# Baresip Changelog All notable changes to baresip will be documented in this file. The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/), and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html). ## 3.11.0 - 2024-04-09 ### What's Changed * account: read catchall flag from accounts file by @cspiel1 in https://github.com/baresip/baresip/pull/2925 * vp8/encode: optimizations and target_bitrate fix by @sreimers in https://github.com/baresip/baresip/pull/2936 * vp8,vp9: fix deprecated decode codec init by @sreimers in https://github.com/baresip/baresip/pull/2952 * aureceiver: fix mtx_unlock on discard by @sreimers in https://github.com/baresip/baresip/pull/2955 * release v3.10.1 by @sreimers in https://github.com/baresip/baresip/pull/2958 * message: return 403 instead of 488 by @maximilianfridrich in https://github.com/baresip/baresip/pull/2953 * netroam/cmake: add optional netlink detection by @sreimers in https://github.com/baresip/baresip/pull/2960 * ci/sanitizers: add mmap rnd_bits workaround by @sreimers in https://github.com/baresip/baresip/pull/2967 * account: set inreq_allowed=yes as default by @maximilianfridrich in https://github.com/baresip/baresip/pull/2961 * account: use correct format %zu for printing outbound by @maximilianfridrich in https://github.com/baresip/baresip/pull/2963 * stream: fix empty rtcp_stats for rtx.ssrc reception reports by @sreimers in https://github.com/baresip/baresip/pull/2969 * stream: avoid sanitizer warnings for strm->tx by @cspiel1 in https://github.com/baresip/baresip/pull/2949 * avcodec: remove re_h264 extra header by @sreimers in https://github.com/baresip/baresip/pull/2971 * play: err handling and ensure eof by @cspiel1 in https://github.com/baresip/baresip/pull/2972 * stream: add stream_jbuf_stats() by @sreimers in https://github.com/baresip/baresip/pull/2973 * sndfile: write correct sample rate to WAV header by @cspiel1 in https://github.com/baresip/baresip/pull/2976 * tls: add session resumption setter by @maximilianfridrich in https://github.com/baresip/baresip/pull/2977 * avcodec: use util function to decode H.264 STAP-A by @alfredh in https://github.com/baresip/baresip/pull/2978 * mixausrc: fix ausrc resampling by @cspiel1 in https://github.com/baresip/baresip/pull/2981 * ci/build: remove obsolete for loop by @cspiel1 in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2955 ## 3.10.0 - 2024-03-06 ## What's Changed * cmake: use default value for CMAKE_C_EXTENSIONS by @sreimers in https://github.com/baresip/baresip/pull/2893 * cmake: add /usr/{local,}/include/re and /usr/{local,}/lib{64,} to FindRE.cmake by @robert-scheck in https://github.com/baresip/baresip/pull/2900 * test/main: fix NULL pointer arg on err by @sreimers in https://github.com/baresip/baresip/pull/2902 * ci: add Fedora workflow to avoid e.g. rpath issues by @robert-scheck in https://github.com/baresip/baresip/pull/2904 * mediatrack/start: add audio_decoder_set by @sreimers in https://github.com/baresip/baresip/pull/2910 * config: support distribution-specific/default CA paths by @robert-scheck in https://github.com/baresip/baresip/pull/2905 * readme: cosmetic changes by @robert-scheck in https://github.com/baresip/baresip/pull/2911 * ci/fedora: fix dependency by @sreimers in https://github.com/baresip/baresip/pull/2912 * config: add default CA path for Android by @robert-scheck in https://github.com/baresip/baresip/pull/2913 * transp,tls: add TLS client verification by @maximilianfridrich in https://github.com/baresip/baresip/pull/2888 * account,message,ua: secure incoming SIP MESSAGEs by @maximilianfridrich in https://github.com/baresip/baresip/pull/2877 * aufile: avoid race condition in case of fast destruction by @cspiel1 in https://github.com/baresip/baresip/pull/2917 * aufile: join thread if write fails by @cspiel1 in https://github.com/baresip/baresip/pull/2922 * video: add video_req_keyframe api by @sreimers in https://github.com/baresip/baresip/pull/2920 * call: start streams in sipsess_estab_handler by @maximilianfridrich in https://github.com/baresip/baresip/pull/2909 * webrtc: add av1 codec by @alfredh in https://github.com/baresip/baresip/pull/2916 * cmake: fix relative source dir find paths by @juha-h in https://github.com/baresip/baresip/pull/2924 * echo: fix re_snprintf pointer ARG by @sreimers in https://github.com/baresip/baresip/pull/2927 * cmake: Add include PATH so that GST is found also on Debian 11 by @juha-h in https://github.com/baresip/baresip/pull/2928 * call: improve glare handling by @maximilianfridrich in https://github.com/baresip/baresip/pull/2929 * call: set estdir in call_set_media_direction by @maximilianfridrich in https://github.com/baresip/baresip/pull/2940 * audio,aur: start audio player after early-video by @cspiel1 in https://github.com/baresip/baresip/pull/2941 * ctrl_dbus: add busctl example to module documentation by @maximilianfridrich in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2832 * aureceiver: fix overflow multiplications by @sreimers in https://github.com/baresip/baresip/pull/2851 * aur: entirely use mbuf in aurecv_debug() by @cspiel1 in https://github.com/baresip/baresip/pull/2852 * test: call - add AUDIO_MODE_THREAD to test_call_aufilt by @cspiel1 in https://github.com/baresip/baresip/pull/2853 * cmake: add only non-system link paths to rpath (fixes #2849) by @robert-scheck in https://github.com/baresip/baresip/pull/2850 * Renamed gzrtp ARRAY_SIZE macro by @juha-h in https://github.com/baresip/baresip/pull/2855 * avcapture: fix deprecated AVCaptureDeviceTypeExternalUnknown by @sreimers in https://github.com/baresip/baresip/pull/2854 * aur: fix uninitialized warning in auplay handler by @cspiel1 in https://github.com/baresip/baresip/pull/2857 * magic: use assert() instead of BREAKPOINT by @alfredh in https://github.com/baresip/baresip/pull/2847 * sipsess: refactor and simplify SDP negotiation state by @maximilianfridrich in https://github.com/baresip/baresip/pull/2818 * aur: set audio format correctly by @cspiel1 in https://github.com/baresip/baresip/pull/2859 * misc: bump year by @sreimers in https://github.com/baresip/baresip/pull/2860 * video: use viddec_packet by @sreimers in https://github.com/baresip/baresip/pull/2861 * audio: solve concurrency failures of TX thread by @cspiel1 in https://github.com/baresip/baresip/pull/2862 * aur: a mutex for aubuf allocation by @cspiel1 in https://github.com/baresip/baresip/pull/2867 * call: remove unused error handling of some API functions by @cspiel1 in https://github.com/baresip/baresip/pull/2870 * menu: an incoming call should not change the current call by @cspiel1 in https://github.com/baresip/baresip/pull/2869 * misc: HAVE_INET6 is always defined by @alfredh in https://github.com/baresip/baresip/pull/2872 * mqtt: improve disconnect reconnect handling by @sreimers in https://github.com/baresip/baresip/pull/2866 * misc: rx thread activate by @cspiel1 in https://github.com/baresip/baresip/pull/2828 * cmake: refactor module detection by @sreimers in https://github.com/baresip/baresip/pull/2875 * ua: improve SIP 404 warning by @cspiel1 in https://github.com/baresip/baresip/pull/2871 * call: fix race condition for call_set_video_dir() by @cspiel1 in https://github.com/baresip/baresip/pull/2876 * message: allow SIP MESSAGE with application/json ctype by @maximilianfridrich in https://github.com/baresip/baresip/pull/2878 * account/debug: fix wrong and redundant rtcp_mux printf by @sreimers in https://github.com/baresip/baresip/pull/2892 * mk: bump version to 3.9.0 by @alfredh in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2810 * menu: fix outgoing early media limit by @cspiel1 in https://github.com/baresip/baresip/pull/2817 * menu: add follow up invite timer by @cspiel1 in https://github.com/baresip/baresip/pull/2812 * rx thread: refactoring part 2 - separate audio receiver by @cspiel1 in https://github.com/baresip/baresip/pull/2795 * pulse: log underruns/overruns after stream terminated by @cspiel1 in https://github.com/baresip/baresip/pull/2820 * call: add call_transp getter by @maximilianfridrich in https://github.com/baresip/baresip/pull/2821 * Decode url in custom header by @nltd101 in https://github.com/baresip/baresip/pull/2816 * video: fix thread sanitizer warning by @cspiel1 in https://github.com/baresip/baresip/pull/2826 * call/rtprecv: fix doxygen comments by @alfredh in https://github.com/baresip/baresip/pull/2825 * uag: use catchall flag for fallback UA selection by @cspiel1 in https://github.com/baresip/baresip/pull/2827 * audio,aur: audio_set_player() only stops auplay by @cspiel1 in https://github.com/baresip/baresip/pull/2830 * cmake: Fix rpath on MacOS by @larsimmisch in https://github.com/baresip/baresip/pull/2831 * cmake/plc: use system include to hide third party warnings by @sreimers in https://github.com/baresip/baresip/pull/2836 * cmake: add link paths to rpath by @sreimers in https://github.com/baresip/baresip/pull/2839 * cmake: bump minimum to 3.14 by @alfredh in https://github.com/baresip/baresip/pull/2838 * readme: update supported OpenSSL/LibreSSL versions by @robert-scheck in https://github.com/baresip/baresip/pull/2843 * cmake: add RE_LIBS config and add atomic check by @sreimers in https://github.com/baresip/baresip/pull/2834 * readme: update supported compiler versions by @robert-scheck in https://github.com/baresip/baresip/pull/2844 * ci: use actions/checkout@v4 by @robert-scheck in https://github.com/baresip/baresip/pull/2845 ## New Contributors * @nltd101 made their first contribution in https://github.com/baresip/baresip/pull/2816 **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 https://github.com/baresip/baresip/pull/2755 * test: call - add test_call_100rel_video by @cspiel1 in https://github.com/baresip/baresip/pull/2762 * call: delay for the initial re-invite after call established by @cspiel1 in https://github.com/baresip/baresip/pull/2764 * Implement OPTIONS ping by @maximilianfridrich in https://github.com/baresip/baresip/pull/2765 * test: call - improve tests for call progress by @cspiel1 in https://github.com/baresip/baresip/pull/2770 * call,event: add CALL_HOLD and CALL_RESUME events and fix call resume requests by @cspiel1 in https://github.com/baresip/baresip/pull/2771 * stream: extract thread safe RTP receiver by @cspiel1 in https://github.com/baresip/baresip/pull/2685 * test: call - count audio frames by @cspiel1 in https://github.com/baresip/baresip/pull/2776 * main: add pre-proc switch avoids warning by @cspiel1 in https://github.com/baresip/baresip/pull/2778 * rtprecv: fix possible rtprecv_metric null pointer deref by @juha-h in https://github.com/baresip/baresip/pull/2786 * rtprecv: add NULL pointer checks by @cspiel1 in https://github.com/baresip/baresip/pull/2787 * test: call - add call on-hold/resume test by @cspiel1 in https://github.com/baresip/baresip/pull/2775 * test: activate RTP stats by @cspiel1 in https://github.com/baresip/baresip/pull/2789 * test: call - more stable test_call_change_videodir by @cspiel1 in https://github.com/baresip/baresip/pull/2790 * test: call - add 100rel test for audio by @cspiel1 in https://github.com/baresip/baresip/pull/2779 * test: call - wait for ACK after SDP answer by @cspiel1 in https://github.com/baresip/baresip/pull/2792 * test: call - remove unstable check by @cspiel1 in https://github.com/baresip/baresip/pull/2794 * Debian version upgrade by @juha-h in https://github.com/baresip/baresip/pull/2796 * cmake/modules: exclude ctrl_dbus from Darwin/macOS by @sreimers in https://github.com/baresip/baresip/pull/2798 * ci: use macos-latest by @alfredh in https://github.com/baresip/baresip/pull/2799 * config: fix/split config_print arg lengths by @sreimers in https://github.com/baresip/baresip/pull/2801 * test/call.c: extend test_call_hold_resume by @maximilianfridrich in https://github.com/baresip/baresip/pull/2800 * rtpstat: fix stream_metric_get_rx_n_err stream arg by @sreimers in https://github.com/baresip/baresip/pull/2803 * test/ua: fix reg_dns size_t format by @sreimers in https://github.com/baresip/baresip/pull/2804 * test/main: fix unused i if HAVE_GETOPT is not available by @sreimers in https://github.com/baresip/baresip/pull/2805 * audio: fix inbound dtmf END event by @cspiel1 in https://github.com/baresip/baresip/pull/2802 * video: check vidcodec argument in video_decoder_set() by @alfredh in https://github.com/baresip/baresip/pull/2806 * gtk: close GTK on unsupported icon by @mbattista in https://github.com/baresip/baresip/pull/2808 * stream: lock tx.pt_enc fixes sanitizer warning by @cspiel1 in https://github.com/baresip/baresip/pull/2809 * avcodec: fix FFmpeg 6.1 AVframe key_frame deprecation by @sreimers in https://github.com/baresip/baresip/pull/2807 * jbuf: fix memory leak in jbuf_debug() by @cspiel1 in https://github.com/baresip/baresip/pull/2813 * jbuf: add NULL pointer check for mbuf by @cspiel1 in https://github.com/baresip/baresip/pull/2814 ## New Contributors * @larsimmisch made their first contribution in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2701 * video: use const struct video for videnc_update_h and viddec_update_h by @sreimers in https://github.com/baresip/baresip/pull/2670 * misc: fd_listen fhs alloc rewrite by @sreimers in https://github.com/baresip/baresip/pull/2688 * ctrl_tcp: fix netstring enum warning by @sreimers in https://github.com/baresip/baresip/pull/2730 * ua, static_menu: Fix 100rel cmd by @maximilianfridrich in https://github.com/baresip/baresip/pull/2731 * tools: jbuf plots by @cspiel1 in https://github.com/baresip/baresip/pull/2733 * tools: fix and cleanup ajb plots by @cspiel1 in https://github.com/baresip/baresip/pull/2736 * ua: move adding of norefersub extension to create_register_clients by @maximilianfridrich in https://github.com/baresip/baresip/pull/2734 * main: add re_trace.json if enabled by @sreimers in https://github.com/baresip/baresip/pull/2738 * jbuf: move from re to baresip by @cspiel1 in https://github.com/baresip/baresip/pull/2743 * avcodec/decode: refactor hw_frame handling by @sreimers in https://github.com/baresip/baresip/pull/2720 * call: include Referred-by: tag in REFERs by @rodrigodeppe in https://github.com/baresip/baresip/pull/2739 * ci: bump pr-dependency-action@v0.6 by @sreimers in https://github.com/baresip/baresip/pull/2746 * video: add video decode error trace by @sreimers in https://github.com/baresip/baresip/pull/2748 * video: protect shared resources in video_debug by @paresy in https://github.com/baresip/baresip/pull/2747 * video: delay video_destructor by @sreimers in https://github.com/baresip/baresip/pull/2751 * avcodec/decode: revert hw_frame handling and fix unref frame by @sreimers in https://github.com/baresip/baresip/pull/2752 * avcodec/decode: fix last av_frame memory leak by @sreimers in https://github.com/baresip/baresip/pull/2753 * test: call - count video frames in videodir tests by @cspiel1 in https://github.com/baresip/baresip/pull/2758 * test call - fix logical and by @cspiel1 in https://github.com/baresip/baresip/pull/2763 ## New Contributors * @rodrigodeppe made their first contribution in https://github.com/baresip/baresip/pull/2739 * @paresy made their first contribution in https://github.com/baresip/baresip/pull/2747 **Full Changelog**: https://github.com/baresip/baresip/compare/v3.5.1...v3.6.0 ## 3.5.1 - 2023-09-12 ## What's Changed * cmake: fix RE_DEFINITIONS by @sreimers in https://github.com/baresip/baresip/pull/2716 **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 https://github.com/baresip/baresip/pull/2675 * call: never set sent_answer to false by @cspiel1 in https://github.com/baresip/baresip/pull/2674 * video: add source and display name getters by @sreimers in https://github.com/baresip/baresip/pull/2669 * cmake: fix clang gnu-zero-variadic-macro-arguments warning by @sreimers in https://github.com/baresip/baresip/pull/2677 * test call cancel rules by @cspiel1 in https://github.com/baresip/baresip/pull/2667 * stream: declare ext_len when assigned by @alfredh in https://github.com/baresip/baresip/pull/2678 * ci/mingw: remove cmake workaround by @sreimers in https://github.com/baresip/baresip/pull/2679 * call: fix Refer-To URI angle brackets by @sreimers in https://github.com/baresip/baresip/pull/2681 * test: call - replace re_cancel in CALL_RTCP and REMOTE_SDP by rule by @cspiel1 in https://github.com/baresip/baresip/pull/2684 * test: add rtcp_mux test by @cspiel1 in https://github.com/baresip/baresip/pull/2692 * test: remove unused local variable in test_call_bundle_base() by @cspiel1 in https://github.com/baresip/baresip/pull/2693 * account,docs: cleanup for accounts config by @cspiel1 in https://github.com/baresip/baresip/pull/2691 * test: call - combine cancel rules with logical AND by @cspiel1 in https://github.com/baresip/baresip/pull/2687 * ccheck: add PRI*64 check (use %L instead) by @sreimers in https://github.com/baresip/baresip/pull/2695 * test: call - replace stop_on_rtp by cancel rules by @cspiel1 in https://github.com/baresip/baresip/pull/2697 * webrtc/js: add rtcpMuxPolicy require policy by @sreimers in https://github.com/baresip/baresip/pull/2699 * gst: gst_deinit() should be last gst call by @cspiel1 in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2641 * webrtc: add media track sdp direction by @sreimers in https://github.com/baresip/baresip/pull/2636 * webrtc: remove G711 module by @alfredh in https://github.com/baresip/baresip/pull/2643 * cmake: fix default path in FindAMR.cmake by @cspiel1 in https://github.com/baresip/baresip/pull/2644 * cmake: FindPNG.cmake - correct else path for PNG_FOUND by @cspiel1 in https://github.com/baresip/baresip/pull/2645 * webrtc: fix format and minor improvements by @alfredh in https://github.com/baresip/baresip/pull/2647 * cmake: add /usr/lib{64,}/glib-2.0 to FindGST.cmake by @robert-scheck in https://github.com/baresip/baresip/pull/2648 * ci/mingw: downgrade cmake by @sreimers in https://github.com/baresip/baresip/pull/2656 * call: logic fix in call_modify. remove call->sent_answer = false by @cHuberCoffee in https://github.com/baresip/baresip/pull/2655 * cmake/modules: fix ffmpeg static linking order by @sreimers in https://github.com/baresip/baresip/pull/2659 * webrtc: use 640x480 resolution for both sending and receiving by @alfredh in https://github.com/baresip/baresip/pull/2658 * readme: update list of RFCs by @alfredh in https://github.com/baresip/baresip/pull/2657 * test/cmake: list source files in alphabetical order by @alfredh in https://github.com/baresip/baresip/pull/2661 * test: fix typo in call test by @cspiel1 in https://github.com/baresip/baresip/pull/2666 * video,stream: stop natpinhole timer on re-invite by @cspiel1 in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2591 * audio: remove obsolete aurx fields by @cspiel1 in https://github.com/baresip/baresip/pull/2590 * mixausrc: sweep fading for performance by @cHuberCoffee in https://github.com/baresip/baresip/pull/2605 * config: clean up config template by @alfredh in https://github.com/baresip/baresip/pull/2609 * Handle UA_EVENT_CALL_REDIRECT (formerly UA_EVENT_CALL_BLIND_TRANSFER)~ by @juha-h in https://github.com/baresip/baresip/pull/2602 * ci/mingw: use cv2pdb for debug info conversion by @sreimers in https://github.com/baresip/baresip/pull/2610 * config: some refactoring by @alfredh in https://github.com/baresip/baresip/pull/2611 * config: missing updates for config example and template by @cspiel1 in https://github.com/baresip/baresip/pull/2613 * Fix oneway video sdp on answering call by @maximilianfridrich in https://github.com/baresip/baresip/pull/2615 * call,aucodec: add aucodec_print to internal core API by @cspiel1 in https://github.com/baresip/baresip/pull/2616 * account,stream: natpinhole account parameter orthogonal to medianat by @cspiel1 in https://github.com/baresip/baresip/pull/2618 * docs: update default value for natpinhole by @cspiel1 in https://github.com/baresip/baresip/pull/2619 * stream: increase udp socket buffer for video by @sreimers in https://github.com/baresip/baresip/pull/2620 * stream: faster video jitter buffer offloading by @sreimers in https://github.com/baresip/baresip/pull/2622 * config: add default_audio_path() by @alfredh in https://github.com/baresip/baresip/pull/2621 * event: correct class name for SDP events by @cspiel1 in https://github.com/baresip/baresip/pull/2625 * call: send LOCAL_SDP event if we send SDP by @cspiel1 in https://github.com/baresip/baresip/pull/2626 * config: refactor sip_cafile to make the code more clean by @alfredh in https://github.com/baresip/baresip/pull/2627 * audio: rework on codec changes by @cspiel1 in https://github.com/baresip/baresip/pull/2630 * event: add local SDP direction by @cspiel1 in https://github.com/baresip/baresip/pull/2629 * account: set sip_auroredirect default to no by @cspiel1 in https://github.com/baresip/baresip/pull/2632 * stream: also decode audio as long as jbuf returns EAGAIN by @cspiel1 in https://github.com/baresip/baresip/pull/2635 * aufile: init sampv buffer; NULL pointer check by @cspiel1 in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2550 * cmake: fix win32 dbghelp by @sreimers in https://github.com/baresip/baresip/pull/2552 * cmake/gtk3: make sure gtk3 libs are found by @landryb in https://github.com/baresip/baresip/pull/2554 * pipewire: add pipewire module by @cspiel1 in https://github.com/baresip/baresip/pull/2439 * audio: count TX underruns correctly (#2535) by @cspiel1 in https://github.com/baresip/baresip/pull/2553 * variadic function fixes by @maximilianfridrich in https://github.com/baresip/baresip/pull/2523 * ua: unescape incoming Refer-To header by @maximilianfridrich in https://github.com/baresip/baresip/pull/2541 * pipewire/cmake: declare include dirs as system by @sreimers in https://github.com/baresip/baresip/pull/2556 * sndio: re-add sndio module for OpenBSD by @landryb in https://github.com/baresip/baresip/pull/2555 * Client cert renegotiation in http by @fAuernigg in https://github.com/baresip/baresip/pull/2461 * aufile: joind already terminated thread frees stack by @cspiel1 in https://github.com/baresip/baresip/pull/2557 * ccheck: c11 err handling exclude mutex\_alloc by @sreimers in https://github.com/baresip/baresip/pull/2560 * alsa: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2558 * Add instruction to build the doxygen docs by @gibix in https://github.com/baresip/baresip/pull/2561 * fakevideo: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2565 * cmake/findOpus: fix Could NOT find OPUS (missing: OPUS\_INCLUDE\_DIR) by @jobo-zt in https://github.com/baresip/baresip/pull/2568 * cmake/findsdl: fix Could NOT find SDL (missing: SDL\_INCLUDE\_DIR) by @jobo-zt in https://github.com/baresip/baresip/pull/2571 * avformat, g722: add macro UNISTD switch support for Windows by @jobo-zt in https://github.com/baresip/baresip/pull/2574 * v4l2: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2567 * mc: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2566 * avformat: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2564 * ausine: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2563 * aubridge: use atomic for run flag by @cspiel1 in https://github.com/baresip/baresip/pull/2562 * config: add different options for audio and video jitter buffer by @sreimers in https://github.com/baresip/baresip/pull/2569 * Revert "ua: unescape incoming Refer-To header (#2541)" by @maximilianfridrich in https://github.com/baresip/baresip/pull/2577 * stream: log last RTP packet debug after 100ms by @sreimers in https://github.com/baresip/baresip/pull/2587 * ci/sanitizers: exit on first undefined behavior by @sreimers in https://github.com/baresip/baresip/pull/2589 ## New Contributors * @landryb made their first contribution in https://github.com/baresip/baresip/pull/2554 * @gibix made their first contribution in https://github.com/baresip/baresip/pull/2561 * @jobo-zt made their first contribution in https://github.com/baresip/baresip/pull/2568 **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 https://github.com/baresip/baresip/pull/2490 * gzrtp: RX thread - safe stop by @cspiel1 in https://github.com/baresip/baresip/pull/2492 * ci: avoid hardcoded OpenSSL path on macOS by @robert-scheck in https://github.com/baresip/baresip/pull/2505 * fix cmake modules by @sreimers in https://github.com/baresip/baresip/pull/2507 * cmake/mqtt: fix MOSQUITTO_LIBRARY by @sreimers in https://github.com/baresip/baresip/pull/2508 * mc: send module event whenever receiver is stopped by @cspiel1 in https://github.com/baresip/baresip/pull/2509 * menu: limit early audio TX streams by @cspiel1 in https://github.com/baresip/baresip/pull/2503 * call: check if SIP UPDATE is allowed, but always update local media by @cspiel1 in https://github.com/baresip/baresip/pull/2504 * account: increase line handler size to 1024 characters by @juha-h in https://github.com/baresip/baresip/pull/2511 * cmake: avoid include of /usr/local/include by @cspiel1 in https://github.com/baresip/baresip/pull/2506 * call,audio: respect SDP media dir on audio start similar to video by @cspiel1 in https://github.com/baresip/baresip/pull/2501 * video: refactor paced and burst sending by @sreimers in https://github.com/baresip/baresip/pull/2482 * ctrl_dbus,ice,png_vf: Fix format string usage by @maximilianfridrich in https://github.com/baresip/baresip/pull/2517 * menu limit early video by @cspiel1 in https://github.com/baresip/baresip/pull/2514 * play: flush of the aubuf directly before the replay starts by @cspiel1 in https://github.com/baresip/baresip/pull/2512 * stream: fix setting of RTP tos for IPv6 by @cspiel1 in https://github.com/baresip/baresip/pull/2527 * call: only flush audio stream when stream starts by @cspiel1 in https://github.com/baresip/baresip/pull/2526 * menu: use busy tone when call declined (scode 603) by @cspiel1 in https://github.com/baresip/baresip/pull/2529 * ua: incoming DTMF key=0 should be reported as DTMF end by @cspiel1 in https://github.com/baresip/baresip/pull/2528 * video: fix possible 32bit overflow by @sreimers in https://github.com/baresip/baresip/pull/2534 * ua: deref call on reset_transp fail by @maximilianfridrich in https://github.com/baresip/baresip/pull/2532 * uag: avoid transport reset if local address has not changed by @juha-h in https://github.com/baresip/baresip/pull/2537 * ci: add gcc-12 for Ubuntu 22.04 (ubuntu-latest) by @robert-scheck in https://github.com/baresip/baresip/pull/2542 * docs: remove librem from README files by @robert-scheck in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2452 * merge rem into re by @alfredh in https://github.com/baresip/baresip/pull/2442 * main: fix async init order (after config load) by @sreimers in https://github.com/baresip/baresip/pull/2457 * ci: install pkg-config on mac-os by @cspiel1 in https://github.com/baresip/baresip/pull/2459 * ci: remove rem in sanitizers and valgrind yml by @cspiel1 in https://github.com/baresip/baresip/pull/2458 * video: fix vidqueue_poll list_move by @sreimers in https://github.com/baresip/baresip/pull/2465 * Dshow fixes by @tomek-o in https://github.com/baresip/baresip/pull/2467 * Moved adding of custom headers from ua_connect_dir to ua_call_alloc by @juha-h in https://github.com/baresip/baresip/pull/2470 * Include also params to MESSAGE URI by @juha-h in https://github.com/baresip/baresip/pull/2469 * video: remove unused qent->dst by @sreimers in https://github.com/baresip/baresip/pull/2474 * call: Fix delayed (auto) answer if awaiting PRACK by @maximilianfridrich in https://github.com/baresip/baresip/pull/2473 * video: add TX thread by @sreimers in https://github.com/baresip/baresip/pull/2460 * ccheck: add check_list_unlink check by @sreimers in https://github.com/baresip/baresip/pull/2471 * stream: add stream_enable_tx() api by @sreimers in https://github.com/baresip/baresip/pull/2479 * audio: align Audio TX thread name by @sreimers in https://github.com/baresip/baresip/pull/2480 * Send event when dump file is opened by @juha-h in https://github.com/baresip/baresip/pull/2486 * video: add NULL pointer check for vidisp by @cspiel1 in https://github.com/baresip/baresip/pull/2483 * ua: Fix calls of ua_event() by @maximilianfridrich in https://github.com/baresip/baresip/pull/2495 * call: Fix calls of call_event_handler by @maximilianfridrich in https://github.com/baresip/baresip/pull/2496 ## New Contributors * @Koshub made their first contribution in https://github.com/baresip/baresip/pull/2452 * @tomek-o made their first contribution in https://github.com/baresip/baresip/pull/2467 **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 https://github.com/baresip/baresip/pull/2409 * dtls_srtp: enable single DTLS connection mode by @alfredh in https://github.com/baresip/baresip/pull/2411 * ci: try to fix flaky azure mirrors by @sreimers in https://github.com/baresip/baresip/pull/2413 * cmake/pulse: Remove pulse-simple library lookup by @robert-scheck in https://github.com/baresip/baresip/pull/2414 * webrtc_aecm: use C11 mutex by @juha-h in https://github.com/baresip/baresip/pull/2415 * pulse: replace obsolete string pulse_async (makes baresip PipeWire compatible) by @cspiel1 in https://github.com/baresip/baresip/pull/2412 * vidpacket: add keyframe flag by @alfredh in https://github.com/baresip/baresip/pull/2416 * av1: use keyframe instead of new-flag by @alfredh in https://github.com/baresip/baresip/pull/2418 * av1: fix warnings by @alfredh in https://github.com/baresip/baresip/pull/2419 * make rtcp interval configureable by @sreimers in https://github.com/baresip/baresip/pull/2420 * sndio: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/2422 * PRACK refactoring by @maximilianfridrich in https://github.com/baresip/baresip/pull/2401 * ci: merge build and cmake by @alfredh in https://github.com/baresip/baresip/pull/2425 * menu: ringback/early audio handling for parallel calls by @cspiel1 in https://github.com/baresip/baresip/pull/2403 * magic: use C99 __func__ macro by @alfredh in https://github.com/baresip/baresip/pull/2427 * stream: remove stream_decode from internal API by @cspiel1 in https://github.com/baresip/baresip/pull/2430 * use RE_ARRAY_SIZE() macro by @alfredh in https://github.com/baresip/baresip/pull/2429 * cmake: link RESOLV_LIBRARY by @sreimers in https://github.com/baresip/baresip/pull/2432 * ci/build: fix Ubuntu 22.04 workaround by @sreimers in https://github.com/baresip/baresip/pull/2435 * avcapture: use RE_ARRAY_SIZE macro by @alfredh in https://github.com/baresip/baresip/pull/2434 * pulse: remove obsolete doxygen note to be experimental by @cspiel1 in https://github.com/baresip/baresip/pull/2436 * gtk: return NULL on mtx_init() != thrd_success by @robert-scheck in https://github.com/baresip/baresip/pull/2440 * ci: add libgtk-3-dev to build GTK+ 3 module by @robert-scheck in https://github.com/baresip/baresip/pull/2441 * event: missing class name case for RTPESTAB event by @cspiel1 in https://github.com/baresip/baresip/pull/2447 * ci: add sanitizers by @sreimers in https://github.com/baresip/baresip/pull/2449 * bump version numbers to 2.12.0 by @alfredh in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2345 * stream: send RTP NAT pinhole opener until RTP is received by @cspiel1 in https://github.com/baresip/baresip/pull/2346 * mediatrack: add audio and video getters by @sreimers in https://github.com/baresip/baresip/pull/2347 * Added rtcp_mux related API functions by @juha-h in https://github.com/baresip/baresip/pull/2352 * make: remove deprecated makefile by @alfredh in https://github.com/baresip/baresip/pull/2354 * Removed rtcp_mux config variable by @juha-h in https://github.com/baresip/baresip/pull/2353 * Use bool instead of "yes"/"no" in account API functions by @juha-h in https://github.com/baresip/baresip/pull/2355 * aubuf: add AUBUF_FILE mode by @cspiel1 in https://github.com/baresip/baresip/pull/2363 * play: flush aubuf before restart by @cspiel1 in https://github.com/baresip/baresip/pull/2364 * call: avoid unwanted re-invites on ESTABLISHED event by @cspiel1 in https://github.com/baresip/baresip/pull/2362 * avcodec: constrain bitrate by @sreimers in https://github.com/baresip/baresip/pull/2365 * pulse: rename to pulse_simple.so by @alfredh in https://github.com/baresip/baresip/pull/2371 * module: remove module_tmp by @alfredh in https://github.com/baresip/baresip/pull/2373 * audio: remove unused last_sampc by @alfredh in https://github.com/baresip/baresip/pull/2372 * audio: add rtpext_find() (refactoring) by @alfredh in https://github.com/baresip/baresip/pull/2375 * multicast: remove ref to pthread by @alfredh in https://github.com/baresip/baresip/pull/2379 * video: add rtcp-fb Generic NACK by @sreimers in https://github.com/baresip/baresip/pull/2380 * call: set media dir also for MNAT case by @cspiel1 in https://github.com/baresip/baresip/pull/2382 * pulse: rename pulse_async.so to pulse.so (default) by @alfredh in https://github.com/baresip/baresip/pull/2381 * RTP Resend by @sreimers in https://github.com/baresip/baresip/pull/2378 * make: remove unused srcs.mk by @alfredh in https://github.com/baresip/baresip/pull/2387 * TLS server support SNI based certificate selection by @cspiel1 in https://github.com/baresip/baresip/pull/2330 * audiounit: use C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/2386 * webrtc_aec: use C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/2384 * coreaudio: use C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/2388 * gtk: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2391 * remove pulse_simple.so -- use pulse.so by @alfredh in https://github.com/baresip/baresip/pull/2383 * ci: rename ccheck.yml to lint.yml by @alfredh in https://github.com/baresip/baresip/pull/2394 * fritzbox2baresip: use open with explicitly specifying an encoding by @robert-scheck in https://github.com/baresip/baresip/pull/2396 * test: remove mock_aufilt (unused) by @alfredh in https://github.com/baresip/baresip/pull/2392 * Ci pylint by @alfredh in https://github.com/baresip/baresip/pull/2395 * gzrtp: use C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/2393 * C11 mutex by @mbattista in https://github.com/baresip/baresip/pull/2397 * tls multiple server certs by @cspiel1 in https://github.com/baresip/baresip/pull/2399 * call: return EINVAL if answer not possible by @maximilianfridrich in https://github.com/baresip/baresip/pull/2405 * ccheck: fix some pylint warnings by @alfredh in https://github.com/baresip/baresip/pull/2398 * Fixed account debug of mwi and call_transfer by @juha-h in https://github.com/baresip/baresip/pull/2406 * avformat: fix printf format for samplerate and channels by @alfredh in https://github.com/baresip/baresip/pull/2407 * cmake: increase minimum required version by @cspiel1 in https://github.com/baresip/baresip/pull/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 https://github.com/baresip/baresip/pull/2285 * vidinfo: allow all pixel formats by @alfredh in https://github.com/baresip/baresip/pull/2291 * vid: add support for YUV422P pixel format by @alfredh in https://github.com/baresip/baresip/pull/2280 * avformat: fix hwaccel vaapi by @alfredh in https://github.com/baresip/baresip/pull/2299 * mk: add deprecate notice by @alfredh in https://github.com/baresip/baresip/pull/2302 * mingw: upgrade to OpenSSL 3.0.7 by @alfredh in https://github.com/baresip/baresip/pull/2303 * dshow: fix some warnings by @alfredh in https://github.com/baresip/baresip/pull/2305 * dshow: fix pragma warning by @alfredh in https://github.com/baresip/baresip/pull/2306 * ci: install libsdl2 development package by @alfredh in https://github.com/baresip/baresip/pull/2307 * sdl: work in progress fixes for multi-threading by @alfredh in https://github.com/baresip/baresip/pull/2300 * Stop segfaulting when no URI is passed to dial command by @SimonHyde-BBC in https://github.com/baresip/baresip/pull/2311 * ice: local candidate policy config by @sreimers in https://github.com/baresip/baresip/pull/2312 * auresamp: check handler arguments by @alfredh in https://github.com/baresip/baresip/pull/2313 * fixes 2315 and GTK errors on quit by @mbattista in https://github.com/baresip/baresip/pull/2316 * auresamp: avoid division by zero (#2293) by @cspiel1 in https://github.com/baresip/baresip/pull/2317 * cmake: check for XShm.h (#2318) by @cspiel1 in https://github.com/baresip/baresip/pull/2319 * pulse_async: avoid integer overrun for timestamps in recorder by @cspiel1 in https://github.com/baresip/baresip/pull/2321 * ua: use sdp connection data instead origin by @sreimers in https://github.com/baresip/baresip/pull/2298 * rtpext: move from baresip to re by @alfredh in https://github.com/baresip/baresip/pull/2322 * acc,stream: add rtcp_mux account param by @sreimers in https://github.com/baresip/baresip/pull/2320 * video: video_update cleanup by @sreimers in https://github.com/baresip/baresip/pull/2324 * aufile/src: add auframe support by @sreimers in https://github.com/baresip/baresip/pull/2325 * ice/tmr_async_handler: fix possible segfault by @sreimers in https://github.com/baresip/baresip/pull/2326 * webrtc: fix browser offer handling by @sreimers in https://github.com/baresip/baresip/pull/2327 * Space at the beginning of sip: creates errors by @mbattista in https://github.com/baresip/baresip/pull/2329 * opus_multistream: update mimetype to ad-hoc standard by @alfredh in https://github.com/baresip/baresip/pull/2328 * webrtc: add offerer and recvonly options by @sreimers in https://github.com/baresip/baresip/pull/2331 * test: replace RSA cert with EC cert by @alfredh in https://github.com/baresip/baresip/pull/2332 * Add OPTIONS handling for webrtc demo by @RenSym in https://github.com/baresip/baresip/pull/2333 * mk: remove rtpext.c from srcs.mk by @cspiel1 in https://github.com/baresip/baresip/pull/2336 * ua: change refer log to info() by @alfredh in https://github.com/baresip/baresip/pull/2338 ### New Contributors * @SimonHyde-BBC made their first contribution in https://github.com/baresip/baresip/pull/2311 * @RenSym made their first contribution in https://github.com/baresip/baresip/pull/2333 **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 https://github.com/baresip/baresip/pull/2165 * log: optional timestamps by @cspiel1 in https://github.com/baresip/baresip/pull/2169 * avcodec: remove H263 codec by @alfredh in https://github.com/baresip/baresip/pull/2182 * mk: bump PROJECT_NUMBER in Doxyfile by @cspiel1 in https://github.com/baresip/baresip/pull/2201 * stream: correct Doxygen for peer field by @cspiel1 in https://github.com/baresip/baresip/pull/2202 * cmake: add pre version handling by @sreimers in https://github.com/baresip/baresip/pull/2203 * cmake,debian: use dh-cmake by @sreimers in https://github.com/baresip/baresip/pull/2204 * cmake: add pkgconfig by @robert-scheck in https://github.com/baresip/baresip/pull/2205 * Avoid webrtc_aecm module C++20 extension warnings by @juha-h in https://github.com/baresip/baresip/pull/2215 * cmake/ctrld_dbus: ninja and subdirectory fixes by @sreimers in https://github.com/baresip/baresip/pull/2221 * cmake: link CMAKE_CURRENT_BINARY_DIR modules by @sreimers in https://github.com/baresip/baresip/pull/2223 * cmake,debian: fix libbaresip dependency by @sreimers in https://github.com/baresip/baresip/pull/2224 * cmake: set C only flags by @sreimers in https://github.com/baresip/baresip/pull/2226 * FindPNG needs to find also include directory by @juha-h in https://github.com/baresip/baresip/pull/2230 * FindVPX needs to find also include directory by @juha-h in https://github.com/baresip/baresip/pull/2231 * Multicast send events on mcreg enable commands by @cHuberCoffee in https://github.com/baresip/baresip/pull/2219 * call, menu: support display name for outgoing calls by @cspiel1 in https://github.com/baresip/baresip/pull/2220 * call: hangup call on transp reset if necessary by @maximilianfridrich in https://github.com/baresip/baresip/pull/2229 * portaudio: add mediadev_add with mediadev driver fields by @sreimers in https://github.com/baresip/baresip/pull/2233 * call: fix mnat call_streams_alloc by @sreimers in https://github.com/baresip/baresip/pull/2242 * jack: fix CodeQL uninitialized local variable by @sreimers in https://github.com/baresip/baresip/pull/2244 * Avoid snapshot compiler warnings by @juha-h in https://github.com/baresip/baresip/pull/2239 * avformat: remove old call to avcodec_register_all() by @alfredh in https://github.com/baresip/baresip/pull/2246 * avformat: remove LIBAVUTIL_VERSION_MAJOR check by @alfredh in https://github.com/baresip/baresip/pull/2247 * ua: wording for warning in ua_refer_send() by @cspiel1 in https://github.com/baresip/baresip/pull/2249 * ua: use mbuf functions for ua_connect_dir by @cspiel1 in https://github.com/baresip/baresip/pull/2250 * ci: use actions/checkout@v3 by @sreimers in https://github.com/baresip/baresip/pull/2254 * avcodec: remove av_packet_free() wrapper by @alfredh in https://github.com/baresip/baresip/pull/2255 * selfview: create window in encode_update by @alfredh in https://github.com/baresip/baresip/pull/2253 * alsa: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2256 * config: fix template for avcodec_xxx by @alfredh in https://github.com/baresip/baresip/pull/2258 * avformat: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2259 * v4l2: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2261 * avcodec: remove LIBAVUTIL_VERSION_MAJOR check by @alfredh in https://github.com/baresip/baresip/pull/2260 * multicast: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2262 * menu fix display name by @cspiel1 in https://github.com/baresip/baresip/pull/2251 * account: do not complete dial URI if scheme is included by @cspiel1 in https://github.com/baresip/baresip/pull/2267 * menu: simplify URI complete by @cspiel1 in https://github.com/baresip/baresip/pull/2268 * gtk: use new function account_uri_complete_strdup() by @cspiel1 in https://github.com/baresip/baresip/pull/2273 * Removed module avformat dependency on libpostproc by @juha-h in https://github.com/baresip/baresip/pull/2274 * make: detect and add swscale module in modules.mk by @agorgl in https://github.com/baresip/baresip/pull/2281 * cmake: add APP_MODULES symlinks by @sreimers in https://github.com/baresip/baresip/pull/2286 * cmake: use CMAKE_SHARED_MODULE_SUFFIX by @sreimers in https://github.com/baresip/baresip/pull/2292 ## New Contributors * @ninp0 made their first contribution in https://github.com/baresip/baresip/pull/2165 * @agorgl made their first contribution in https://github.com/baresip/baresip/pull/2281 **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 https://github.com/baresip/baresip/pull/2196 ## [2.8.0] - 2022-10-01 * opensles cmake by @juha-h in https://github.com/baresip/baresip/pull/2108 * test/call: Add test\_call\_change\_videodir by @maximilianfridrich in https://github.com/baresip/baresip/pull/2080 * cmake: bump min version to 3.10 by @alfredh in https://github.com/baresip/baresip/pull/2112 * zrtp: remove module, use gzrtp instead by @alfredh in https://github.com/baresip/baresip/pull/2109 * Avoid gzrtp compile warnings by @juha-h in https://github.com/baresip/baresip/pull/2110 * Update video in menu when UA\_EVENT\_CALL\_REMOTE\_SDP is recieved by @juha-h in https://github.com/baresip/baresip/pull/2113 * http/https requests with large body by @fAuernigg in https://github.com/baresip/baresip/pull/2100 * call: send reinvite after established handlers by @maximilianfridrich in https://github.com/baresip/baresip/pull/2117 * refer out of dialog by @cspiel1 in https://github.com/baresip/baresip/pull/2115 * remove unused functions in baresip.h by @cspiel1 in https://github.com/baresip/baresip/pull/2122 * webrtc/demo: make https optional by @sreimers in https://github.com/baresip/baresip/pull/2120 * Restored original working behavior in uag request\_handler by @juha-h in https://github.com/baresip/baresip/pull/2124 * uag: out-of-dialog REFER handler checks to.tag by @cspiel1 in https://github.com/baresip/baresip/pull/2125 * Update media fixes by @cspiel1 in https://github.com/baresip/baresip/pull/2116 * account: set 100rel default to no by @cspiel1 in https://github.com/baresip/baresip/pull/2128 * avcodec: remove usage of old FFmpeg api (before 4.1.9) by @alfredh in https://github.com/baresip/baresip/pull/2126 * rtp: Improve media synchronization by @sreimers in https://github.com/baresip/baresip/pull/2129 * avformat: remove usage of old FFmpeg api by @alfredh in https://github.com/baresip/baresip/pull/2130 * i2s: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/2131 * ci: migrate to CMake by @alfredh in https://github.com/baresip/baresip/pull/2132 * menu: during early media switch on/off ringback by @cspiel1 in https://github.com/baresip/baresip/pull/2133 * call, event, audio: send DTMF via hidden call by @cspiel1 in https://github.com/baresip/baresip/pull/2134 * ua,reg,serreg: fix serial registration mode by @cspiel1 in https://github.com/baresip/baresip/pull/2139 * CodeQL fixes by @sreimers in https://github.com/baresip/baresip/pull/2143 * cmake: set atomic-implicit-seq-cst only for C language by @sreimers in https://github.com/baresip/baresip/pull/2145 * cmake: define -Wshorten-64-to-32 C only by @sreimers in https://github.com/baresip/baresip/pull/2146 * Cmake of webrtc\_aec module plus remove of unused aec.cpp var by @juha-h in https://github.com/baresip/baresip/pull/2144 * cmake: make include dir public by @sreimers in https://github.com/baresip/baresip/pull/2147 * cmake: add APP\_MODULES and APP\_MODULES\_DIR by @sreimers in https://github.com/baresip/baresip/pull/2148 * Added cmake of gst module by @juha-h in https://github.com/baresip/baresip/pull/2149 * Improved call closed message by @juha-h in https://github.com/baresip/baresip/pull/2151 * gtk & menu: Fix potential memory leaks by @maximilianfridrich in https://github.com/baresip/baresip/pull/2153 * call: allocate streams after peer\_uri was set by @cspiel1 in https://github.com/baresip/baresip/pull/2154 * dshow/cmake: fix stdc++ linking with MSVC by @sreimers in https://github.com/baresip/baresip/pull/2156 * cmake: fix MSVC library output name by @sreimers in https://github.com/baresip/baresip/pull/2157 * webrtc: add HAVE\_GETOPT by @sreimers in https://github.com/baresip/baresip/pull/2158 * config: ignore dirent.h on win32 by @sreimers in https://github.com/baresip/baresip/pull/2159 * ua: do not duplicate request URI parameters by @maximilianfridrich in https://github.com/baresip/baresip/pull/2152 * cmake: add netroam module by @robert-scheck in https://github.com/baresip/baresip/pull/2170 * cmake: add portaudio module by @robert-scheck in https://github.com/baresip/baresip/pull/2173 * cmake: add jack module by @robert-scheck in https://github.com/baresip/baresip/pull/2172 * avcodec,config: add setting for keyframe interval by @cspiel1 in https://github.com/baresip/baresip/pull/2171 * cmake: add sdl module by @robert-scheck in https://github.com/baresip/baresip/pull/2174 * call: set peer URI early for incoming calls by @cspiel1 in https://github.com/baresip/baresip/pull/2168 * cmake: Add options -DDEFAULT\_CAFILE="…" and -DDEFAULT\_AUDIO\_DEVICE="…" by @robert-scheck in https://github.com/baresip/baresip/pull/2179 * cmake: add gtk module by @robert-scheck in https://github.com/baresip/baresip/pull/2176 * cmake: add opus\_multistream module by @robert-scheck in https://github.com/baresip/baresip/pull/2175 * cmake: synchronize behaviour of -DSHARE\_PATH="…" with GNU Makefiles by @robert-scheck in https://github.com/baresip/baresip/pull/2180 * cmake: synchronize behaviour of -DMOD\_PATH="…" with GNU Makefiles by @robert-scheck in https://github.com/baresip/baresip/pull/2181 * Move docs/COPYING to LICENSE and update content to match with re/rem by @robert-scheck in https://github.com/baresip/baresip/pull/2188 * cmake: add ABI (soname) versioning by @robert-scheck in https://github.com/baresip/baresip/pull/2187 * misc: Use example domains and IPs by @robert-scheck in https://github.com/baresip/baresip/pull/2186 * cmake: symlink modules by @sreimers in https://github.com/baresip/baresip/pull/2190 * cmake: add mpa module by @robert-scheck in https://github.com/baresip/baresip/pull/2191 ## [2.7.0] - 2022-09-01 * menu: fix menu_ua_carg data preference by @sreimers in https://github.com/baresip/baresip/pull/2045 * call: remember media dir for established state by @cspiel1 in https://github.com/baresip/baresip/pull/2046 * avformat: fix ffmpeg channel_layout deprecation by @sreimers in https://github.com/baresip/baresip/pull/2048 * cmake: add multicast module by @cHuberCoffee in https://github.com/baresip/baresip/pull/2049 * play: ring tone fixes if file_ausrc is set by @cspiel1 in https://github.com/baresip/baresip/pull/2050 * add peerconnection and mediatrack by @alfredh in https://github.com/baresip/baresip/pull/2054 * main,test: close re async before tmr_debug by @sreimers in https://github.com/baresip/baresip/pull/2055 * http: new file for HTTP functions by @alfredh in https://github.com/baresip/baresip/pull/2056 * http: add http_reply_json() by @alfredh in https://github.com/baresip/baresip/pull/2057 * play: tmr_polling has to check if ausrc is used by @cspiel1 in https://github.com/baresip/baresip/pull/2061 * cmake: use object instead of static for modules by @sreimers in https://github.com/baresip/baresip/pull/2064 * [WIP] import baresip-webrtc by @alfredh in https://github.com/baresip/baresip/pull/2059 * FindAMR.cmake fixes/improvements by @juha-h in https://github.com/baresip/baresip/pull/2066 * cmake: fix modules install path and install share files by @sreimers in https://github.com/baresip/baresip/pull/2068 * hook up webrtc to main cmake file by @alfredh in https://github.com/baresip/baresip/pull/2067 * avformat: check shared for both audio+video by @alfredh in https://github.com/baresip/baresip/pull/2069 * cmake: add V4L2 module by @alfredh in https://github.com/baresip/baresip/pull/2071 * Omx remove by @alfredh in https://github.com/baresip/baresip/pull/2070 * cmake: add directfb module by @alfredh in https://github.com/baresip/baresip/pull/2072 * main,webrtc/main: add re_thread_async_init by @sreimers in https://github.com/baresip/baresip/pull/2076 * cmake: add wincons and winwave modules by @alfredh in https://github.com/baresip/baresip/pull/2077 * cmake: add sndfile module by @alfredh in https://github.com/baresip/baresip/pull/2078 * Mention actual GTK+ 3 usage (instead of 2) in README.md by @robert-scheck in https://github.com/baresip/baresip/pull/2079 * ctrl_tcp: change unsafe operations on an mbuf to the safe mbuf interface by @cHuberCoffee in https://github.com/baresip/baresip/pull/2082 * gzrtp: Call event hander when SAS needs to be verified by @juha-h in https://github.com/baresip/baresip/pull/2081 * Generate also MENC_EVENT_PEER_VERIFIED event by @juha-h in https://github.com/baresip/baresip/pull/2084 * gzrtp: Generate only one MENC_EVENT_PEER_VERIFIED event when all streams are verified by @juha-h in https://github.com/baresip/baresip/pull/2086 * config,net: add use_getaddrinfo/dns_getaddrinfo option by @sreimers in https://github.com/baresip/baresip/pull/2087 * cmake: add_compile_options and use re config by @sreimers in https://github.com/baresip/baresip/pull/2089 * cmake/modules: build syslog only if available by @sreimers in https://github.com/baresip/baresip/pull/2090 * cmake: add selftest by @sreimers in https://github.com/baresip/baresip/pull/2093 * cmake: add win32 linklibs by @sreimers in https://github.com/baresip/baresip/pull/2091 * cmake: add mqtt by @sreimers in https://github.com/baresip/baresip/pull/2094 * Improve C11 cchecks by @sreimers in https://github.com/baresip/baresip/pull/2096 * Added cmake of gzrtp module by @juha-h in https://github.com/baresip/baresip/pull/2095 * Gzrtp cmake by @juha-h in https://github.com/baresip/baresip/pull/2102 * Added cmake of webrtc_aecm module by @juha-h in https://github.com/baresip/baresip/pull/2104 * Suppressed unused var warnings in webrtc_aecm module by @juha-h in https://github.com/baresip/baresip/pull/2105 * call: do not set call state to answered, if session progress (PRACK) by @RobertMi21 in https://github.com/baresip/baresip/pull/2106 --- ## [2.6.0] - 2022-08-01 * conf: check input arguments by @alfredh in https://github.com/baresip/baresip/pull/1932 * dtls_srtp: print TLS cipher name by @alfredh in https://github.com/baresip/baresip/pull/1933 * cmake: add AAC module by @alfredh in https://github.com/baresip/baresip/pull/1935 * call, menu: make selective early media RFC-3261 conform by @cspiel1 in https://github.com/baresip/baresip/pull/1929 * config: add flag to enable/disable linklocal by @alfredh in https://github.com/baresip/baresip/pull/1934 * audio: update filters if codec changes by @cspiel1 in https://github.com/baresip/baresip/pull/1937 * Fix CMake fails when OpenSSL is not present by @widgetii in https://github.com/baresip/baresip/pull/1939 * sip: add RFC 3262 support by @maximilianfridrich in https://github.com/baresip/baresip/pull/1930 * Add CMake target to install baresip executable, library and modules by @widgetii in https://github.com/baresip/baresip/pull/1940 * audio: fix SEGV if stream_alloc() fails by @cspiel1 in https://github.com/baresip/baresip/pull/1942 * gst_video: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1943 * ci: test cmake by @alfredh in https://github.com/baresip/baresip/pull/1944 * cmake: add aptx module by @alfredh in https://github.com/baresip/baresip/pull/1945 * avcodec: remove avcodec_free_context wrapper by @alfredh in https://github.com/baresip/baresip/pull/1947 * avcodec: remove old call to avcodec_init() by @alfredh in https://github.com/baresip/baresip/pull/1948 * cmake: add ffmpeg modules by @alfredh in https://github.com/baresip/baresip/pull/1949 * cmake: add codec2 module by @alfredh in https://github.com/baresip/baresip/pull/1950 * thread: thrd_error fixes by @sreimers in https://github.com/baresip/baresip/pull/1955 * Revert PR #1922 by @juha-h in https://github.com/baresip/baresip/pull/1964 * sip: add RFC 3311 support by @maximilianfridrich in https://github.com/baresip/baresip/pull/1941 * cmake: add amr module by @alfredh in https://github.com/baresip/baresip/pull/1962 * ci/misc: bump pr-dependency-action@v0.5 by @sreimers in https://github.com/baresip/baresip/pull/1968 * ci: add cmake/macos by @alfredh in https://github.com/baresip/baresip/pull/1961 * Feature: add user data to call by @copiltembel in https://github.com/baresip/baresip/pull/1951 * cmake: add audiounit module by @alfredh in https://github.com/baresip/baresip/pull/1969 * cmake: add avcapture module by @alfredh in https://github.com/baresip/baresip/pull/1970 * cmake: add coreaudio module by @alfredh in https://github.com/baresip/baresip/pull/1972 * audio: remove unused aubuf for decoding by @cspiel1 in https://github.com/baresip/baresip/pull/1974 * Modules cmake by @viordash in https://github.com/baresip/baresip/pull/1975 * Modules cmake by @viordash in https://github.com/baresip/baresip/pull/1978 * audio: always start reading in TX thread by @cspiel1 in https://github.com/baresip/baresip/pull/1979 * audio: always start reading in TX poll mode by @cspiel1 in https://github.com/baresip/baresip/pull/1980 * multicast: always start reading of TX aubuf by @cspiel1 in https://github.com/baresip/baresip/pull/1981 * pulse_async: reduce number of reconnect attempts by @RobertMi21 in https://github.com/baresip/baresip/pull/1977 * ci/build: replace deprecated macos-10.15 by @sreimers in https://github.com/baresip/baresip/pull/1984 * ci/build/macos: disable dbus by @sreimers in https://github.com/baresip/baresip/pull/1985 * Improve RFC 3262 by @maximilianfridrich in https://github.com/baresip/baresip/pull/1973 * call: do not stop streams on session progress by @cspiel1 in https://github.com/baresip/baresip/pull/1986 * audio: revert some TX commits and fix TX poll mode by @cspiel1 in https://github.com/baresip/baresip/pull/1987 * call: fix heap-buffer-overflow in prack_handler by @sreimers in https://github.com/baresip/baresip/pull/1988 * Improve re_atomic handling by @sreimers in https://github.com/baresip/baresip/pull/1982 * mk/ctrl_dbus: fix atomic implicit warnings by @sreimers in https://github.com/baresip/baresip/pull/1991 * cmake: add mixminus module by @sreimers in https://github.com/baresip/baresip/pull/1994 * cmake: add dtls_srtp module by @alfredh in https://github.com/baresip/baresip/pull/1993 * Stun uri cred by @viordash in https://github.com/baresip/baresip/pull/1960 * event: fix wrong place of the err check by @copiltembel in https://github.com/baresip/baresip/pull/1992 * Added mwi module cmake build by @juha-h in https://github.com/baresip/baresip/pull/1995 * call: disable prack_handler temporarily by @sreimers in https://github.com/baresip/baresip/pull/1998 * Fix prack handling by @maximilianfridrich in https://github.com/baresip/baresip/pull/1999 * ci: test re/rem with cmake by @alfredh in https://github.com/baresip/baresip/pull/1997 * Added cmake of zrtp module by @juha-h in https://github.com/baresip/baresip/pull/2005 * Added cmake of zrtp module by @juha-h in https://github.com/baresip/baresip/pull/2006 * Added cmake of uuid module by @juha-h in https://github.com/baresip/baresip/pull/2007 * cmake: fix openssl linking by @sreimers in https://github.com/baresip/baresip/pull/2008 * Load also pulse-simple library if exists by @juha-h in https://github.com/baresip/baresip/pull/2010 * Added cmake of presence module by @juha-h in https://github.com/baresip/baresip/pull/2011 * cmake: add more libs, stable branch and static build by @sreimers in https://github.com/baresip/baresip/pull/2012 * Added cmake of selfview module by @juha-h in https://github.com/baresip/baresip/pull/2014 * Added cmake of vp8 and vp9 modules by @juha-h in https://github.com/baresip/baresip/pull/2016 * Added cmake of g722 module by @juha-h in https://github.com/baresip/baresip/pull/2015 * Added cmake of srtp module by @juha-h in https://github.com/baresip/baresip/pull/2017 * cmake: add module override option by @sreimers in https://github.com/baresip/baresip/pull/2020 * cmake: add EXTRA_MODULES option by @sreimers in https://github.com/baresip/baresip/pull/2021 * ci/cmake: add brew packages by @sreimers in https://github.com/baresip/baresip/pull/2023 * Added cmake of g726 module by @juha-h in https://github.com/baresip/baresip/pull/2022 * cmake: add ctrl_dbus module by @cspiel1 in https://github.com/baresip/baresip/pull/2000 * cmake: refactor module prefix by @sreimers in https://github.com/baresip/baresip/pull/2024 * Added cmake of snapshot module by @juha-h in https://github.com/baresip/baresip/pull/2026 * Cmake add dshow by @alfredh in https://github.com/baresip/baresip/pull/2031 * fakevideo: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2032 * cmake: add evdev module by @alfredh in https://github.com/baresip/baresip/pull/2033 * aubridge: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2035 * ausine: use C11 threads by @alfredh in https://github.com/baresip/baresip/pull/2036 * cmake: check for HAVE_UNISTD_H by @alfredh in https://github.com/baresip/baresip/pull/2039 * cmake,mk: prepare main version for release by @sreimers in https://github.com/baresip/baresip/pull/2040 * gsm: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/2034 * cmake: add g7221 module by @alfredh in https://github.com/baresip/baresip/pull/2041 ### New Contributors * @widgetii made their first contribution in https://github.com/baresip/baresip/pull/1939 * @maximilianfridrich made their first contribution in https://github.com/baresip/baresip/pull/1930 * @copiltembel made their first contribution in https://github.com/baresip/baresip/pull/1951 --- ## [2.5.0] - 2022-07-01 * audio: add optional decoding buffer by @cspiel1 in https://github.com/baresip/baresip/pull/1842 * audio: RX filter thread needs separate sampv buffer by @cspiel1 in https://github.com/baresip/baresip/pull/1879 * aufile: fix possible data race warning by @cspiel1 in https://github.com/baresip/baresip/pull/1880 * audiounit,coreaudio: fix kAudioObjectPropertyElementMaster deprecation by @sreimers in https://github.com/baresip/baresip/pull/1881 * av1: explicitly check for supported OBU types by @alfredh in https://github.com/baresip/baresip/pull/1882 * audiounit/coreaudio: fix kAudioObjectPropertyElementMain by @sreimers in https://github.com/baresip/baresip/pull/1885 * ci/build: bump macos min. sdk to 10.12 by @sreimers in https://github.com/baresip/baresip/pull/1883 * ci: run only for pull requests and main branch by @sreimers in https://github.com/baresip/baresip/pull/1887 * multicast: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1892 * dtls_srtp: enable ECC by default, remove RSA by @alfredh in https://github.com/baresip/baresip/pull/1891 * ci/build: add ubuntu 22.04 by @sreimers in https://github.com/baresip/baresip/pull/1890 * test: add check for memory leaks by @sreimers in https://github.com/baresip/baresip/pull/1896 * stream,metric: RX real-time - make metric thread-safe by @cspiel1 in https://github.com/baresip/baresip/pull/1895 * Cmake findre by @alfredh in https://github.com/baresip/baresip/pull/1893 * test: wait for both audio and video to be established by @alfredh in https://github.com/baresip/baresip/pull/1903 * docs: remove old TODO file by @alfredh in https://github.com/baresip/baresip/pull/1902 * audio: fixed check for aubuf started flag by @cspiel1 in https://github.com/baresip/baresip/pull/1904 * use new mutex interface by @cspiel1 in https://github.com/baresip/baresip/pull/1905 * audio: make rx.filtl thread-safe by @cspiel1 in https://github.com/baresip/baresip/pull/1897 * audio: allocate correct buffer size for static auplay srate by @cspiel1 in https://github.com/baresip/baresip/pull/1906 * Pulseaudio Async Interface Module by @cHuberCoffee in https://github.com/baresip/baresip/pull/1907 * Do not destroy register client when it is unregistered by @juha-h in https://github.com/baresip/baresip/pull/1908 * Two spaces are required after email address by @juha-h in https://github.com/baresip/baresip/pull/1909 * cmake: add alsa module by @alfredh in https://github.com/baresip/baresip/pull/1910 * cmake: fix static openssl and thread linking by @sreimers in https://github.com/baresip/baresip/pull/1911 * In start_registering, create register clients if reg list is empty by @juha-h in https://github.com/baresip/baresip/pull/1913 * ctrl_dbus: use new thread and mtx interface by @cspiel1 in https://github.com/baresip/baresip/pull/1916 * cmake: add pulse and pulse_async module by @cHuberCoffee in https://github.com/baresip/baresip/pull/1919 * Un-subscribe mwi at un-register by @juha-h in https://github.com/baresip/baresip/pull/1918 * call: update media on session progress. by @RobertMi21 in https://github.com/baresip/baresip/pull/1922 * ctrl_dbus send event in main thread by @cspiel1 in https://github.com/baresip/baresip/pull/1921 * uag: add timestamps to SIP trace by @cspiel1 in https://github.com/baresip/baresip/pull/1914 * main: fix open timers check by @sreimers in https://github.com/baresip/baresip/pull/1925 * cmake: add account module by @alfredh in https://github.com/baresip/baresip/pull/1926 --- ## [2.4.0] - 2022-06-01 * mulitcast unmute bad quality by @cspiel1 in https://github.com/baresip/baresip/pull/1821 * menu ringback for parallel call by @cspiel1 in https://github.com/baresip/baresip/pull/1827 * multicast: support error code EAGAIN of jbuf_get() by @cspiel1 in https://github.com/baresip/baresip/pull/1832 * use RTP clock rate for timestamp calculation by @cspiel1 in https://github.com/baresip/baresip/pull/1834 * av1 obu by @alfredh in https://github.com/baresip/baresip/pull/1835 * av1 packetizer by @alfredh in https://github.com/baresip/baresip/pull/1836 * av1: depacketizer by @alfredh in https://github.com/baresip/baresip/pull/1837 * Disabled debug statement by @juha-h in https://github.com/baresip/baresip/pull/1838 * h264: move from rem to re by @sreimers in https://github.com/baresip/baresip/pull/1839 * ua: send new event UA_EVENT_CREATE at successful ua allocation by @cHuberCoffee in https://github.com/baresip/baresip/pull/1840 * evdev: fix wrong ioctl size by @sreimers in https://github.com/baresip/baresip/pull/1843 * aufile: ausrc_prm has to be copied when source is allocated by @cspiel1 in https://github.com/baresip/baresip/pull/1844 * conf: missing pointer initialization found by clang analyzer by @cspiel1 in https://github.com/baresip/baresip/pull/1845 * mk/modules: fix omx RPI detection by @sreimers in https://github.com/baresip/baresip/pull/1847 * auconv: add auconv_to_float (fixes #1833) by @alfredh in https://github.com/baresip/baresip/pull/1849 * avfilter: migrate to C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1850 * avformat: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1851 * selfview: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1852 * audio: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1853 * metric: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1854 * play: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1855 * dns: add query cache by @sreimers in https://github.com/baresip/baresip/pull/1848 * video: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1856 * aufile: C11 threads by @alfredh in https://github.com/baresip/baresip/pull/1858 * audio: add more locking by @alfredh in https://github.com/baresip/baresip/pull/1857 * aufile/play: fix run data race by @sreimers in https://github.com/baresip/baresip/pull/1859 * mc: multicast receiver enable state fix by @cHuberCoffee in https://github.com/baresip/baresip/pull/1861 * audio: C11 thread by @alfredh in https://github.com/baresip/baresip/pull/1860 * av1: add packetize handler by @alfredh in https://github.com/baresip/baresip/pull/1865 * net/net_debug: add default route hint by @sreimers in https://github.com/baresip/baresip/pull/1864 * ice: fix local prio calculation by @sreimers in https://github.com/baresip/baresip/pull/1863 * avformat: open codec if not passthrough by @alfredh in https://github.com/baresip/baresip/pull/1866 * dtls_srtp: Minor whitespace fix by @robert-scheck in https://github.com/baresip/baresip/pull/1870 * vp8: add packetize handler by @alfredh in https://github.com/baresip/baresip/pull/1868 * vp9: add packetizer by @alfredh in https://github.com/baresip/baresip/pull/1871 * debug_cmd: support absolute path for command aufileinfo by @cspiel1 in https://github.com/baresip/baresip/pull/1875 * event: add diverter URI to UA event by @cspiel1 in https://github.com/baresip/baresip/pull/1876 * aufileinfo with synchronous response by @cspiel1 in https://github.com/baresip/baresip/pull/1877 --- ## [2.3.0] - 2022-05-01 * mc: multicast mute function by @cHuberCoffee in https://github.com/baresip/baresip/pull/1805 * mc: reject incoming call if high prio multicast is received by @cHuberCoffee in https://github.com/baresip/baresip/pull/1804 * mc: mcplayer stream fade-out and fade-in by @cHuberCoffee in https://github.com/baresip/baresip/pull/1802 * clean_number now will remove all non-digit chars by @mbattista in https://github.com/baresip/baresip/pull/1806 * Workflows cmakelint by @alfredh in https://github.com/baresip/baresip/pull/1808 * ccheck: check all CMakeLists.txt files by @sreimers in https://github.com/baresip/baresip/pull/1810 * mk: remove win32 MSVC project files by @alfredh in https://github.com/baresip/baresip/pull/1811 * cmake: add modules by @sreimers in https://github.com/baresip/baresip/pull/1812 * ajb,aubuf: timestamp is given in [us] by @cspiel1 in https://github.com/baresip/baresip/pull/1809 * call: allow optional leading space in SIP INFO for dtmf-relay by @thomas-karl in https://github.com/baresip/baresip/pull/1814 * conf: add fs_file_extension() by @alfredh in https://github.com/baresip/baresip/pull/1816 * Updated debian version by @juha-h in https://github.com/baresip/baresip/pull/1817 * pulse: fix timestamp integer overrun for arm by @cspiel1 in https://github.com/baresip/baresip/pull/1818 * fix audio multicast artefacts by @cspiel1 in https://github.com/baresip/baresip/pull/1819 * audio: flush aubuf if ssrc changes by @cspiel1 in https://github.com/baresip/baresip/pull/1822 * Debian control dependency update by @juha-h in https://github.com/baresip/baresip/pull/1823 * pulse: support restart of pulseaudio during stream by @cspiel1 in https://github.com/baresip/baresip/pull/1824 * version 2.3.0 by @alfredh in https://github.com/baresip/baresip/pull/1826 ### New Contributors * @thomas-karl made their first contribution in https://github.com/baresip/baresip/pull/1814 --- ## [2.0.2] - 2022-04-09 * Added API function call_diverteruri by @juha-h in https://github.com/baresip/baresip/pull/1780 * Avoid undeclared 'CLOCK_REALTIME' on RHEL/CentOS 7 (fixes #1781) by @robert-scheck in https://github.com/baresip/baresip/pull/1782 * audio: add lock in audio_send_digit by @GGGO in https://github.com/baresip/baresip/pull/1786 * vumeter: use new auframe_level() by @sreimers in https://github.com/baresip/baresip/pull/1788 * reg.c: use already declared acc by @GGGO in https://github.com/baresip/baresip/pull/1789 * aubuf adaptive jitter buffer by @cspiel1 in https://github.com/baresip/baresip/pull/1784 * multicast set aubuf silence by @cspiel1 in https://github.com/baresip/baresip/pull/1791 * ccheck: fix line number in error print by @cspiel1 in https://github.com/baresip/baresip/pull/1793 * test: check the correct stream in UA_EVENT_CALL_MENC by @alfredh in https://github.com/baresip/baresip/pull/1794 * audio: missing lock around stream_send by @GGGO in https://github.com/baresip/baresip/pull/1796 * docs: remove obsolete jitter_buffer_wish from config example by @cspiel1 in https://github.com/baresip/baresip/pull/1798 * Multicast jbuf and aubuf changes by @cHuberCoffee in https://github.com/baresip/baresip/pull/1797 * uag: uag_hold_resume() should not return err if there is no call to hold by @cspiel1 in https://github.com/baresip/baresip/pull/1799 * stream: remove mbuf_get_left check in rtp_handler by @GGGO in https://github.com/baresip/baresip/pull/1801 * cmake: preliminary support by @alfredh in https://github.com/baresip/baresip/pull/1800 ### New Contributors * @GGGO made their first contribution in https://github.com/baresip/baresip/pull/1786 --- ## [2.0.1] - 2022-03-27 ### What's Changed * audio: fix rx_thread (adaptive jitter buffer) by @sreimers in https://github.com/baresip/baresip/pull/1769 * test: init fixture by @alfredh in https://github.com/baresip/baresip/pull/1772 * test: refactoring of test_account_uri_complete by @alfredh in https://github.com/baresip/baresip/pull/1773 * mk: check also if extensions/XShm.h is present by @cspiel1 in https://github.com/baresip/baresip/pull/1774 * menu: support custom SIP headers by @cspiel1 in https://github.com/baresip/baresip/pull/1775 * menu: use new sdp_dir_decode by @cspiel1 in https://github.com/baresip/baresip/pull/1776 * menu: avoid multiple hash entries with same key by @cspiel1 in https://github.com/baresip/baresip/pull/1777 * menu: support audio file config value "none" by @cspiel1 in https://github.com/baresip/baresip/pull/1778 * intercom: add video preview call by @cspiel1 in https://github.com/baresip/baresip/pull/1779 --- ## [2.0.0] - 2022-03-11 ### What's Changed * debug_cmd: use module_event() for aufileinfo events by @cspiel1 in https://github.com/baresip/baresip/pull/1345 * multicast: use module_event() for sending events by @cspiel1 in https://github.com/baresip/baresip/pull/1346 * ctrl_dbus: use module_event() to send exported event by @cspiel1 in https://github.com/baresip/baresip/pull/1347 * ua,call: add CALL_EVENT_OUTGOING by @cspiel1 in https://github.com/baresip/baresip/pull/1348 * GTK caller history by @mbattista in https://github.com/baresip/baresip/pull/1350 * Convert FRITZ!Box XML phone book into Baresip contacts by @robert-scheck in https://github.com/baresip/baresip/pull/1382 * menu: play ringtone on audio_alert device by @cspiel1 in https://github.com/baresip/baresip/pull/1396 * menu: use str_isset() for command parameter by @cspiel1 in https://github.com/baresip/baresip/pull/1397 * dtls_srtp: use elliptic curve cryptography by @cHuberCoffee in https://github.com/baresip/baresip/pull/1385 * Support for s16 playback in jack. Needed for play tones by @srperens in https://github.com/baresip/baresip/pull/1399 * Check that account ;sipnat param has valid value by @juha-h in https://github.com/baresip/baresip/pull/1401 * Tls sipcert per acc by @cHuberCoffee in https://github.com/baresip/baresip/pull/1376 * Vidsrc add packet handler by @alfredh in https://github.com/baresip/baresip/pull/1402 * ToS for video and sip by @cspiel1 in https://github.com/baresip/baresip/pull/1393 * account: add accounts parameter to force media address family by @cspiel1 in https://github.com/baresip/baresip/pull/1395 * Selective early media by @cspiel1 in https://github.com/baresip/baresip/pull/1398 * ua,uag: split ua.c and uag.c by @cspiel1 in https://github.com/baresip/baresip/pull/1349 * Account media af template by @cspiel1 in https://github.com/baresip/baresip/pull/1406 * account: add missing client certificate parameter to template by @cHuberCoffee in https://github.com/baresip/baresip/pull/1408 * account: update answermode values in template by @cspiel1 in https://github.com/baresip/baresip/pull/1405 * menu: command uafind raises UA to head by @cspiel1 in https://github.com/baresip/baresip/pull/1407 * ctrl_dbus: fix possible memleak on failed initialization by @cspiel1 in https://github.com/baresip/baresip/pull/1410 * video passthrough by @alfredh in https://github.com/baresip/baresip/pull/1418 * menu: enable auto answer calls also for command dialdir by @cspiel1 in https://github.com/baresip/baresip/pull/1412 * menu: add command for settings media local direction by @cspiel1 in https://github.com/baresip/baresip/pull/1413 * Accounts addr params by @cspiel1 in https://github.com/baresip/baresip/pull/1414 * Accounts example cleanup by @cspiel1 in https://github.com/baresip/baresip/pull/1415 * menu,call: fix hangup for outgoing call by @cspiel1 in https://github.com/baresip/baresip/pull/1417 * multicast: add source and player API calls by @cHuberCoffee in https://github.com/baresip/baresip/pull/1403 * menu: add command /uareg by @alfredh in https://github.com/baresip/baresip/pull/1421 * menu: return complete URI for commands dial,dialdir by @cspiel1 in https://github.com/baresip/baresip/pull/1424 * menu: in command dialdir call uag_find_requri() with uri by @cspiel1 in https://github.com/baresip/baresip/pull/1425 * gst: replace variable length array (buf) with mem_zalloc by @sreimers in https://github.com/baresip/baresip/pull/1426 * menu: avoid possible memleaks for dial/dialdir commands by @cspiel1 in https://github.com/baresip/baresip/pull/1430 * uag: use local cuser for selecting user-agent (#1433) by @cspiel1 in https://github.com/baresip/baresip/pull/1434 * Work on Intercom module by @cspiel1 in https://github.com/baresip/baresip/pull/1432 * Attended Transfer on GTK by @mbattista in https://github.com/baresip/baresip/pull/1435 * Update README.md with configuration suggestion by @webstean in https://github.com/baresip/baresip/pull/1438 * README fixes by @juha-h in https://github.com/baresip/baresip/pull/1440 * Accounts examples and template by @cspiel1 in https://github.com/baresip/baresip/pull/1441 * serreg: use a timer for registration restart by @cspiel1 in https://github.com/baresip/baresip/pull/1445 * gst: audio playback not correct for some WAV files. by @RobertMi21 in https://github.com/baresip/baresip/pull/1442 * Working on intercom (ringtone override) by @cspiel1 in https://github.com/baresip/baresip/pull/1436 * Use line number 0 if user did not provide any line number by @negbie in https://github.com/baresip/baresip/pull/1451 * AMR Bandwidth Efficient mode support by @srperens in https://github.com/baresip/baresip/pull/1423 * Working on Intercom (menu: allow other modules to reject a call) by @cspiel1 in https://github.com/baresip/baresip/pull/1437 * auframe: add samplerate and channels by @sreimers in https://github.com/baresip/baresip/pull/1452 * account: comment out very basic example in template by @cspiel1 in https://github.com/baresip/baresip/pull/1458 * call answer media dir by @cspiel1 in https://github.com/baresip/baresip/pull/1449 * Account auto answer beep by @cspiel1 in https://github.com/baresip/baresip/pull/1461 * serreg: unregister correct User-Agents on registration failure by @cspiel1 in https://github.com/baresip/baresip/pull/1462 * mk: enable auto-detect of av1 module by @alfredh in https://github.com/baresip/baresip/pull/1463 * ctrl dbus makefile depends by @cspiel1 in https://github.com/baresip/baresip/pull/1457 * stream: check if media is present before enabling the RTP timeout by @cspiel1 in https://github.com/baresip/baresip/pull/1465 * ctrl_dbus: generate dbus code and documentation in makefile by @cspiel1 in https://github.com/baresip/baresip/pull/1456 * auframe: always set srate and ch by @janh in https://github.com/baresip/baresip/pull/1468 * auto answer beep per alert info URI by @cspiel1 in https://github.com/baresip/baresip/pull/1466 * auframe: move to rem by @sreimers in https://github.com/baresip/baresip/pull/1470 * mixminus: add conference feature by @sreimers in https://github.com/baresip/baresip/pull/1411 * vidbridge: check vidbridge_disp_display args fixes segfault by @sreimers in https://github.com/baresip/baresip/pull/1471 * gst: fixed some memory leaks by @RobertMi21 in https://github.com/baresip/baresip/pull/1476 * ua, menu: move auto answer delay handling to menu (#1474) by @cspiel1 in https://github.com/baresip/baresip/pull/1475 * ua,menu: move handling of ANSWERMODE_AUTO to menu (#1474) by @cspiel1 in https://github.com/baresip/baresip/pull/1478 * ausine: support for multiple samplerates by @alfredh in https://github.com/baresip/baresip/pull/1479 * account: fix IPv6 only URI for account_uri_complete() by @cspiel1 in https://github.com/baresip/baresip/pull/1472 * ilbc: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1483 * aubridge/device: remove unused sampv_out (old resample code) by @sreimers in https://github.com/baresip/baresip/pull/1484 * pkg-config version check by @sreimers in https://github.com/baresip/baresip/pull/1481 * mk: support more locations for libre.pc and librem.pc by @cspiel1 in https://github.com/baresip/baresip/pull/1486 * net: remove unused domain by @alfredh in https://github.com/baresip/baresip/pull/1489 * audio: fix aufilt_setup update handling by @sreimers in https://github.com/baresip/baresip/pull/1498 * SIP redirect callbackfunction by @cHuberCoffee in https://github.com/baresip/baresip/pull/1495 * add secure websocket tls context by @sreimers in https://github.com/baresip/baresip/pull/1499 * test: add stunuri by @alfredh in https://github.com/baresip/baresip/pull/1503 * turn: refactoring, add compv by @alfredh in https://github.com/baresip/baresip/pull/1505 * fmt: add string to bool function by @cspiel1 in https://github.com/baresip/baresip/pull/1501 * mk: check glib-2.0 at least like in ubuntu 18.04 by @cspiel1 in https://github.com/baresip/baresip/pull/1507 * registration fixes by @cspiel1 in https://github.com/baresip/baresip/pull/1510 * uag,menu: add commands to enable/disable UDP/TCP/TLS by @cspiel1 in https://github.com/baresip/baresip/pull/1502 * config,audio: add setting audio.telev_pt by @cspiel1 in https://github.com/baresip/baresip/pull/1509 * stream: fix telephone event (#1494) by @cspiel1 in https://github.com/baresip/baresip/pull/1506 * Fix I2S compile error, use auframe by @andreaswatch in https://github.com/baresip/baresip/pull/1512 * ci/tools: fix pylint by @sreimers in https://github.com/baresip/baresip/pull/1515 * config: not all audio config was printed by @cspiel1 in https://github.com/baresip/baresip/pull/1516 * net: replace network_if_getname with net_if_getname by @sreimers in https://github.com/baresip/baresip/pull/1518 * account: add setting audio payload type for telephone-event by @cspiel1 in https://github.com/baresip/baresip/pull/1517 * uag,menu: simplify transport enable/disable and support also ws/wss by @cspiel1 in https://github.com/baresip/baresip/pull/1514 * rst: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1519 * turn: add TCP and TLS transports by @alfredh in https://github.com/baresip/baresip/pull/1520 * speex_pp: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1521 * call: allow video calls by only rejecting a call without any common codecs by @cHuberCoffee in https://github.com/baresip/baresip/pull/1523 * multicast: add missing join for multicast addresses by @cHuberCoffee in https://github.com/baresip/baresip/pull/1524 * confg,uag: rework on sip_transports setting by @cspiel1 in https://github.com/baresip/baresip/pull/1525 * ua: check if peer is capable of video for early video by @cHuberCoffee in https://github.com/baresip/baresip/pull/1526 * mqtt/subscribe: replace fixed command buf and increase response size by @sreimers in https://github.com/baresip/baresip/pull/1527 * mqtt: add reconnect handling (lost broker connection) by @sreimers in https://github.com/baresip/baresip/pull/1528 * event: increase module_event buffer size by @sreimers in https://github.com/baresip/baresip/pull/1532 * mqtt/subscribe: use safe odict_string to prevent crashes by @sreimers in https://github.com/baresip/baresip/pull/1534 * stream: add stream_set_label by @alfredh in https://github.com/baresip/baresip/pull/1537 * Makefile dependency check improvements by @sreimers in https://github.com/baresip/baresip/pull/1531 * account: add enable/disable flag for video by @cspiel1 in https://github.com/baresip/baresip/pull/1536 * audio: use account specific audio telev pt correctly by @cspiel1 in https://github.com/baresip/baresip/pull/1542 * net: add missing HAVE_INET6 by @cspiel1 in https://github.com/baresip/baresip/pull/1543 * account: remove unused API function for video enable by @cspiel1 in https://github.com/baresip/baresip/pull/1544 * gst: changed log level for end of file message by @RobertMi21 in https://github.com/baresip/baresip/pull/1548 * multicast: add new configurable multicast TTL config parameter by @cHuberCoffee in https://github.com/baresip/baresip/pull/1545 * call: fix early video capability check (wrong SDP direction checked) by @cHuberCoffee in https://github.com/baresip/baresip/pull/1549 * audio: catch end of file message in ausrc error handler (#1539) by @RobertMi21 in https://github.com/baresip/baresip/pull/1550 * menu: added stopringing command by @RobertMi21 in https://github.com/baresip/baresip/pull/1551 * stream: remove obsolete rx.jbuf_started by @cspiel1 in https://github.com/baresip/baresip/pull/1552 * ua: downgrade level of message "ua: using best effort AF" by @viordash in https://github.com/baresip/baresip/pull/1553 * outgoing calls early callid by @cspiel1 in https://github.com/baresip/baresip/pull/1547 * audio: changed log level for ausrc error handler messages by @RobertMi21 in https://github.com/baresip/baresip/pull/1554 * SIP default protocol by @cspiel1 in https://github.com/baresip/baresip/pull/1538 * serreg: fix server selection in case all server were unavailable by @cHuberCoffee in https://github.com/baresip/baresip/pull/1557 * multicast: fix missing unlock by @alfredh in https://github.com/baresip/baresip/pull/1559 * config: replace strcpy by saver re_snprintf (#1558) by @cspiel1 in https://github.com/baresip/baresip/pull/1560 * multicast: fix coverity scan by @alfredh in https://github.com/baresip/baresip/pull/1561 * odict: hide struct odict_entry by @sreimers in https://github.com/baresip/baresip/pull/1562 * ctrl_dbus: use mqueue to trigger processing of command in remain thread by @cspiel1 in https://github.com/baresip/baresip/pull/1565 * multicast,config: add separate jitter buffer configuration by @cspiel1 in https://github.com/baresip/baresip/pull/1566 * ua: emit CALL_CLOSED event when user agent is deleted by @cspiel1 in https://github.com/baresip/baresip/pull/1564 * core: move stream_enable_rtp_timeout to api by @sreimers in https://github.com/baresip/baresip/pull/1569 * stream: add mid sdp attribute by @alfredh in https://github.com/baresip/baresip/pull/1570 * rtpext: change length type to size_t by @alfredh in https://github.com/baresip/baresip/pull/1573 * avcodec: remove old backwards compat wrapper by @alfredh in https://github.com/baresip/baresip/pull/1575 * main: Added option (-a) to set the ua agent string. by @RobertMi21 in https://github.com/baresip/baresip/pull/1576 * menu fix tones for parallel outgoing calls by @cspiel1 in https://github.com/baresip/baresip/pull/1577 * Fix win32 by @viordash in https://github.com/baresip/baresip/pull/1579 * Fix static analyzer warnings by @viordash in https://github.com/baresip/baresip/pull/1580 * call: added auto dtmf mode by @RobertMi21 in https://github.com/baresip/baresip/pull/1583 * RTP inbound telephone events should not lead to packet loss by @cspiel1 in https://github.com/baresip/baresip/pull/1581 * Running tests in a win32 project by @viordash in https://github.com/baresip/baresip/pull/1585 * stream: wrong media direction after setting stream to hold by @RobertMi21 in https://github.com/baresip/baresip/pull/1587 * move network check to module by @cspiel1 in https://github.com/baresip/baresip/pull/1584 * serreg: do not ignore returned errors of ua_register() by @cspiel1 in https://github.com/baresip/baresip/pull/1589 * Bundle media mux by @alfredh in https://github.com/baresip/baresip/pull/1588 * mixausrc: no warnings flood when sampc changes by @cspiel1 in https://github.com/baresip/baresip/pull/1595 * ua: select laddr with route to SDP offer address by @cspiel1 in https://github.com/baresip/baresip/pull/1590 * net,uag: allow incoming peer-to-peer calls with user@domain by @cspiel1 in https://github.com/baresip/baresip/pull/1591 * uag: in uag_reset_transp() select laddr with route to SDP raddr by @cspiel1 in https://github.com/baresip/baresip/pull/1592 * uag: exit if transport could not be added by @cspiel1 in https://github.com/baresip/baresip/pull/1593 * avcodec: use const AVCodec by @alfredh in https://github.com/baresip/baresip/pull/1602 * module: deprecate module_tmp by @alfredh in https://github.com/baresip/baresip/pull/1600 * test: use ausine as audio source by @alfredh in https://github.com/baresip/baresip/pull/1601 * Selftest fakevideo by @alfredh in https://github.com/baresip/baresip/pull/1604 * When adding local address, check that it has not been added already by @juha-h in https://github.com/baresip/baresip/pull/1606 * start without network by @cspiel1 in https://github.com/baresip/baresip/pull/1607 * config: add netroam module by @sreimers in https://github.com/baresip/baresip/pull/1608 * multicast: allow any port number for sender and receiver by @cHuberCoffee in https://github.com/baresip/baresip/pull/1609 * netroam: add netlink immediate network change detection by @cspiel1 in https://github.com/baresip/baresip/pull/1612 * remove uag transp rm (#1611) by @cspiel1 in https://github.com/baresip/baresip/pull/1616 * net dns srv get by @cspiel1 in https://github.com/baresip/baresip/pull/1615 * move calls to stream_start_rtcp to call.c by @alfredh in https://github.com/baresip/baresip/pull/1617 * video: null pointer check for the display handler by @cspiel1 in https://github.com/baresip/baresip/pull/1621 * audio: add lock by @alfredh in https://github.com/baresip/baresip/pull/1619 * ua: select proper af and laddr for outgoing IP calls by @cspiel1 in https://github.com/baresip/baresip/pull/1618 * audio: lock stream by @alfredh in https://github.com/baresip/baresip/pull/1622 * test: replace mock ausrc with ausine by @alfredh in https://github.com/baresip/baresip/pull/1623 * menu ringback session progress by @cspiel1 in https://github.com/baresip/baresip/pull/1625 * New module providing webrtc aec mobile mode filter by @juha-h in https://github.com/baresip/baresip/pull/1626 * uag: respect setting sip_listen (#1627) by @cspiel1 in https://github.com/baresip/baresip/pull/1628 * select laddr for SDP with respect to net_interface by @cspiel1 in https://github.com/baresip/baresip/pull/1630 * stream: do not start audio during early-video by @cspiel1 in https://github.com/baresip/baresip/pull/1629 * remove struct media_ctx by @alfredh in https://github.com/baresip/baresip/pull/1632 * ci: add libwebrtc-audio-processing-dev (module webrtc_aec) by @sreimers in https://github.com/baresip/baresip/pull/1635 * auconv: new module for audio format conversion by @alfredh in https://github.com/baresip/baresip/pull/1634 * Support for IPv6 link local address for streams by @cspiel1 in https://github.com/baresip/baresip/pull/1624 * call: check if address family is valid also for video stream by @cspiel1 in https://github.com/baresip/baresip/pull/1636 * audio: pass pointer to tx->ausrc_prm instead of local variable by @cspiel1 in https://github.com/baresip/baresip/pull/1637 * menu: add an event for call transfer by @cspiel1 in https://github.com/baresip/baresip/pull/1641 * netroam: error handling for reset transport by @cspiel1 in https://github.com/baresip/baresip/pull/1642 * mk: use CC_TEST for auto detect modules by @sreimers in https://github.com/baresip/baresip/pull/1647 * test: use dtls_srtp.so module instead of mock by @alfredh in https://github.com/baresip/baresip/pull/1646 * stream: create jbuf only if use_rtp is set by @cspiel1 in https://github.com/baresip/baresip/pull/1648 * multicast: fix memleak in player destructor by @cspiel1 in https://github.com/baresip/baresip/pull/1653 * stream: split up sender/receiver by @alfredh in https://github.com/baresip/baresip/pull/1654 * set sdp laddr to SIP src address by @cspiel1 in https://github.com/baresip/baresip/pull/1645 * serreg fix fallback accounts by @cspiel1 in https://github.com/baresip/baresip/pull/1660 * ctrl_dbus: print command with the warning by @cspiel1 in https://github.com/baresip/baresip/pull/1662 * call: new transfer call state to handle transfered calls correctly by @cHuberCoffee in https://github.com/baresip/baresip/pull/1658 * serreg: prevent fast register retries if offline by @cspiel1 in https://github.com/baresip/baresip/pull/1663 * av1: update packetization code by @alfredh in https://github.com/baresip/baresip/pull/1657 * call: magic check in sipsess_desc_handler() by @cspiel1 in https://github.com/baresip/baresip/pull/1664 * alsa: use snd_pcm_drop instead of snd_pcm_drain by @sreimers in https://github.com/baresip/baresip/pull/1669 * Increased debian compat level to 10 by @juha-h in https://github.com/baresip/baresip/pull/1667 * conf: fix conf_configure_buf() config parse by @sreimers in https://github.com/baresip/baresip/pull/1666 * stream flush rtp socket by @cspiel1 in https://github.com/baresip/baresip/pull/1671 * Transfer like rfc5589 by @cHuberCoffee in https://github.com/baresip/baresip/pull/1678 * GTK: mem_derefer call earlier by @mbattista in https://github.com/baresip/baresip/pull/1682 * netroam: add fail counter and event by @cspiel1 in https://github.com/baresip/baresip/pull/1685 * Added API functions stream_metric_get_(tx|rx)_bitrate by @juha-h in https://github.com/baresip/baresip/pull/1686 * Multicast new functions by @cHuberCoffee in https://github.com/baresip/baresip/pull/1687 * avcodec: Enable pass-through for more codecs by @abrodkin in https://github.com/baresip/baresip/pull/1692 * menu: filter for the correct call state in menu_selcall by @cHuberCoffee in https://github.com/baresip/baresip/pull/1693 * test: fix warning on mingw32 by @alfredh in https://github.com/baresip/baresip/pull/1696 * menu: Play ringback in play device by @myrkr in https://github.com/baresip/baresip/pull/1698 * sip: add optional TCP source port by @cspiel1 in https://github.com/baresip/baresip/pull/1695 * rtpext: change id unsigned -> uint8_t by @alfredh in https://github.com/baresip/baresip/pull/1701 * ci: add mingw build test by @sreimers in https://github.com/baresip/baresip/pull/1700 * test: use mediaenc srtp instead of mock by @alfredh in https://github.com/baresip/baresip/pull/1702 * test: remove mock mediaenc by @alfredh in https://github.com/baresip/baresip/pull/1704 * descr: add session_description by @alfredh in https://github.com/baresip/baresip/pull/1706 * use fs_isfile() by @alfredh in https://github.com/baresip/baresip/pull/1709 * stream: only call rtp_clear for audio by @alfredh in https://github.com/baresip/baresip/pull/1710 * checks if call is available before calling call, closes #1708 by @mbattista in https://github.com/baresip/baresip/pull/1712 * conf: add conf_loadfile by @alfredh in https://github.com/baresip/baresip/pull/1713 * ice: remove ice_mode by @sreimers in https://github.com/baresip/baresip/pull/1714 * audio: use auframe in encode_rtp_send, ref #1699 by @alfredh in https://github.com/baresip/baresip/pull/1715 * Increased account's max video codec count from four to eight by @juha-h in https://github.com/baresip/baresip/pull/1717 * gtk: Avoid duplicate call_timer registration by @myrkr in https://github.com/baresip/baresip/pull/1719 * Attended call transfer by @cHuberCoffee in https://github.com/baresip/baresip/pull/1718 * menu: exclude given call when searching for active call by @cspiel1 in https://github.com/baresip/baresip/pull/1721 * menu: play call waiting tone on audio_player device by @cspiel1 in https://github.com/baresip/baresip/pull/1722 * ci/build/macos: link ffmpeg@4 by @sreimers in https://github.com/baresip/baresip/pull/1725 * module auresamp by @cspiel1 in https://github.com/baresip/baresip/pull/1705 * test: remove h264 testcode, already in retest by @alfredh in https://github.com/baresip/baresip/pull/1726 * h265: move from avcodec to rem by @alfredh in https://github.com/baresip/baresip/pull/1728 * mc: send more details at receiver - timeout event by @cHuberCoffee in https://github.com/baresip/baresip/pull/1731 * h265: move packetizer from avcodec to rem by @alfredh in https://github.com/baresip/baresip/pull/1732 * FFmpeg 5 by @sreimers in https://github.com/baresip/baresip/pull/1734 * Fixing clang ThreadSanitizer warnings by @sreimers in https://github.com/baresip/baresip/pull/1730 * auresamp: replace anonymous union for pre C11 compilers by @cspiel1 in https://github.com/baresip/baresip/pull/1738 * aufile: align naming of alloc handlers by @sreimers in https://github.com/baresip/baresip/pull/1739 * auresamp fixes by @cspiel1 in https://github.com/baresip/baresip/pull/1741 * mc: new priority handling with multicast state by @cHuberCoffee in https://github.com/baresip/baresip/pull/1740 * remove support for Solaris platform by @alfredh in https://github.com/baresip/baresip/pull/1745 * Allow hanging up call that has not been ACKed yet by @juha-h in https://github.com/baresip/baresip/pull/1747 * Multicast identical condition and fmt string fix by @cHuberCoffee in https://github.com/baresip/baresip/pull/1751 * audio: allocate aubuf before ausrc_alloc (fixes data race) by @sreimers in https://github.com/baresip/baresip/pull/1748 * call: send supported header for 200 answering/ok by @cHuberCoffee in https://github.com/baresip/baresip/pull/1752 * event: check if media line is present for encoding audio/video dir by @cspiel1 in https://github.com/baresip/baresip/pull/1754 * Removed unused variable in modules/webrtc_aec/aec.cpp by @juha-h in https://github.com/baresip/baresip/pull/1756 * audio use module auconv by @cspiel1 in https://github.com/baresip/baresip/pull/1742 * test: use aufile module by @alfredh in https://github.com/baresip/baresip/pull/1757 * x11grab: remove module, use avformat.so instead by @alfredh in https://github.com/baresip/baresip/pull/1758 * audio: declare iterator inside for-loop (C99) by @alfredh in https://github.com/baresip/baresip/pull/1759 * aufile: set run=true before write thread starts (#1727) by @cspiel1 in https://github.com/baresip/baresip/pull/1762 * Added new API function call_supported() and used it in menu module by @juha-h in https://github.com/baresip/baresip/pull/1761 * aufile: separate aufile_src.c from aufile.c by @cspiel1 in https://github.com/baresip/baresip/pull/1765 * ctrl_dbus: fix possible data race (#1727) by @cspiel1 in https://github.com/baresip/baresip/pull/1764 * menu select other call on hangup by @cspiel1 in https://github.com/baresip/baresip/pull/1763 * event: encode also combined media direction by @cspiel1 in https://github.com/baresip/baresip/pull/1766 ## New Contributors * @srperens made their first contribution in https://github.com/baresip/baresip/pull/1399 * @negbie made their first contribution in https://github.com/baresip/baresip/pull/1451 * @andreaswatch made their first contribution in https://github.com/baresip/baresip/pull/1512 * @viordash made their first contribution in https://github.com/baresip/baresip/pull/1553 * @abrodkin made their first contribution in https://github.com/baresip/baresip/pull/1692 * @myrkr made their first contribution in https://github.com/baresip/baresip/pull/1698 --- ## [1.1.0] - 2021-04-24 - cons: emulate key-release -- ref #1329 - Correct reverse domain name notation (#1342) [#1342] - gtk with account_uri_complete (#1339) [#1339] - bump version to 1.1.0 -- ref #1333 - ui: fix leaking of cmd_ctx (#1338) [#1338] - DTMF tones for A B C D (#1340) [#1340] - account: use a fixed username for the template - contact: update contacts template - config: disable ctrl_dbus in config template - Module event (#1335) [#1335] - add event UA_EVENT_MODULE to tell to app when snapshot has been written (#1330) [#1330] - ringtone: generated busy and ringback tone (#1332) [#1332] - audio: prevent restart of rx_thread on call termination (#1331) [#1331] - modules: update auplay/ausrc modules - Auplay remove inheritance (#1328) [#1328] - h264: add doxygen comment - vidloop: add VIDEO_SRATE - vidloop: check error - vidloop: add vidframe_clear - vidloop: split enable_codec into encoder/decoder - Ausrc remove inheritance (#1326) [#1326] - ua: remove prev call (#1323) [#1323] - sndfile: get number of bytes from auframe - plc: check format of struct auframe - speex_pp: check format of struct auframe - webrtc_aec: use format from struct auframe - README: update codecs and RFCs - menu: use uri complete for command dialdir (#1321) [#1321] - video: check for video display before calling handler - Changed name and made public (#1319) [#1319] - menu: return call-id for dial and dialdir (#1320) [#1320] - Fixes for account uri complete (#1318) [#1318] - Avoid compiler warnings: - Avoid compiler warnings (I haven't found anything wrong with the code) - vidfilt: fix warning - vidfilt: split parameters into encode/decode - snapshot: fix warnings - video: group functions from vidutil.c - avfilter: fix warnings - vumeter: use format from audio frame - replaced ua_uri_complete with account_uri_complete (#1317) [#1317] - aulevel: move to librem - omx: fix warning - vidisp: remove inheritance (#1316) [#1316] - docs: change video settings to match the default values (#1315) [#1315] - menu: select call in cmd_find_call() (#1314) [#1314] - menu: use menu_stop_play() (#1311) [#1311] - main: unload app modules in signal handler (#1310) [#1310] - avformat: replace const double with double - avformat: clean up ifdefs (#1313) [#1313] - ci: drop ubuntu 16.04 support - end of life - avformat: proper code formatting - avcodec: add avcodec prefix to log messages - avcodec: check length of H265 packet - x11grab: remove vidsrc inheritance - v4l2: remove vs inheritance - vidsrc: remove concept of baseclass/inheritance - ua,menu: remove uag_find_call_state (#1304) [#1304] - Updated homepage - sdl: correct aspect-ratio in fullscreen mode - vidloop: add vidisp parameters - auloop: use auframe_size - audio: use auframe_size - Auplay use auframe (#1305) [#1305] - Docs examples config (#1302) [#1302] - Serreg fixes (#1301) [#1301] - Update config.c [#1303] - contact: use uag_find_requri() - ua: use new tls function to set cafile and path [#1300] - config: add sip_capath config line - Call event answered fixes alsa issue (#1299) [#1299] - ctrl_dbus: send DBUS signal when dbus interface is ready (#1296) [#1296] - Multicast call priority (#1291) [#1291] - Menu fixes for play tones2 (#1294) [#1294] - gst: add missing include unistd.h [#1297] - multicast: cleanup function description and fix doxygen warning (#1292) [#1292] - menu: remove call resume for command hangup (#1289) [#1289] - ua: add a generic filter API for calls (#1293) [#1293] - Merge pull request #1288 from cspiel1/remove_call_resume_on_termination [#1288] - menu: remove call resume on termination - multicast: fix build error when using HAVE_PTHREAD= - alsa_play.c add suggestion to use dmix (#1283) [#1283] - readme.md: added multicast module (#1282) [#1282] - audiounit: fix typo - update copyright year (#1287) [#1287] - config cleanup (#1286) [#1286] - update copyright year (#1285) [#1285] - conf: add call_hold_other_calls config option (#1280) [#1280] - config.c: added rtmp to config template (#1284) [#1284] - main.c: update year [#1281] - The avformat_decoder should be optional (#1277) [#1277] - src/audio: set started false with audio_stop (#1278) [#1278] - readme: update baresip fork links - ausine: mono support and stereo_left/right option [#1274] - menu: fix incoming calls are not selected on call termination (#1271) [#1271] - test: remove mock_aucodec, using g711 instead - opengl: remove deprecated module (#1268) [#1268] - Added account_dtmfmode and account_set_dtmfmode API functions (#1269) [#1269] - avcodec: remove support for MPEG4 codec - call: start streams asynchronously (issue #1261) (#1267) [#1267] - audio: remove special handling of Comfort Noise - multicast: fix one doxygen warning - menu: update doxygen comment - menu: correct hangupall command for parallel call feature (#1264) [#1264] - menu: on call termination select another active call (#1260) [#1260] - ua: correct doxygen of uag_hold_resume() [#1262] - menu: simplify cmd_hangupall() (#1259) [#1259] - support for sending of DTMF INFO (#1258) [#1258] - Menu optional call parameter (#1254) [#1254] - cleanup tabs and spaces [#1256] - ua: correct doxygen for uag_hold_others() - ua: add doxygen for call find functions - menu: add doxygen to cmd_hangup(), cmd_hold(), cmd_resume() - menu: command accept searches all User-Agents for an incoming call - ua: add function uag_find_call_state() - menu: print correct warning for hangup, accept, hold, resume - menu: add optional parameter call-id to cmd_call_resume() - menu: add optional parameter call-id to cmd_call_hold() - menu: add optional parameter call-id to cmd_hangup() - menu: add optional parameter call-id to cmd_answerdir() - menu: add utility function that decodes complex command parameters - menu: use SDP_SENDRECV for cmd_answerdir() as fallback - menu: add optional parameter call-id to cmd_answer() - ua: add call find per call-id function - call: call_info() prints also the call-id - ua: in ua_print_calls() print User-Agent info in header - menu: ua NULL check for answer command - replace spaces with tab [#1249] - removed newline - undid httpreq spacing - fixed line too long - moved multicast template to end of config template - ua: fix uag_hold_others use of wrong list element [#1253] - added multicast enabled message (#1251) [#1251] - updated date and added multicast to signaling (#1252) [#1252] - Merge pull request #1248 from webstean/patch-2 [#1248] - Added newline to multicast comment - Menu ensure only one established call (#1247) [#1247] - Call resume on hangup (#1246) [#1246] - menu: for call answer search all UAs for calls to put on hold - ua: ua_answer() should answer same call like ua_hold_answer() - ua: make ua_find_call_state() global usable - Add multicast_listener to config template (#1245) [#1245] - Update config template to include multicast module (#1244) [#1244] - menu: if a call becomes established then put others on hold - ua: add uag_hold_others() - Fix multiple resumed calls (#1242) [#1242] - Merge pull request #1241 from cHuberCoffee/cmd_hangupall [#1241] - RFC: Make avformat decode mjpeg v4l2 with vaapi (#1216) [#1216] - ua: add doxygen for new uag_hold_resume() - menu: fix missing callid of menu at call closed - menu: use uag_hold_resume to ensure only one active call - ua: on call resume check for other active calls - menu: new hangupall command with direction parameter - readme: update supported compilers and ssl libs - menu: fix redial - Fix spaces - Multicast module (#1231) [#1231] - menu: use print backend pointer pf correctly (#1222) [#1222] - menu: start ringback only once for parallel calls (#1238) [#1238] - jack: support port pattern in config file (#1237) [#1237] - config: disables server verification if sip_verify_server is missing (#1236) [#1236] - ua: for UA selection allow arbitrary aor for regint=0 accounts (#1234) [#1234] - Ctrl dbus synchronize (#1232) [#1232] - event: encode also remote audio direction (#1227) [#1227] - Merge pull request #1235 from cspiel1/event_add_string_for_UA_EVENT_CUSTOM [#1235] - event: add string for UA_EVENT_CUSTOM - Mimic ifdef on avutil version for hwcontext - Fix to tabs and improve checks - src/config: show sip_cafile warning only if sip_verify_server is enabled - Avoid compiler warnings using casts [#1228] - test: disable SIP TLS server verification [#1224] - config,ua: add config flag disable SIP TLS server verification - alsa/play: snd_pcm_writei error codes are negative - alsa: fix clang warnings "conversion loses integer precision" [#1223] - Intelligent call answer (#1218) [#1218] - Remove uag next (#1207) [#1207] - Merge pull request #1219 from cspiel1/message_reply_once [#1219] - menu: update switch_audio_player - Make vaapi/mjpeg options of avformat - src/config: no sip_cafile wording - message: reply only once - src/ua: only warn if tls_add_ca fails, same as undefined cafile [#1214] - src/config: add sip_cafile warning and enable by default - ua: change log message from warning to info - video: fix video payload text - Make avformat decode mjpeg v4l2 with vaapi - ua: improve UA selection for incoming calls (#1206) [#1206] - ua: limit account matches for incoming calls to non-registrar accounts - ua: check for NULL parameter in uag_find_msg() - ua: early exit for AF_UNSPEC in uri_match_af() - ua: use sip_transp_decode() in uri_match_transport() - ua: use arrays in uri_host_local() - test: add test for deny UDP peer-to-peer call - ua: improve UA selection for incoming calls - Sip message to application (#1201) [#1201] - opus: Ensure (re)init of fmtp strings (#1209) [#1209] - ctrl_dbus: generate dbus interface during build (#1208) [#1208] - mod_gtk: switch to gtk 3 (#1203) [#1203] - menu: set_answer_mode: apply all uas - menu: find_call: search all user-agents - menu: fix usage of ua - isac: remove deprecated module (#1204) [#1204] - menu: cmd_print_calls: print all uas - Fix interaction between CLI menu and GTK menu (#1202) [#1202] - menu: rename menu_current() to menu_uacur() - webrtc_aec: fix compilation with gcc 4.9 (fix #1193) - win32: add cons module, fixes #1197 - ua: remove ua_aor() -- use account_aor() instead - gtk: use account_aor() - menu: use account_aor() - presence: use account_aor() - modules: use account_aor() - account: fix video codes decode (#1196) [#1196] - core: use account_aor() - Merge pull request #1198 from baresip/av1 [#1198] - Avoid unused parameter warning - debug_cmd: add UA_EVENT_CUSTOM (#1194) [#1194] - fix decoder changed debug text - cairo: minor debug tuning - menu: add uadelall to delete all user agents [#1195] - use account_aor() - mctrl: remove support for media-control (deprecated) - update doxygen comments - ua: minor cleanup - ua: split struct uag from instance - README: add RFC 5373 - menu: fix segfault on last account deletion (#1192) [#1192] - call: extend SIP auto answer support for incoming calls (#1191) [#1191] - Sip auto answer caller (#1188) [#1188] - win32: remove timer.c - ua: give a nice name to 'global' struct - ua: remove ua_cur - move uag_current to menu module - menu: pass ua from mqtt to menu via opaque data - Sip autoanswer callee (#1187) [#1187] - ua: for answer-mode early also send INCOMING event (#1185) [#1185] - gst: The error handler call for end of stream is now (#1182) [#1182] - mk: also detect mqtt.so in SYSROOT_ALT - contact: add ua_lookup_domain - video: minor tuning of pipeline text - gst: playback of read only audio files failed (#1183) [#1183] - gtk: make a local pointer to current ua - menu: clean up usage of uag_current() - call: correction of remote video direction info at SDP-offer (#1181) [#1181] - debug_cmd: print all user-agents - presence: one command with status as argument - ua: rename presence status to pstat - ua: remove LIBRE_HAVE_SIPTRACE check, always enabled - update doxygen comments - mk: update doxygen config file - menu: initialize menu with zeros (#1179) [#1179] - Re mk cross build2 (#1161) [#1161] - net: make fallback DNS ignored message debug only - mixausrc: improve logging [#1176] - mixausrc: fix shorten-64-to-32 warnings - config: template for osx/ios - Supressed clang zero length array warning - Added ctx param to video_stop/video_stop_source and set ctx to null (#1173) [#1173] - avformat: add empty line after base class - Make macos warnings into errors (#1171) [#1171] - disable mixausrc until warnings are fixed - clang shorten-64-to-32 warnings (#1170) [#1170] - Mixausrc (#1159) [#1159] - aufile: fix warning on OSX - alsa: print warning if running, fixed #1162 - Don't default stunuser/pass to account authuser/pass (#1164) [#1164] - Audio file info (#1157) [#1157] - gitignore: clangd cache, compile_commands.json and cleanup - Merge pull request #1167 from baresip/video_display [#1167] - Reordered video_stop_display - Expose video_stop_display() to API - Video dir rename (#1158) [#1158] - ci: use baresip/rem repo - stream: add function to send a RTP dummy packet (#1156) [#1156] - Play aufile extended support (#1155) [#1155] - video: move video related start/stop/update into video file (#1151) [#1151] - aufile: add audio player to write speaker data to wav file (#1153) [#1153] - Fix compiler warnings (#1152) [#1152] - play: fix warning - play ausrc (#1147) [#1147] - README: add more status badges - README: replace travis status badge - menu: fix uint16_t scode [#1149] - config: revert dirent.h changes - audio: fix HAVE_PTHREAD audio_destructor - gst ready for file play (#1148) [#1148] - debug_cmd: mem_deref of player fixes segfault (#1146) [#1146] - net: remove deprecated net_domain() - update contact examples - fix freeze on hangup (#1135) (#1145) [#1145] - menu: make audio files configurable (#1144) [#1144] - aptx: declare variable outside for-loop - fix warnings on openbsd - jack: declare variable outside for loop - account: declare variable outside for loop - coreaudio: declare variable outside for loop - menu: initialize menu.play fixes segfault (#1143) [#1143] - ausine: declare variable outside for loop - timer: remove tmr_jiffies_usec (replaced by libre) (#1141) [#1141] - Adaptive jbuf (#1112) [#1112] - Update build.yml (#1140) [#1140] - mqtt: allow to separate pub from sub topic base (#1139) [#1139] - video: fix warning - mqtt: fix printing port and add tls support (#1138) [#1138] - httpreq: in cmd_setauth check if parameter was given (#1134) [#1134] - Merge pull request #1132 from baresip/pr-dependency-action [#1132] - ci: add pull request dependency checkouts - audio: remove redundant union - menu: use menu_ as prefix for global symbols - menu: use menu_ as prefix for global symbols - ci: add apt-get update - menu: module refactoring (#1129) [#1129] - audio, video, stream: check payload type before put to jbuf (#1128) [#1128] - Cmd dialdir (#1126) [#1126] - Cmd acceptdir (#1125) [#1125] - event: add register fallback to event string and class name (#1124) [#1124] - avformat: use %u for unsigned - modify event type and check if peeruri null (#1119) [#1119] - event: move code from ua.c (#1118) [#1118] - Valgrind ci (#1117) [#1117] - h264 cleanup, second part (#1115) [#1115] - h264 cleanup (#1114) [#1114] - Merge pull request #1113 from baresip/github-actions-v2 [#1113] - ci: remove travis - ci: add github actions - replaces travisci - qtcapture: remove deprecated module (#1107) [#1107] - test: prepare for dualstack - test: add mock dns_server_add_aaaa - make EXTRA_MODULES last, not first (#1106) [#1106] - httpreq: fix cmd_settimeout - test: bind network to localhost, a fix for #1090 - modules/webrtc_aec: link flags fixes (#1105) [#1105] - menu: commands in alphabetical order - httpreq: fix warning about unused args - serreg: fix warnings about unused argument - menu: fix warnings about unused argument - Add a HTTP request module with authorization (#1099) [#1099] - Menu: corrections for ring tones and call status by means of a global call counter (#1102) [#1102] - mk: remove dirent.h - Updating .vcxproj file for windows builds (#1097) [#1097] - ccheck: change license to BSD license - Merge pull request #1095 from baresip/websocket [#1095] - Serial registration (#1083) [#1083] - Ctrl dbus (#1085) [#1085] - README: remove references to creytiv.com - Branch of baresip that includes Alfred's sip websocket patch - Merge pull request #1091 from baresip/debian [#1091] - ua, menu: new command to print certificate issuer and subject (#1078) [#1078] - .gitignore: add ctags and Vim swp files (#1084) [#1084] ### Contributors (many thanks) - [alfredh](https://github.com/alfredh) - [robert-scheck](https://github.com/robert-scheck) - [mbattista](https://github.com/mbattista) - [cspiel1](https://github.com/cspiel1) - [juha-h](https://github.com/juha-h) - [ahinrichs](https://github.com/ahinrichs) - [jurjen-van-dijk](https://github.com/jurjen-van-dijk) - [sreimers](https://github.com/sreimers) - [cHuberCoffee](https://github.com/cHuberCoffee) - [webstean](https://github.com/webstean) - [viric](https://github.com/viric) - [agramner](https://github.com/agramner) - [weili-jiang](https://github.com/weili-jiang) - [thillux](https://github.com/thillux) - [wkiswk](https://github.com/wkiswk) - [philippbachmann08](https://github.com/philippbachmann08) - [ursfassler](https://github.com/ursfassler) - [RobertMi21](https://github.com/RobertMi21) - [alberanid](https://github.com/alberanid) - [agranig](https://github.com/agranig) - [nanguantong](https://github.com/nanguantong) - [johnjuuljensen](https://github.com/johnjuuljensen) --- ## [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](https://github.com/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 [#960]: https://github.com/baresip/baresip/pull/960 [#966]: https://github.com/baresip/baresip/pull/966 [#973]: https://github.com/baresip/baresip/pull/973 [#977]: https://github.com/baresip/baresip/pull/977 [#978]: https://github.com/baresip/baresip/pull/978 [#980]: https://github.com/baresip/baresip/pull/980 [#981]: https://github.com/baresip/baresip/pull/981 [#986]: https://github.com/baresip/baresip/pull/986 [#992]: https://github.com/baresip/baresip/pull/992 [#993]: https://github.com/baresip/baresip/pull/993 [#994]: https://github.com/baresip/baresip/pull/994 [#996]: https://github.com/baresip/baresip/pull/996 [#1006]: https://github.com/baresip/baresip/pull/1006 [#1007]: https://github.com/baresip/baresip/pull/1007 [#1009]: https://github.com/baresip/baresip/pull/1009 [#1010]: https://github.com/baresip/baresip/pull/1010 [#1011]: https://github.com/baresip/baresip/pull/1011 [#1012]: https://github.com/baresip/baresip/pull/1012 [#1013]: https://github.com/baresip/baresip/pull/1013 [#1015]: https://github.com/baresip/baresip/pull/1015 [#1016]: https://github.com/baresip/baresip/pull/1016 [#1018]: https://github.com/baresip/baresip/pull/1018 [#1019]: https://github.com/baresip/baresip/pull/1019 [#1020]: https://github.com/baresip/baresip/pull/1020 [#1021]: https://github.com/baresip/baresip/pull/1021 [#1022]: https://github.com/baresip/baresip/pull/1022 [#1023]: https://github.com/baresip/baresip/pull/1023 [#1025]: https://github.com/baresip/baresip/pull/1025 [#1028]: https://github.com/baresip/baresip/pull/1028 [#1029]: https://github.com/baresip/baresip/pull/1029 [#1030]: https://github.com/baresip/baresip/pull/1030 [#1031]: https://github.com/baresip/baresip/pull/1031 [#1034]: https://github.com/baresip/baresip/pull/1034 [#1037]: https://github.com/baresip/baresip/pull/1037 [#1038]: https://github.com/baresip/baresip/pull/1038 [#1043]: https://github.com/baresip/baresip/pull/1043 [#1056]: https://github.com/baresip/baresip/pull/1056 [#1059]: https://github.com/baresip/baresip/pull/1059 [#1061]: https://github.com/baresip/baresip/pull/1061 [#1062]: https://github.com/baresip/baresip/pull/1062 [#1065]: https://github.com/baresip/baresip/pull/1065 [#1068]: https://github.com/baresip/baresip/pull/1068 [#1069]: https://github.com/baresip/baresip/pull/1069 [#1073]: https://github.com/baresip/baresip/pull/1073 [#1078]: https://github.com/baresip/baresip/pull/1078 [#1079]: https://github.com/baresip/baresip/pull/1079 [#1081]: https://github.com/baresip/baresip/pull/1081 [#1082]: https://github.com/baresip/baresip/pull/1082 [#1083]: https://github.com/baresip/baresip/pull/1083 [#1084]: https://github.com/baresip/baresip/pull/1084 [#1085]: https://github.com/baresip/baresip/pull/1085 [#1091]: https://github.com/baresip/baresip/pull/1091 [#1095]: https://github.com/baresip/baresip/pull/1095 [#1097]: https://github.com/baresip/baresip/pull/1097 [#1099]: https://github.com/baresip/baresip/pull/1099 [#1102]: https://github.com/baresip/baresip/pull/1102 [#1105]: https://github.com/baresip/baresip/pull/1105 [#1106]: https://github.com/baresip/baresip/pull/1106 [#1107]: https://github.com/baresip/baresip/pull/1107 [#1112]: https://github.com/baresip/baresip/pull/1112 [#1113]: https://github.com/baresip/baresip/pull/1113 [#1114]: https://github.com/baresip/baresip/pull/1114 [#1115]: https://github.com/baresip/baresip/pull/1115 [#1117]: https://github.com/baresip/baresip/pull/1117 [#1118]: https://github.com/baresip/baresip/pull/1118 [#1119]: https://github.com/baresip/baresip/pull/1119 [#1124]: https://github.com/baresip/baresip/pull/1124 [#1125]: https://github.com/baresip/baresip/pull/1125 [#1126]: https://github.com/baresip/baresip/pull/1126 [#1128]: https://github.com/baresip/baresip/pull/1128 [#1129]: https://github.com/baresip/baresip/pull/1129 [#1132]: https://github.com/baresip/baresip/pull/1132 [#1134]: https://github.com/baresip/baresip/pull/1134 [#1138]: https://github.com/baresip/baresip/pull/1138 [#1139]: https://github.com/baresip/baresip/pull/1139 [#1140]: https://github.com/baresip/baresip/pull/1140 [#1141]: https://github.com/baresip/baresip/pull/1141 [#1143]: https://github.com/baresip/baresip/pull/1143 [#1144]: https://github.com/baresip/baresip/pull/1144 [#1145]: https://github.com/baresip/baresip/pull/1145 [#1146]: https://github.com/baresip/baresip/pull/1146 [#1147]: https://github.com/baresip/baresip/pull/1147 [#1148]: https://github.com/baresip/baresip/pull/1148 [#1149]: https://github.com/baresip/baresip/pull/1149 [#1151]: https://github.com/baresip/baresip/pull/1151 [#1152]: https://github.com/baresip/baresip/pull/1152 [#1153]: https://github.com/baresip/baresip/pull/1153 [#1155]: https://github.com/baresip/baresip/pull/1155 [#1156]: https://github.com/baresip/baresip/pull/1156 [#1157]: https://github.com/baresip/baresip/pull/1157 [#1158]: https://github.com/baresip/baresip/pull/1158 [#1159]: https://github.com/baresip/baresip/pull/1159 [#1161]: https://github.com/baresip/baresip/pull/1161 [#1164]: https://github.com/baresip/baresip/pull/1164 [#1167]: https://github.com/baresip/baresip/pull/1167 [#1170]: https://github.com/baresip/baresip/pull/1170 [#1171]: https://github.com/baresip/baresip/pull/1171 [#1173]: https://github.com/baresip/baresip/pull/1173 [#1176]: https://github.com/baresip/baresip/pull/1176 [#1179]: https://github.com/baresip/baresip/pull/1179 [#1181]: https://github.com/baresip/baresip/pull/1181 [#1182]: https://github.com/baresip/baresip/pull/1182 [#1183]: https://github.com/baresip/baresip/pull/1183 [#1185]: https://github.com/baresip/baresip/pull/1185 [#1187]: https://github.com/baresip/baresip/pull/1187 [#1188]: https://github.com/baresip/baresip/pull/1188 [#1191]: https://github.com/baresip/baresip/pull/1191 [#1192]: https://github.com/baresip/baresip/pull/1192 [#1194]: https://github.com/baresip/baresip/pull/1194 [#1195]: https://github.com/baresip/baresip/pull/1195 [#1196]: https://github.com/baresip/baresip/pull/1196 [#1198]: https://github.com/baresip/baresip/pull/1198 [#1201]: https://github.com/baresip/baresip/pull/1201 [#1202]: https://github.com/baresip/baresip/pull/1202 [#1203]: https://github.com/baresip/baresip/pull/1203 [#1204]: https://github.com/baresip/baresip/pull/1204 [#1206]: https://github.com/baresip/baresip/pull/1206 [#1207]: https://github.com/baresip/baresip/pull/1207 [#1208]: https://github.com/baresip/baresip/pull/1208 [#1209]: https://github.com/baresip/baresip/pull/1209 [#1214]: https://github.com/baresip/baresip/pull/1214 [#1216]: https://github.com/baresip/baresip/pull/1216 [#1218]: https://github.com/baresip/baresip/pull/1218 [#1219]: https://github.com/baresip/baresip/pull/1219 [#1222]: https://github.com/baresip/baresip/pull/1222 [#1223]: https://github.com/baresip/baresip/pull/1223 [#1224]: https://github.com/baresip/baresip/pull/1224 [#1227]: https://github.com/baresip/baresip/pull/1227 [#1228]: https://github.com/baresip/baresip/pull/1228 [#1231]: https://github.com/baresip/baresip/pull/1231 [#1232]: https://github.com/baresip/baresip/pull/1232 [#1234]: https://github.com/baresip/baresip/pull/1234 [#1235]: https://github.com/baresip/baresip/pull/1235 [#1236]: https://github.com/baresip/baresip/pull/1236 [#1237]: https://github.com/baresip/baresip/pull/1237 [#1238]: https://github.com/baresip/baresip/pull/1238 [#1241]: https://github.com/baresip/baresip/pull/1241 [#1242]: https://github.com/baresip/baresip/pull/1242 [#1244]: https://github.com/baresip/baresip/pull/1244 [#1245]: https://github.com/baresip/baresip/pull/1245 [#1246]: 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