu: ua NULL check for answer command
- replace spaces with tab #1249
- removed newline
- undid httpreq spacing
- fixed line too long
- moved multicast template to end of config template
- ua: fix uag_hold_others use of wrong list element #1253
- added multicast enabled message (#1251) #1251
- updated date and added multicast to signaling (#1252) #1252
- Merge pull request #1248 from webstean/patch-2 #1248
- Added newline to multicast comment
- Menu ensure only one established call (#1247) #1247
- Call resume on hangup (#1246) #1246
- menu: for call answer search all UAs for calls to put on hold
- ua: ua_answer() should answer same call like ua_hold_answer()
- ua: make ua_find_call_state() global usable
- Add multicast_listener to config template (#1245) #1245
- Update config template to include multicast module (#1244) #1244
- menu: if a call becomes established then put others on hold
- ua: add uag_hold_others()
- Fix multiple resumed calls (#1242) #1242
- Merge pull request #1241 from cHuberCoffee/cmd_hangupall #1241
- RFC: Make avformat decode mjpeg v4l2 with vaapi (#1216) #1216
- ua: add doxygen for new uag_hold_resume()
- menu: fix missing callid of menu at call closed
- menu: use uag_hold_resume to ensure only one active call
- ua: on call resume check for other active calls
- menu: new hangupall command with direction parameter
- readme: update supported compilers and ssl libs
- menu: fix redial
- Fix spaces
- Multicast module (#1231) #1231
- menu: use print backend pointer pf correctly (#1222) #1222
- menu: start ringback only once for parallel calls (#1238) #1238
- jack: support port pattern in config file (#1237) #1237
- config: disables server verification if sip_verify_server is missing (#1236) #1236
- ua: for UA selection allow arbitrary aor for regint=0 accounts (#1234) #1234
- Ctrl dbus synchronize (#1232) #1232
- event: encode also remote audio direction (#1227) #1227
- Merge pull request #1235 from cspiel1/event_add_string_for_UA_EVENT_CUSTOM #1235
- event: add string for UA_EVENT_CUSTOM
- Mimic ifdef on avutil version for hwcontext
- Fix to tabs and improve checks
- src/config: show sip_cafile warning only if sip_verify_server is enabled
- Avoid compiler warnings using casts #1228
- test: disable SIP TLS server verification #1224
- config,ua: add config flag disable SIP TLS server verification
- alsa/play: snd_pcm_writei error codes are negative
- alsa: fix clang warnings "conversion loses integer precision" #1223
- Intelligent call answer (#1218) #1218
- Remove uag next (#1207) #1207
- Merge pull request #1219 from cspiel1/message_reply_once #1219
- menu: update switch_audio_player
- Make vaapi/mjpeg options of avformat
- src/config: no sip_cafile wording
- message: reply only once
- src/ua: only warn if tls_add_ca fails, same as undefined cafile #1214
- src/config: add sip_cafile warning and enable by default
- ua: change log message from warning to info
- video: fix video payload text
- Make avformat decode mjpeg v4l2 with vaapi
- ua: improve UA selection for incoming calls (#1206) #1206
- ua: limit account matches for incoming calls to non-registrar accounts
- ua: check for NULL parameter in uag_find_msg()
- ua: early exit for AF_UNSPEC in uri_match_af()
- ua: use sip_transp_decode() in uri_match_transport()
- ua: use arrays in uri_host_local()
- test: add test for deny UDP peer-to-peer call
- ua: improve UA selection for incoming calls
- Sip message to application (#1201) #1201
- opus: Ensure (re)init of fmtp strings (#1209) #1209
- ctrl_dbus: generate dbus interface during build (#1208) #1208
- mod_gtk: switch to gtk 3 (#1203) #1203
- menu: set_answer_mode: apply all uas
- menu: find_call: search all user-agents
- menu: fix usage of ua
- isac: remove deprecated module (#1204) #1204
- menu: cmd_print_calls: print all uas
- Fix interaction between CLI menu and GTK menu (#1202) #1202
- menu: rename menu_current() to menu_uacur()
- webrtc_aec: fix compilation with gcc 4.9 (fix #1193)
- win32: add cons module, fixes #1197
- ua: remove ua_aor() -- use account_aor() instead
- gtk: use account_aor()
- menu: use account_aor()
- presence: use account_aor()
- modules: use account_aor()
- account: fix video codes decode (#1196) #1196
- core: use account_aor()
- Merge pull request #1198 from baresip/av1 #1198
- Avoid unused parameter warning
- debug_cmd: add UA_EVENT_CUSTOM (#1194) #1194
- fix decoder changed debug text
- cairo: minor debug tuning
- menu: add uadelall to delete all user agents #1195
- use account_aor()
- mctrl: remove support for media-control (deprecated)
- update doxygen comments
- ua: minor cleanup
- ua: split struct uag from instance
- README: add RFC 5373
- menu: fix segfault on last account deletion (#1192) #1192
- call: extend SIP auto answer support for incoming calls (#1191) #1191
- Sip auto answer caller (#1188) #1188
- win32: remove timer.c
- ua: give a nice name to 'global' struct
- ua: remove ua_cur
- move uag_current to menu module
- menu: pass ua from mqtt to menu via opaque data
- Sip autoanswer callee (#1187) #1187
- ua: for answer-mode early also send INCOMING event (#1185) #1185
- gst: The error handler call for end of stream is now (#1182) #1182
- mk: also detect mqtt.so in SYSROOT_ALT
- contact: add ua_lookup_domain
- video: minor tuning of pipeline text
- gst: playback of read only audio files failed (#1183) #1183
- gtk: make a local pointer to current ua
- menu: clean up usage of uag_current()
- call: correction of remote video direction info at SDP-offer (#1181) #1181
- debug_cmd: print all user-agents
- presence: one command with status as argument
- ua: rename presence status to pstat
- ua: remove LIBRE_HAVE_SIPTRACE check, always enabled
- update doxygen comments
- mk: update doxygen config file
- menu: initialize menu with zeros (#1179) #1179
- Re mk cross build2 (#1161) #1161
- net: make fallback DNS ignored message debug only
- mixausrc: improve logging #1176
- mixausrc: fix shorten-64-to-32 warnings
- config: template for osx/ios
- Supressed clang zero length array warning
- Added ctx param to video_stop/video_stop_source and set ctx to null (#1173) #1173
- avformat: add empty line after base class
- Make macos warnings into errors (#1171) #1171
- disable mixausrc until warnings are fixed
- clang shorten-64-to-32 warnings (#1170) #1170
- Mixausrc (#1159) #1159
- aufile: fix warning on OSX
- alsa: print warning if running, fixed #1162
- Don't default stunuser/pass to account authuser/pass (#1164) #1164
- Audio file info (#1157) #1157
- gitignore: clangd cache, compile_commands.json and cleanup
- Merge pull request #1167 from baresip/video_display #1167
- Reordered video_stop_display
- Expose video_stop_display() to API
- Video dir rename (#1158) #1158
- ci: use baresip/rem repo
- stream: add function to send a RTP dummy packet (#1156) #1156
- Play aufile extended support (#1155) #1155
- video: move video related start/stop/update into video file (#1151) #1151
- aufile: add audio player to write speaker data to wav file (#1153) #1153
- Fix compiler warnings (#1152) #1152
- play: fix warning
- play ausrc (#1147) #1147
- README: add more status badges
- README: replace travis status badge
- menu: fix uint16_t scode #1149
- config: revert dirent.h changes
- audio: fix HAVE_PTHREAD audio_destructor
- gst ready for file play (#1148) #1148
- debug_cmd: mem_deref of player fixes segfault (#1146) #1146
- net: remove deprecated net_domain()
- update contact examples
- fix freeze on hangup (#1135) (#1145) #1145
- menu: make audio files configurable (#1144) #1144
- aptx: declare variable outside for-loop
- fix warnings on openbsd
- jack: declare variable outside for loop
- account: declare variable outside for loop
- coreaudio: declare variable outside for loop
- menu: initialize menu.play fixes segfault (#1143) #1143
- ausine: declare variable outside for loop
- timer: remove tmr_jiffies_usec (replaced by libre) (#1141) #1141
- Adaptive jbuf (#1112) #1112
- Update build.yml (#1140) #1140
- mqtt: allow to separate pub from sub topic base (#1139) #1139
- video: fix warning
- mqtt: fix printing port and add tls support (#1138) #1138
- httpreq: in cmd_setauth check if parameter was given (#1134) #1134
- Merge pull request #1132 from baresip/pr-dependency-action #1132
- ci: add pull request dependency checkouts
- audio: remove redundant union
- menu: use menu_ as prefix for global symbols
- menu: use menu_ as prefix for global symbols
- ci: add apt-get update
- menu: module refactoring (#1129) #1129
- audio, video, stream: check payload type before put to jbuf (#1128) #1128
- Cmd dialdir (#1126) #1126
- Cmd acceptdir (#1125) #1125
- event: add register fallback to event string and class name (#1124) #1124
- avformat: use %u for unsigned
- modify event type and check if peeruri null (#1119) #1119
- event: move code from ua.c (#1118) #1118
- Valgrind ci (#1117) #1117
- h264 cleanup, second part (#1115) #1115
- h264 cleanup (#1114) #1114
- Merge pull request #1113 from baresip/github-actions-v2 #1113
- ci: remove travis
- ci: add github actions - replaces travisci
- qtcapture: remove deprecated module (#1107) #1107
- test: prepare for dualstack
- test: add mock dns_server_add_aaaa
- make EXTRA_MODULES last, not first (#1106) #1106
- httpreq: fix cmd_settimeout
- test: bind network to localhost, a fix for #1090
- modules/webrtc_aec: link flags fixes (#1105) #1105
- menu: commands in alphabetical order
- httpreq: fix warning about unused args
- serreg: fix warnings about unused argument
- menu: fix warnings about unused argument
- Add a HTTP request module with authorization (#1099) #1099
- Menu: corrections for ring tones and call status by means of a global call counter (#1102) #1102
- mk: remove dirent.h
- Updating .vcxproj file for windows builds (#1097) #1097
- ccheck: change license to BSD license
- Merge pull request #1095 from baresip/websocket #1095
- Serial registration (#1083) #1083
- Ctrl dbus (#1085) #1085
- README: remove references to creytiv.com
- Branch of baresip that includes Alfred's sip websocket patch
- Merge pull request #1091 from baresip/debian #1091
- ua, menu: new command to print certificate issuer and subject (#1078) #1078
- .gitignore: add ctags and Vim swp files (#1084) #1084
- alfredh
- robert-scheck
- mbattista
- cspiel1
- juha-h
- ahinrichs
- jurjen-van-dijk
- sreimers
- cHuberCoffee
- webstean
- viric
- agramner
- weili-jiang
- thillux
- wkiswk
- philippbachmann08
- ursfassler
- RobertMi21
- alberanid
- agranig
- nanguantong
- johnjuuljensen
1.0.0 - 2020-09-11
- aac: add AAC_STREAMTYPE_AUDIO enum value
- aac: add AAC_ prefix
- Video mode param to call_answer(), ua_answer() and ua_hold_answer #966
- video_stop_display() API function #977
- module: add path to module_load() function
- conf: add conf_configure_buf
- test: add usage of g711.so module #978
- JSON initial codec state command and response #973
- account_set_video_codecs() API function #981
- net: add fallback dns nameserver #996
- gtk: show call_peername in notify title #1006
- call: Added call_state() API function that returns enum state of the call #1013
- account_set_stun_user() and account_set_stun_pass() API functions #1015
- API functions account_stun_uri and account_set_stun_uri. #1018
- ausine: Audio sine wave input module #1021
- gtk/menu: replace spaces from uri #1007
- jack: allowing jack client name to be specified in the config file #1025 #1020
- snapshot: Add snapshot_send and snapshot_recv commands #1029
- webrtc_aec: 'extended_filter' config option #1030
- avfilter: FFmpeg filter graphs integration #1038
- reg: view proxy expiry value in reg_status #1068
- account: add parameter rwait for re-register interval #1069
- call, stream, menu: add cmd to set the direction of video stream #1073
- Added AMRWBENC_PATH env var to amr module module.mk #1081
- Using baresip/re fork now
- audio: move calculation to audio_jb_current_value
- avformat: clean up docs
- gzrtp: update docs
- account: increased size of audio codec list to 16
- video: make video_sdp_attr_decode public
- config: Derive default audio driver from default audio device #1009
- jack: modifying info message on jack client creation #1019
- call: when video stream is disabled, stop also video display #1023
- dtls_srtp: use tls_set_selfsigned_rsa with keysize 2048 #1062 #1056
- rst: use a min ptime of 20ms
- aac: change ptime to 4ms
- avcodec: fix H.264 interop with Firefox
- winwave: waveInGetPosition is no longer supported for use as of Windows Vista #960
- avcodec: call av_hwdevice_ctx_create before if-statement
- account: use single quote instead of backtick
- ice: fix segfault in connh #980
- call: Update call->got_offer when re-INVITE or answer to re-INVITE is received #986
- mk: Test also for /usr/lib64/libspeexdsp.so to cover Fedora/RHEL/CentOS #992
- config: Allow distribution specific CA trust bundle locations (fixes #993
- config: Allow distribution specific default audio device (fixes #994
- mqtt: fix err is never read (found by clang static analyzer)
- avcodec: fix err is never read (found by clang static analyzer)
- gtk: notification buttons do not work on Systems #1012
- gtk: fix dtmf_tone and add tones as feedback #1010
- pulse: drain pulse buffers before freeing #1016
- jack: jack_play connect all physical ports #1028
- Makefile: do not try to install modules if build is static #1031
- gzrtp: media_alloc function is missing #1034 #1022
- call: when updating video, check if video stream has been disabled #1037
- amr: fix length check, fixes #1011
- modules: fix search path for avdevice.h #1043
- gtk: declare variables C89 style
- config: init newly added member
- menu: fix segfault in ua_event_handler #1059 #1061
- debug_cmd: fix OpenSSL no-deprecated #1065
- aac: handle missing bitrate parameter in SDP format
- av1: properly configure encoder
- call: When terminating outgoing call, terminate also possible refer subscription #1082
- menu: fix segfault in /aubitrate command
- amr: should check if file (instead of directory) exists
- ice: remove support for ICE-lite
- ice: remove ice_debug, use log level DEBUG instead
- ice: make stun server optional
- config: remove ice_debug option (unused)
- opengles: remove module (not working) #1079
- Alfred E. Heggestad
- Alexander Gramner
- Andrew Webster
- Christian Spielberger
- Christoph Huber
- Davide Alberani
- Ethan Funk
- Juha Heinanen
- mbattista
- Michael Malone
- Mikl Kurkov
- ndilieto
- Robert Scheck
- Roger Sandholm
- Sebastian Reimers