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Baresip Changelog

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.

v3.18.0 - 2024-12-11

What's Changed

  • cmake: fix clang 19 c23 extension warnings by @sreimers in baresip#3189
  • cmake: install logo.png and *.wav files again (fixes #3191) by @robert-scheck in baresip#3194
  • menu dnd event by @cspiel1 in baresip#3186
  • ci/build: use GCC 14 on Ubuntu 24.04 by @robert-scheck in baresip#3193
  • ci/fedora: adapt workflow for RPM 4.20 in Fedora 41 (fixes #3183) by @robert-scheck in baresip#3192
  • test: disable test_call_webrtc in thread mode by @alfredh in baresip#3190
  • debian: release v3.17.1 by @sreimers in baresip#3195
  • ci/mingw: use ubuntu-24.04 and bump openssl by @sreimers in baresip#3196
  • menu: fix 302 Moved Temporarily redirect call by @cspiel1 in baresip#3198
  • core: remove obsolete dnd flag by @cspiel1 in baresip#3200
  • wasapi: Add WASAPI Audio module by @sreimers in baresip#3165
  • menu: a config flag for SIP MESSAGE tone by @cspiel1 in baresip#3203
  • mixausrc: fixing some type conversion warnings (MSVC) by @sreimers in baresip#3205
  • winwave: remove (replaced by WASAPI module) by @sreimers in baresip#3204
  • pulse: use default string as default device by @sreimers in baresip#3206
  • mixausrc: reset mixstatus ptime if ptime changes by @maximilianfridrich in baresip#3216
  • bevent: Add local URI information to events by @mrkiko in baresip#3217
  • ua: update doxygen comment by @alfredh in baresip#3218
  • AAudio module by @juha-h in baresip#3208
  • wasapi/src: increase record buffer size by @sreimers in baresip#3221
  • ua,call: add API for rejecting incoming call by @cspiel1 in baresip#3228
  • video: add picture update bool to vidpacket by @maximilianfridrich in baresip#3227
  • video: allow RTCP FIR packets as payload-specific feedback by @maximilianfridrich in baresip#3230
  • cmake: fix APPLE RPATH workaround by @sreimers in baresip#3232
  • readme: sync with wiki by @alfredh in baresip#3241
  • bump version to 3.18.0 by @alfredh in baresip#3243

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v3.17.0...v3.18.0

v3.17.0 - 2024-11-06

What's Changed

  • conf: use str_dup for conf_path_set by @sreimers in baresip#3150
  • menu: fix early audio limit for video less call by @cspiel1 in baresip#3152
  • jbuf: add video generic NACK support by @sreimers in baresip#3154
  • rtprecv: remove obsolete code by @sreimers in baresip#3156
  • mediatrack: decode sdp attributes for audio by @alfredh in baresip#3161
  • avcodec: remove swresample from list of packages by @alfredh in baresip#3159
  • bevent: avoid deprecated warnings in selftest by @cspiel1 in baresip#3163
  • test: rename bevent tests to match module api by @alfredh in baresip#3168
  • conf: remove conf_get_float by @juha-h in baresip#3169
  • GTK added standalone main window by @mbattista in baresip#3127
  • augain: add audio volume filter module by @juha-h in baresip#3166
  • test/call.c: test_call_sni: silence warnings by @maximilianfridrich in baresip#3173
  • v4l2/cmake: exclude from WIN32 by @sreimers in baresip#3174
  • ci/build: upgrade to ubuntu 24.04 by @alfredh in baresip#3172
  • test: enable RTP/RTCP multiplexing for test_call_webrtc by @alfredh in baresip#3178
  • webrtc_aecm: added -Wno-missing-field-initializers compile option by @juha-h in baresip#3177
  • audio: fix variadic macro warning of clang by @cspiel1 in baresip#3181
  • test/call: add test_call_cancel by @maximilianfridrich in baresip#3180
  • ci/fedora: Use Fedora 40 until CI is updated for Fedora 41 by @robert-scheck in baresip#3184
  • dtls_srtp: use struct udp_sock for socket type (ref #3175) by @alfredh in baresip#3182

Full Changelog: https://github.com/baresip/baresip/compare/v3.16.0...v3.17.0

v3.16.0 - 2024-08-28

What's Changed

  • ua,menu: do not accept call by default by @cspiel1 in baresip#3123
  • call: fix emitting of call and ua events by @maximilianfridrich in baresip#3133
  • ausrc, aufile, gst, debug_cmd: rework on ausrc duration retrieval by @cspiel1 in baresip#3105
  • ua: doxygen for ua_accept() by @cspiel1 in baresip#3139
  • audio: initialize ausrc_prm by @cspiel1 in baresip#3138
  • sndfile: module_event from main thread by @cspiel1 in baresip#3141
  • jack: add jack_server_name config by @sreimers in baresip#3143
  • bevent: move dnd to menu by @cspiel1 in baresip#3140
  • call: a getter for local media directions by @cspiel1 in baresip#3147
  • menu: stop event processing on DnD by @cspiel1 in baresip#3148

Full Changelog: https://github.com/baresip/baresip/compare/v3.15.0...v3.16.0

v3.15.0 - 2024-08-28

What's Changed

  • audiounit: remove deprecated iOS session handling by @sreimers in baresip#3083
  • audiounit/cmake: use CMAKE_HOST_SYSTEM_NAME by @sreimers in baresip#3086
  • add filter_registrar option by @maximilianfridrich in baresip#3080
  • feature: in-band dtmf detection and generation by @marcel-behlau-elfin in baresip#3072
  • gst: fix potential SEGVs in error cases by @maximilianfridrich in baresip#3091
  • test/call: fix SNI selftest by @maximilianfridrich in baresip#3092
  • ice: fix crash during ICE on fast disconnect by @paresy in baresip#3093
  • test/call: set global certificate again in SNI test by @maximilianfridrich in baresip#3094
  • uag: match transport and af when finding ua for msg by @maximilianfridrich in baresip#3096
  • bevent: add core bevent API by @cspiel1 in baresip#3090
  • aureceiver: set aubuf id by @sreimers in baresip#3097
  • sip/transp: add client certificate to all TLS transports by @maximilianfridrich in baresip#3095
  • aufilt: add aufilt_enable by @sreimers in baresip#3098
  • account: add set audio devices by @sreimers in baresip#3099
  • bevent: add backwards wrappers by @cspiel1 in baresip#3100
  • sndfile: add close file module events by @sreimers in baresip#3101
  • aufile: fixes for prm->duration by @cspiel1 in baresip#3102
  • account specific video source and display by @paresy in baresip#3104
  • ua: remove all matching headers when calling ua_rm_custom_hdr by @maximilianfridrich in baresip#3122
  • bevent: update function calls in src by @cspiel1 in baresip#3106
  • test: use bevent API by @cspiel1 in baresip#3107
  • menu: use new bevent API by @cspiel1 in baresip#3108
  • ctrl_dbus: use new bevent API by @cspiel1 in baresip#3109
  • ctrl_tcp: use new bevent API by @cspiel1 in baresip#3110
  • echo: use new bevent API by @cspiel1 in baresip#3111
  • gtk: use new bevent API by @cspiel1 in baresip#3112
  • mqtt: use new bevent API by @cspiel1 in baresip#3113
  • ebuacip: use new bevent API by @cspiel1 in baresip#3118
  • serreg: use new bevent API by @cspiel1 in baresip#3117
  • rtcpsummary: use new bevent API by @cspiel1 in baresip#3116
  • presence: use new bevent API by @cspiel1 in baresip#3115
  • mwi: use new bevent API by @cspiel1 in baresip#3114
  • presence: publisher - use new bevent API by @cspiel1 in baresip#3124
  • bevent: fix typo in deprecated warning by @cspiel1 in baresip#3130
  • mwi: use bevent_ua_emit() by @cspiel1 in baresip#3131
  • menu: use new bevent_call_emit() by @cspiel1 in baresip#3132
  • version 3.15.0 by @alfredh in baresip#3134

Full Changelog: https://github.com/baresip/baresip/compare/v3.14.0...v3.15.0

v3.14.0 - 2024-07-23

What's Changed

  • Add .vscode to .gitignore (cosmetic change) by @larsimmisch in baresip#3071
  • mediatrack: fix audio_start_source aufiltl setup order by @sreimers in baresip#3074
  • mixausrc: minor fixes by @marcel-behlau-elfin in baresip#3073
  • srtp: Fix base64 key decoding for AEAD_AES_256_GCM by @weili-jiang in baresip#3079
  • .clangd: suppress -Wgnu-zero-variadic-macro-arguments by @maximilianfridrich in baresip#3078
  • test: add TLS SNI selftest by @maximilianfridrich in baresip#3077
  • portaudio: add hostApi name prefix to mediadev by @sreimers in baresip#3081
  • call: fix return value in call_progress_dir() by @alfredh in baresip#3082
  • pipewire: lock loop while starting and registry scan by @cspiel1 in baresip#3084

Full Changelog: https://github.com/baresip/baresip/compare/v3.13.0...v3.14.0

v3.13.0 - 2024-06-19

What's Changed

  • ua: simplify sdp_connection() by @alfredh in baresip#3031
  • call.c: allow INFO requests with no body by @maximilianfridrich in baresip#3035
  • audio,stream: video only call if no common audio codecs by @cspiel1 in baresip#3037
  • ua: avoid empty IP in SDP as local address by @cspiel1 in baresip#3038
  • audio: fix setup of audio filters by @cspiel1 in baresip#3046
  • event refactoring by @cspiel1 in baresip#3018
  • audio: remove cast to integer of sampv by @cspiel1 in baresip#3050
  • auresamp: correct sampv size for conversion to float by @cspiel1 in baresip#3051
  • ausine: add float format support by @sreimers in baresip#3052
  • auresamp: fix size of sampv if aufmt conversion needed by @cspiel1 in baresip#3053
  • test: call - complete test for auconv by @cspiel1 in baresip#3047
  • webrtc_aec: Add support for extended filter option by @marcel-behlau-elfin in baresip#3048
  • ci/sanitizers: use clang-17 and do not recover by @sreimers in baresip#3054
  • debug_cmd: command aufileinfo return answer by @cspiel1 in baresip#3030
  • gst: set ausrc_prm duration by @cspiel1 in baresip#3055
  • aufile: ptime zero check for ausrc by @cspiel1 in baresip#3056
  • ua,uag: support SUBSCRIBE only if handler is set by @maximilianfridrich in baresip#3057
  • uag: add Allow header to 405 SUBSCRIBE response by @maximilianfridrich in baresip#3058
  • mixausrc: fix downsampling by @cspiel1 in baresip#3059
  • ci/coverage: increase min coverage by @sreimers in baresip#3060
  • cmake/webrtc_aec: use PUBLIC stdc++ linking by @sreimers in baresip#3063
  • call: capitals for SIP INFO dtmf by @cspiel1 in baresip#3062
  • test: call - reset max_calls to default value by @cspiel1 in baresip#3067
  • test/ua: disable dns cache for reg_dns tests by @cspiel1 in baresip#3068

New Contributors

  • @marcel-behlau-elfin made their first contribution in baresip#3048

Full Changelog: https://github.com/baresip/baresip/compare/v3.12.0...v3.13.0

3.12.0 - 2024-05-15

What's Changed

  • video: stream enable/disable for re-INVITE/UPDATE by @cspiel1 in baresip#2982
  • srtp: allow rekeying of running streams by @cHuberCoffee in baresip#2975
  • audio: remove stop_aur() by @cspiel1 in baresip#2991
  • HAVE_INET6 is always defined by @alfredh in baresip#2992
  • audio: respect SDP media disabled flag by @cspiel1 in baresip#2997
  • test: fix test_message() by @alfredh in baresip#2995
  • avformat: do not use deprecated avcodec_close() by @cspiel1 in baresip#3002
  • ua: enforce magic cookie in Via branch by @maximilianfridrich in baresip#3003
  • uag: fix initializer by @cspiel1 in baresip#3001
  • misc: cppcheck fixes by @alfredh in baresip#3007
  • readme: fix lint status badge by @sreimers in baresip#3008
  • video: enable/disable stream at common point by @cspiel1 in baresip#3010
  • srtp: deactivate test_call_srtp_tx_rekey by @cHuberCoffee in baresip#3013
  • g722,g726: use SYSTEM spandsp include by @sreimers in baresip#3017
  • srtp: lock possible re-keying against usage in receive handler by @cHuberCoffee in baresip#3012
  • mc: move multicast to baresip-apps by @cspiel1 in baresip#3015
  • call,audio: remove audio start/stop redundancy by @cspiel1 in baresip#2999
  • aufile: use correct audio format S16LE for aubuf frames by @alfredh in baresip#3020
  • ci: bump pr dependency action by @sreimers in baresip#3023
  • docs,core: remove reference to multicast by @alfredh in baresip#3019
  • bump version by @alfredh in baresip#3027

Full Changelog: https://github.com/baresip/baresip/compare/v3.11.0...v3.12.0

3.11.0 - 2024-04-09

What's Changed

  • account: read catchall flag from accounts file by @cspiel1 in baresip#2925
  • vp8/encode: optimizations and target_bitrate fix by @sreimers in baresip#2936
  • vp8,vp9: fix deprecated decode codec init by @sreimers in baresip#2952
  • aureceiver: fix mtx_unlock on discard by @sreimers in baresip#2955
  • release v3.10.1 by @sreimers in baresip#2958
  • message: return 403 instead of 488 by @maximilianfridrich in baresip#2953
  • netroam/cmake: add optional netlink detection by @sreimers in baresip#2960
  • ci/sanitizers: add mmap rnd_bits workaround by @sreimers in baresip#2967
  • account: set inreq_allowed=yes as default by @maximilianfridrich in baresip#2961
  • account: use correct format %zu for printing outbound by @maximilianfridrich in baresip#2963
  • stream: fix empty rtcp_stats for rtx.ssrc reception reports by @sreimers in baresip#2969
  • stream: avoid sanitizer warnings for strm->tx by @cspiel1 in baresip#2949
  • avcodec: remove re_h264 extra header by @sreimers in baresip#2971
  • play: err handling and ensure eof by @cspiel1 in baresip#2972
  • stream: add stream_jbuf_stats() by @sreimers in baresip#2973
  • sndfile: write correct sample rate to WAV header by @cspiel1 in baresip#2976
  • tls: add session resumption setter by @maximilianfridrich in baresip#2977
  • avcodec: use util function to decode H.264 STAP-A by @alfredh in baresip#2978
  • mixausrc: fix ausrc resampling by @cspiel1 in baresip#2981
  • ci/build: remove obsolete for loop by @cspiel1 in baresip#2985

Full Changelog: https://github.com/baresip/baresip/compare/v3.10.1...v3.11.0

3.10.1 - 2024-03-12

  • aureceiver: security fix mtx_unlock on discard by @sreimers in baresip#2955

3.10.0 - 2024-03-06

What's Changed

  • cmake: use default value for CMAKE_C_EXTENSIONS by @sreimers in baresip#2893
  • cmake: add /usr/{local,}/include/re and /usr/{local,}/lib{64,} to FindRE.cmake by @robert-scheck in baresip#2900
  • test/main: fix NULL pointer arg on err by @sreimers in baresip#2902
  • ci: add Fedora workflow to avoid e.g. rpath issues by @robert-scheck in baresip#2904
  • mediatrack/start: add audio_decoder_set by @sreimers in baresip#2910
  • config: support distribution-specific/default CA paths by @robert-scheck in baresip#2905
  • readme: cosmetic changes by @robert-scheck in baresip#2911
  • ci/fedora: fix dependency by @sreimers in baresip#2912
  • config: add default CA path for Android by @robert-scheck in baresip#2913
  • transp,tls: add TLS client verification by @maximilianfridrich in baresip#2888
  • account,message,ua: secure incoming SIP MESSAGEs by @maximilianfridrich in baresip#2877
  • aufile: avoid race condition in case of fast destruction by @cspiel1 in baresip#2917
  • aufile: join thread if write fails by @cspiel1 in baresip#2922
  • video: add video_req_keyframe api by @sreimers in baresip#2920
  • call: start streams in sipsess_estab_handler by @maximilianfridrich in baresip#2909
  • webrtc: add av1 codec by @alfredh in baresip#2916
  • cmake: fix relative source dir find paths by @juha-h in baresip#2924
  • echo: fix re_snprintf pointer ARG by @sreimers in baresip#2927
  • cmake: Add include PATH so that GST is found also on Debian 11 by @juha-h in baresip#2928
  • call: improve glare handling by @maximilianfridrich in baresip#2929
  • call: set estdir in call_set_media_direction by @maximilianfridrich in baresip#2940
  • audio,aur: start audio player after early-video by @cspiel1 in baresip#2941
  • ctrl_dbus: add busctl example to module documentation by @maximilianfridrich in baresip#2944

Full Changelog: https://github.com/baresip/baresip/compare/v3.9.0...v3.10.0

3.9.0 - 2024-01-31

What's Changed

  • menu autoanswer handling by @cspiel1 in baresip#2832
  • aureceiver: fix overflow multiplications by @sreimers in baresip#2851
  • aur: entirely use mbuf in aurecv_debug() by @cspiel1 in baresip#2852
  • test: call - add AUDIO_MODE_THREAD to test_call_aufilt by @cspiel1 in baresip#2853
  • cmake: add only non-system link paths to rpath (fixes #2849) by @robert-scheck in baresip#2850
  • Renamed gzrtp ARRAY_SIZE macro by @juha-h in baresip#2855
  • avcapture: fix deprecated AVCaptureDeviceTypeExternalUnknown by @sreimers in baresip#2854
  • aur: fix uninitialized warning in auplay handler by @cspiel1 in baresip#2857
  • magic: use assert() instead of BREAKPOINT by @alfredh in baresip#2847
  • sipsess: refactor and simplify SDP negotiation state by @maximilianfridrich in baresip#2818
  • aur: set audio format correctly by @cspiel1 in baresip#2859
  • misc: bump year by @sreimers in baresip#2860
  • video: use viddec_packet by @sreimers in baresip#2861
  • audio: solve concurrency failures of TX thread by @cspiel1 in baresip#2862
  • aur: a mutex for aubuf allocation by @cspiel1 in baresip#2867
  • call: remove unused error handling of some API functions by @cspiel1 in baresip#2870
  • menu: an incoming call should not change the current call by @cspiel1 in baresip#2869
  • misc: HAVE_INET6 is always defined by @alfredh in baresip#2872
  • mqtt: improve disconnect reconnect handling by @sreimers in baresip#2866
  • misc: rx thread activate by @cspiel1 in baresip#2828
  • cmake: refactor module detection by @sreimers in baresip#2875
  • ua: improve SIP 404 warning by @cspiel1 in baresip#2871
  • call: fix race condition for call_set_video_dir() by @cspiel1 in baresip#2876
  • message: allow SIP MESSAGE with application/json ctype by @maximilianfridrich in baresip#2878
  • account/debug: fix wrong and redundant rtcp_mux printf by @sreimers in baresip#2892
  • mk: bump version to 3.9.0 by @alfredh in baresip#2894

Full Changelog: https://github.com/baresip/baresip/compare/v3.8.0...v3.9.0

3.8.1 - 2024-01-02

What's Changed

  • aur: set audio format correctly (#2859)
  • cmake: add only non-system link paths to rpath (fixes #2849) (#2850)

Full Changelog: https://github.com/baresip/baresip/compare/v3.8.0...v3.8.1

3.8.0 - 2023-12-27

What's Changed

  • uag: fallback for registrar-less NAT setups by @cspiel1 in baresip#2810
  • menu: fix outgoing early media limit by @cspiel1 in baresip#2817
  • menu: add follow up invite timer by @cspiel1 in baresip#2812
  • rx thread: refactoring part 2 - separate audio receiver by @cspiel1 in baresip#2795
  • pulse: log underruns/overruns after stream terminated by @cspiel1 in baresip#2820
  • call: add call_transp getter by @maximilianfridrich in baresip#2821
  • Decode url in custom header by @nltd101 in baresip#2816
  • video: fix thread sanitizer warning by @cspiel1 in baresip#2826
  • call/rtprecv: fix doxygen comments by @alfredh in baresip#2825
  • uag: use catchall flag for fallback UA selection by @cspiel1 in baresip#2827
  • audio,aur: audio_set_player() only stops auplay by @cspiel1 in baresip#2830
  • cmake: Fix rpath on MacOS by @larsimmisch in baresip#2831
  • cmake/plc: use system include to hide third party warnings by @sreimers in baresip#2836
  • cmake: add link paths to rpath by @sreimers in baresip#2839
  • cmake: bump minimum to 3.14 by @alfredh in baresip#2838
  • readme: update supported OpenSSL/LibreSSL versions by @robert-scheck in baresip#2843
  • cmake: add RE_LIBS config and add atomic check by @sreimers in baresip#2834
  • readme: update supported compiler versions by @robert-scheck in baresip#2844
  • ci: use actions/checkout@v4 by @robert-scheck in baresip#2845

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v3.7.0...v3.8.0

3.7.0 - 2023-11-22

What's Changed

  • Add UA_EVENT_END_OF_FILE by @larsimmisch in baresip#2755
  • test: call - add test_call_100rel_video by @cspiel1 in baresip#2762
  • call: delay for the initial re-invite after call established by @cspiel1 in baresip#2764
  • Implement OPTIONS ping by @maximilianfridrich in baresip#2765
  • test: call - improve tests for call progress by @cspiel1 in baresip#2770
  • call,event: add CALL_HOLD and CALL_RESUME events and fix call resume requests by @cspiel1 in baresip#2771
  • stream: extract thread safe RTP receiver by @cspiel1 in baresip#2685
  • test: call - count audio frames by @cspiel1 in baresip#2776
  • main: add pre-proc switch avoids warning by @cspiel1 in baresip#2778
  • rtprecv: fix possible rtprecv_metric null pointer deref by @juha-h in baresip#2786
  • rtprecv: add NULL pointer checks by @cspiel1 in baresip#2787
  • test: call - add call on-hold/resume test by @cspiel1 in baresip#2775
  • test: activate RTP stats by @cspiel1 in baresip#2789
  • test: call - more stable test_call_change_videodir by @cspiel1 in baresip#2790
  • test: call - add 100rel test for audio by @cspiel1 in baresip#2779
  • test: call - wait for ACK after SDP answer by @cspiel1 in baresip#2792
  • test: call - remove unstable check by @cspiel1 in baresip#2794
  • Debian version upgrade by @juha-h in baresip#2796
  • cmake/modules: exclude ctrl_dbus from Darwin/macOS by @sreimers in baresip#2798
  • ci: use macos-latest by @alfredh in baresip#2799
  • config: fix/split config_print arg lengths by @sreimers in baresip#2801
  • test/call.c: extend test_call_hold_resume by @maximilianfridrich in baresip#2800
  • rtpstat: fix stream_metric_get_rx_n_err stream arg by @sreimers in baresip#2803
  • test/ua: fix reg_dns size_t format by @sreimers in baresip#2804
  • test/main: fix unused i if HAVE_GETOPT is not available by @sreimers in baresip#2805
  • audio: fix inbound dtmf END event by @cspiel1 in baresip#2802
  • video: check vidcodec argument in video_decoder_set() by @alfredh in baresip#2806
  • gtk: close GTK on unsupported icon by @mbattista in baresip#2808
  • stream: lock tx.pt_enc fixes sanitizer warning by @cspiel1 in baresip#2809
  • avcodec: fix FFmpeg 6.1 AVframe key_frame deprecation by @sreimers in baresip#2807
  • jbuf: fix memory leak in jbuf_debug() by @cspiel1 in baresip#2813
  • jbuf: add NULL pointer check for mbuf by @cspiel1 in baresip#2814

New Contributors

  • @larsimmisch made their first contribution in baresip#2755

Full Changelog: https://github.com/baresip/baresip/compare/v3.6.0...v3.7.0

3.6.0 - 2023-10-17

What's Changed

  • test: call - replace stop_on_audio_video by cancel rule by @cspiel1 in baresip#2701
  • video: use const struct video for videnc_update_h and viddec_update_h by @sreimers in baresip#2670
  • misc: fd_listen fhs alloc rewrite by @sreimers in baresip#2688
  • ctrl_tcp: fix netstring enum warning by @sreimers in baresip#2730
  • ua, static_menu: Fix 100rel cmd by @maximilianfridrich in baresip#2731
  • tools: jbuf plots by @cspiel1 in baresip#2733
  • tools: fix and cleanup ajb plots by @cspiel1 in baresip#2736
  • ua: move adding of norefersub extension to create_register_clients by @maximilianfridrich in baresip#2734
  • main: add re_trace.json if enabled by @sreimers in baresip#2738
  • jbuf: move from re to baresip by @cspiel1 in baresip#2743
  • avcodec/decode: refactor hw_frame handling by @sreimers in baresip#2720
  • call: include Referred-by: tag in REFERs by @rodrigodeppe in baresip#2739
  • ci: bump [email protected] by @sreimers in baresip#2746
  • video: add video decode error trace by @sreimers in baresip#2748
  • video: protect shared resources in video_debug by @paresy in baresip#2747
  • video: delay video_destructor by @sreimers in baresip#2751
  • avcodec/decode: revert hw_frame handling and fix unref frame by @sreimers in baresip#2752
  • avcodec/decode: fix last av_frame memory leak by @sreimers in baresip#2753
  • test: call - count video frames in videodir tests by @cspiel1 in baresip#2758
  • test call - fix logical and by @cspiel1 in baresip#2763

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v3.5.1...v3.6.0

3.5.1 - 2023-09-12

What's Changed

Full Changelog: https://github.com/baresip/baresip/compare/v3.5.0...v3.5.1

3.5.0 - 2023-09-12

What's Changed

  • mc: fix format string by @cspiel1 in baresip#2675
  • call: never set sent_answer to false by @cspiel1 in baresip#2674
  • video: add source and display name getters by @sreimers in baresip#2669
  • cmake: fix clang gnu-zero-variadic-macro-arguments warning by @sreimers in baresip#2677
  • test call cancel rules by @cspiel1 in baresip#2667
  • stream: declare ext_len when assigned by @alfredh in baresip#2678
  • ci/mingw: remove cmake workaround by @sreimers in baresip#2679
  • call: fix Refer-To URI angle brackets by @sreimers in baresip#2681
  • test: call - replace re_cancel in CALL_RTCP and REMOTE_SDP by rule by @cspiel1 in baresip#2684
  • test: add rtcp_mux test by @cspiel1 in baresip#2692
  • test: remove unused local variable in test_call_bundle_base() by @cspiel1 in baresip#2693
  • account,docs: cleanup for accounts config by @cspiel1 in baresip#2691
  • test: call - combine cancel rules with logical AND by @cspiel1 in baresip#2687
  • ccheck: add PRI*64 check (use %L instead) by @sreimers in baresip#2695
  • test: call - replace stop_on_rtp by cancel rules by @cspiel1 in baresip#2697
  • webrtc/js: add rtcpMuxPolicy require policy by @sreimers in baresip#2699
  • gst: gst_deinit() should be last gst call by @cspiel1 in baresip#2703

Full Changelog: https://github.com/baresip/baresip/compare/v3.4.0...v3.5.0

3.4.0 - 2023-08-09

What's Changed

  • tools: add adaptive aubuf plot generation by @cspiel1 in baresip#2641
  • webrtc: add media track sdp direction by @sreimers in baresip#2636
  • webrtc: remove G711 module by @alfredh in baresip#2643
  • cmake: fix default path in FindAMR.cmake by @cspiel1 in baresip#2644
  • cmake: FindPNG.cmake - correct else path for PNG_FOUND by @cspiel1 in baresip#2645
  • webrtc: fix format and minor improvements by @alfredh in baresip#2647
  • cmake: add /usr/lib{64,}/glib-2.0 to FindGST.cmake by @robert-scheck in baresip#2648
  • ci/mingw: downgrade cmake by @sreimers in baresip#2656
  • call: logic fix in call_modify. remove call->sent_answer = false by @cHuberCoffee in baresip#2655
  • cmake/modules: fix ffmpeg static linking order by @sreimers in baresip#2659
  • webrtc: use 640x480 resolution for both sending and receiving by @alfredh in baresip#2658
  • readme: update list of RFCs by @alfredh in baresip#2657
  • test/cmake: list source files in alphabetical order by @alfredh in baresip#2661
  • test: fix typo in call test by @cspiel1 in baresip#2666
  • video,stream: stop natpinhole timer on re-invite by @cspiel1 in baresip#2665

Full Changelog: https://github.com/baresip/baresip/compare/v3.3.0...v3.4.0

3.3.0 - 2023-07-05

What's Changed

  • Removed unused zrtp_hash config setting by @juha-h in baresip#2591
  • audio: remove obsolete aurx fields by @cspiel1 in baresip#2590
  • mixausrc: sweep fading for performance by @cHuberCoffee in baresip#2605
  • config: clean up config template by @alfredh in baresip#2609
  • Handle UA_EVENT_CALL_REDIRECT (formerly UA_EVENT_CALL_BLIND_TRANSFER)~ by @juha-h in baresip#2602
  • ci/mingw: use cv2pdb for debug info conversion by @sreimers in baresip#2610
  • config: some refactoring by @alfredh in baresip#2611
  • config: missing updates for config example and template by @cspiel1 in baresip#2613
  • Fix oneway video sdp on answering call by @maximilianfridrich in baresip#2615
  • call,aucodec: add aucodec_print to internal core API by @cspiel1 in baresip#2616
  • account,stream: natpinhole account parameter orthogonal to medianat by @cspiel1 in baresip#2618
  • docs: update default value for natpinhole by @cspiel1 in baresip#2619
  • stream: increase udp socket buffer for video by @sreimers in baresip#2620
  • stream: faster video jitter buffer offloading by @sreimers in baresip#2622
  • config: add default_audio_path() by @alfredh in baresip#2621
  • event: correct class name for SDP events by @cspiel1 in baresip#2625
  • call: send LOCAL_SDP event if we send SDP by @cspiel1 in baresip#2626
  • config: refactor sip_cafile to make the code more clean by @alfredh in baresip#2627
  • audio: rework on codec changes by @cspiel1 in baresip#2630
  • event: add local SDP direction by @cspiel1 in baresip#2629
  • account: set sip_auroredirect default to no by @cspiel1 in baresip#2632
  • stream: also decode audio as long as jbuf returns EAGAIN by @cspiel1 in baresip#2635
  • aufile: init sampv buffer; NULL pointer check by @cspiel1 in baresip#2637

Full Changelog: https://github.com/baresip/baresip/compare/v3.2.0...v3.3.0

3.2.0 - 2023-05-31

What's Changed

  • 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

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v3.1.0...v3.2.0

3.1.0 - 2023-04-27

What's Changed

  • 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


3.0.0 - 2023-03-20

What's Changed

  • 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

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v2.12.0...v3.0.0


2.12.0 - 2023-02-15

What's Changed

  • 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


2.11.0 - 2023-01-11

What's Changed

  • 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


[2.10.0] - 2022-12-06

What's Changed

  • 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

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v2.9.0...v2.10.0


[2.9.0] - 2022-11-01

What's Changed

  • 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

New Contributors

Full Changelog: https://github.com/baresip/baresip/compare/v2.8.1...v2.9.0

[2.8.1] - 2022-10-01

  • baresip.h: bump BARESIP_VERSION by @cspiel1 in baresip#2196

[2.8.0] - 2022-10-01

  • 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

New Contributors

  • @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


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

New Contributors

  • @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

New Contributors


2.0.1 - 2022-03-27

What's Changed

  • 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

What's Changed

  • 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

New Contributors


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

Contributors (many thanks)


1.0.0 - 2020-09-11

Added

  • 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

Changed

  • 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

Fixed

  • 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

Removed

  • 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

Contributors (many thanks)

  • 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