Baresip 2.0.0 (2022-03-11)

• debug_cmd: use module_event() for aufileinfo events
• multicast: use module_event() for sending events
• ctrl_dbus: use module_event() to send exported event
• ua,call: add CALL_EVENT_OUTGOING
• GTK caller history
• Convert FRITZ!Box XML phone book into Baresip contacts
• menu: play ringtone on audio_alert device
• menu: use str_isset() for command parameter
• dtls_srtp: use elliptic curve cryptography
• Support for s16 playback in jack; needed for play tones
• Check that account ;sipnat param has valid value
• Tls sipcert per account
• Vidsrc add packet handler
• ToS for video and sip
• account: add accounts parameter to force media address family
• Selective early media
• ua,uag: split ua.c and uag.c
• Account media af template
• account: add missing client certificate parameter to template
• account: update answermode values in template
• menu: command uafind raises UA to head
• ctrl_dbus: fix possible memleak on failed initialization
• video passthrough
• menu: enable auto answer calls also for command dialdir
• menu: add command for settings media local direction
• Accounts address params
• Accounts example cleanup
• menu,call: fix hangup for outgoing call
• multicast: add source and player API calls
• menu: add command /uareg
• menu: return complete URI for commands dial,dialdir
• menu: in command dialdir call uag_find_requri() with uri
• gst: replace variable length array (buf) with mem_zalloc by @sreimers in #1426
• menu: avoid possible memleaks for dial/dialdir commands
• uag: use local cuser for selecting user-agent
• Work on Intercom module
• Attended Transfer on GTK
• Update README.md with configuration suggestion
• README fixes
• Accounts examples and template
• serreg: use a timer for registration restart
• gst: audio playback not correct for some WAV files
• Working on intercom (ringtone override)
• Use line number 0 if user did not provide any line number
• AMR Bandwidth Efficient mode support
• Working on Intercom (menu: allow other modules to reject a call)
• auframe: add samplerate and channels
• account: comment out very basic example in template
• call answer media dir
• Account auto answer beep
• serreg: unregister correct User-Agents on registration failure
• mk: enable auto-detect of av1 module
• ctrl_dbus makefile depends
• stream: check if media is present before enabling the RTP timeout
• ctrl_dbus: generate dbus code and documentation in makefile
• auframe: always set srate and ch
• auto answer beep per alert info URI
• auframe: move to rem
• mixminus: add conference feature
• vidbridge: check vidbridge_disp_display args fixes segfault
• gst: fixed some memory leaks
• ua,menu: move auto answer delay handling to menu
• ua,menu: move handling of ANSWERMODE_AUTO to menu
• ausine: support for multiple samplerates by @alfredh in #1479
• account: fix IPv6 only URI for account_uri_complete()
• ilbc: remove deprecated module
• aubridge/device: remove unused sampv_out (old resample code)
• pkg-config version check
• mk: support more locations for libre.pc and librem.pc
• net: remove unused domain
• audio: fix aufilt_setup update handling
• SIP redirect callbackfunction
• add secure websocket tls context
• test: add stunuri
• turn: refactoring, add compv
• fmt: add string to bool function
• mk: check glib-2.0 at least like in ubuntu 18.04
• registration fixes
• uag,menu: add commands to enable/disable UDP/TCP/TLS
• config,audio: add setting audio.telev_pt
• stream: fix telephone event
• Fix I2S compile error, use auframe
• ci/tools: fix pylint
• config: not all audio config was printed
• net: replace network_if_getname with net_if_getname
• account: add setting audio payload type for telephone-event
• uag,menu: simplify transport enable/disable and support also ws/wss
• rst: remove deprecated module
• turn: add TCP and TLS transports
• speex_pp: remove deprecated module
• call: allow video calls by only rejecting a call without any common codecs
• multicast: add missing join for multicast addresses
• config,uag: rework on sip_transports setting
• ua: check if peer is capable of video for early video
• mqtt/subscribe: replace fixed command buf and increase response size
• mqtt: add reconnect handling (lost broker connection)
• event: increase module_event buffer size
• mqtt/subscribe: use safe odict_string to prevent crashes
• stream: add stream_set_label
• Makefile dependency check improvements
• account: add enable/disable flag for video
• audio: use account specific audio telev pt correctly
• net: add missing HAVE_INET6
• account: remove unused API function for video enable
• gst: changed log level for end of file message
• multicast: add new configurable multicast TTL config parameter
• call: fix early video capability check (wrong SDP direction checked)
• audio: catch end of file message in ausrc error handler
• menu: added stopringing command
• stream: remove obsolete rx.jbuf_started
• ua: downgrade level of message "ua: using best effort AF"
• outgoing calls early callid
• audio: changed log level for ausrc error handler messages
• SIP default protocol
• serreg: fix server selection in case all server were unavailable
• multicast: fix missing unlock
• config: replace strcpy by saver re_snprintf
• multicast: fix coverity scan
• odict: hide struct odict_entry
• ctrl_dbus: use mqueue to trigger processing of command in remain thread
• multicast,config: add separate jitter buffer configuration
• ua: emit CALL_CLOSED event when user agent is deleted
• core: move stream_enable_rtp_timeout to api
• stream: add mid sdp attribute
• rtpext: change length type to size_t
• avcodec: remove old backwards compat wrapper
• main: Added option (-a) to set the ua agent string
• menu fix tones for parallel outgoing calls
• Fix win32
• Fix static analyzer warnings
• call: added auto dtmf mode
• RTP inbound telephone events should not lead to packet loss
• Running tests in a win32 project
• stream: wrong media direction after setting stream to hold
• move network check to module
• serreg: do not ignore returned errors of ua_register()
• Bundle media mux
• mixausrc: no warnings flood when sampc changes
• ua: select laddr with route to SDP offer address
• net,uag: allow incoming peer-to-peer calls with user@domain
• uag: in uag_reset_transp() select laddr with route to SDP raddr
• uag: exit if transport could not be added
• avcodec: use const AVCodec
• module: deprecate module_tmp
• test: use ausine as audio source
• Selftest fakevideo
• When adding local address, check that it has not been added already
• start without network
• config: add netroam module
• multicast: allow any port number for sender and receiver
• netroam: add netlink immediate network change detection
• remove uag transp rm
• net dns srv get
• move calls to stream_start_rtcp to call.c
• video: null pointer check for the display handler
• audio: add lock
• ua: select proper af and laddr for outgoing IP calls
• audio: lock stream
• test: replace mock ausrc with ausine
• menu ringback session progress
• New module providing webrtc aec mobile mode filter
• uag: respect setting sip_listen
• select laddr for SDP with respect to net_interface
• stream: do not start audio during early-video
• remove struct media_ctx
• ci: add libwebrtc-audio-processing-dev (module webrtc_aec)
• auconv: new module for audio format conversion
• Support for IPv6 link local address for streams
• call: check if address family is valid also for video stream
• audio: pass pointer to tx->ausrc_prm instead of local variable
• menu: add an event for call transfer
• netroam: error handling for reset transport
• mk: use CC_TEST for auto detect modules
• test: use dtls_srtp.so module instead of mock
• stream: create jbuf only if use_rtp is set
• multicast: fix memleak in player destructor
• stream: split up sender/receiver
• set sdp laddr to SIP src address
• serreg fix fallback accounts
• ctrl_dbus: print command with the warning
• call: new transfer call state to handle transfered calls correctly
• serreg: prevent fast register retries if offline
• av1: update packetization code
• call: magic check in sipsess_desc_handler()
• alsa: use snd_pcm_drop instead of snd_pcm_drain
• Increased debian compat level to 10
• conf: fix conf_configure_buf() config parse
• stream flush rtp socket
• Transfer like rfc5589
• GTK: mem_derefer call earlier
• netroam: add fail counter and event
• Added API functions stream_metric_get_(tx|rx)_bitrate
• Multicast new functions
• avcodec: Enable pass-through for more codecs
• menu: filter for the correct call state in menu_selcall
• test: fix warning on mingw32
• menu: Play ringback in play device
• sip: add optional TCP source port
• rtpext: change id unsigned -> uint8_t
• ci: add mingw build test
• test: use mediaenc srtp instead of mock
• test: remove mock mediaenc
• descr: add session_description
• use fs_isfile()
• stream: only call rtp_clear for audio
• checks if call is available before calling call
• conf: add conf_loadfile
• ice: remove ice_mode
• audio: use auframe in encode_rtp_send
• Increased account's max video codec count from four to eight
• gtk: Avoid duplicate call_timer registration
• Attended call transfer by
• menu: exclude given call when searching for active call
• menu: play call waiting tone on audio_player device
• ci/build/macos: link ffmpeg@4
• module auresamp
• test: remove h264 testcode, already in retest
• h265: move from avcodec to rem
• mc: send more details at receiver - timeout event
• h265: move packetizer from avcodec to rem
• FFmpeg 5
• Fixing clang ThreadSanitizer warnings
• auresamp: replace anonymous union for pre C11 compilers
• aufile: align naming of alloc handlers
• auresamp fixes
• mc: new priority handling with multicast state
• remove support for Solaris platform
• Allow hanging up call that has not been ACKed yet
• Multicast identical condition and fmt string fix
• audio: allocate aubuf before ausrc_alloc (fixes data race)
• call: send supported header for 200 answering/ok
• event: check if media line is present for encoding audio/video dir
• Removed unused variable in modules/webrtc_aec/aec.cpp
• audio use module auconv
• test: use aufile module
• x11grab: remove module, use avformat.so instead
• audio: declare iterator inside for-loop (C99)
• aufile: set run=true before write thread starts
• Added new API function call_supported() and used it in menu module
• aufile: separate aufile_src.c from aufile.c
• ctrl_dbus: fix possible data race
• menu select other call on hangup
• event: encode also combined media direction

libre v2.1.1 (2022-03-12)

• mk: fix ABI versioning

libre v2.1.0 (2022-03-11)

• Tls sipcert per acc
• ToS for video and sip
• sdp: in media_decode() reset rdir if port is zero
• mk/re: add variable length array (-Wvla) compiler warning
• Macos openssl
• pkg-config version check
• sa: add setter and getter for scope id
• net: in net_dst_source_addr_get() make parameter dst const
• Avoid ISO C90 forbids mixed declarations and code warnings
• SIP redirect callbackfunction
• add secure websocket tls context
• fmt: add string to bool function
• fix clang analyze warnings
• fmt: support different separators for parameter parsing
• Refactor inet_ntop and inet_pton
• add essential fields check
• sa: add support for interface suffix for IPv6ll
• net: fix net_if_getname IPv6 support
• udp: add udp_recv_helper
• sa: fix build for old systems
• sa/addrinfo: fix openbsd (drop AI_V4MAPPED flag)
• ci/codeql: add scan-build
• Fixed debian changelog version
• IPv6 link local support
• sip: add fallback transport for transp_find()
• SIP default protocol
• remove orphaned files
• outgoing calls early callid
• sip: fix possible "???" dns srv queries by skipping lines without srvid
• odict: hide struct odict_entry
• tls: add keylogger callback function
• http/client: support other auth token types besides bearer
• tls: fix client certificate replacement
• http/client: support dns ipv6
• rtp: add payload-type helper
• sip: check consistency between CSeq method and that of request line
• Fix win32
• fix warnings from PVS-Studio C++ static analyzer
• RTP inbound telephone events should not lead to packet loss
• support inet6 by default in Win32 project
• sdp: differentiate between media line disabled or rejected
• move network check to module
• odict: move odict_compare from retest to re
• sip: reuse transport protocol of first request in dialog
• json: fix parsing json containing only single value
• ice: fix checklist
• mk: add compile_commands.json (clang only)
• sdp: debug print session and media direction
• add btrace module (linux/unix only)
• mk: add CC_TEST header check
• init dst address
• ice: check if candpair exist before adding
• mk: add CC_TEST cache
• btrace: use HAVE_EXECINFO
• Coverity
• icem: remove dead code (found by coverity 240639)
• hash: switch to simpler "fast algorithm"
• dns: fix dnsc_alloc with IPv6 disabled
• mk: deprecate HAVE_INET6
• Fix for btrace print for memory leaks
• set sdp laddr to SIP src address
• sdp: include all media formats in SDP offer
• ci: add centos 7 build test
• sip: move sip_auth_encode to public api for easier testing
• sipsess: do not call desc handler on shutdown
• stream flush rtp socket
• ci: fix macos openssl build
• http: HTTP Host header conform to RFC for IPv6 addresses
• Increased debian compatibility level from 9 to 10
• mk: move darwin dns LFLAGS to re.mk (fixes static builds)
• build infrastructure: silent and verbose modes
• mk: use posix regex for sed CC major version detection
• dns: fix parse_resolv_conf for OpenBSD
• sip: add optional TCP source port
• ci: add mingw build and test
• net: remove net_hostaddr
• ci/centos7: add openssl
• hmac: use HMAC() api (fixes OpenSSL 3.0 deprecations)
• md5: use EVP_Digest for newer openssl versions
• sha: add new sha1() api
• OpenSSL 3.0
• udp: add win32 qos support
• ci/mingw: fix dependency checkout
• ice: remove ice_mode
• Codeql security
• aubuf insert auframes sorted
• ci: add valgrind
• tls: remove code for openssl 0.9.5
• ice: remove unused file
• main: remove obsolete OPENWRT epoll check
• dns,http,sa: fix HAVE_INET6 off warnings
• preliminary support for cmake
• make,cmake: set SOVERSION to major version
• mk: remove MSVC project files, use cmake instead
• natbd: remove module (deprecated)
• sha: remove backup implementation
• sha,hmac: use Apple CommonCrypto if defined
• stun: add stun_generate_tid
• add cmakelint
• Cmake version
• cmake: add option to enable/disable rtmp module
• lock: use rwlock by default
• cmake: fixes for MSVC 16
• json: fix win32 warnings
• ci: add cmake build
• mqueue: fix win32 warnings
• tcp: fix win32 warnings
• cmake: fix target_link_libraries for win32
• stun: fix win32 warnings
• udp: fix win32 warnings
• tls: fix win32 warnings
• remove HAVE_INTTYPES_H
• udp: fix win32 warnings
• cmake: minor fixes
• cmake: fix MSVC ninja
• tcp: fix win32 warnings
• udp: fix win32 msvc warnings
• rtmp: fix win32 warning
• bfcp: fix win32 warning
• tls: fix libressl 3.5
• fix coverity scan warnings
• Allow hanging up call that has not been ACKed yet
• mk,cmake: add backtrace support and fix linking on OpenBSD
• github: add CMake and Windows workflow
• Windows (VS 2022/Ninja)
• cmake: fixes for Android
• tmr: reuse tmr_jiffies_usec
• trace: use gettid as thread_id on linux
• tmr: use CLOCK_MONOTONIC_RAW if defined
• add atomic support
• Sonarcloud
• sip: fix gcc 6.3.0 warning for logical expression
• add transport-cc rtcp feedback support

librem v2.0.0 (2022-03-12)

• Restored rgb565 pixel format
• vid: remove pixel formats RGB555 and RGB565
• cmake: version 3.7
• mk: bump dev version
• au,aulevel: add AUFMT_S32LE
• aubuf: add aufbuf_resize()
• cmake: add HAVE_UNISTD_H check
• cmake: add relative re include dir
• cmake: minor fixes
• mk: remove win32 project files
• cmake: use version 3.10
• aubuf: fix mem_deref data race with frame_destructor
• h265: move packetizer from avcodec to rem
• vidmix: fix source_put data race
• vidmix: fix possible data race
• h265: move h265_is_keyframe to rem
• h265: move from avcodec to rem
• preliminary support for CMake
• gitignore: add vim swap and ctags files
• ci: fix ccheck main repo path
• aubuf: insert audio frames sorted by timestamp
• auframe: add auframe_update
• h264: fix win32 compiler cast warning
• mk: bump version v1.0.0-dev3
• Increased debian compatibility level from 9 to 10
• aubuf: remove aubuf_sort_auframe return comment
• aubuf: add aubuf_sort_auframe()
• mk: cleanup cache directory
• clangd: add config (headers only)
• git: ignore clangd files
• Fix win32
• mk: bump dev version
• aubuf: add auframe functions
• add resampler 16<->8 and 32<->16 kHz
• aumix: add aumix_source_mute
• update gitignore for visual studio artifacts
• update PlatformToolset to vs2019
• mk: replace pkg-config modversion
• mk: improve dependency
• mk: ignore dependency check on make clean
• debian: add pkg-config file
• ci: remove ubuntu-16.04 test
• mk: support more locations for libre.pc
• mk: add librem.pc Makefile dependency
• mk: add libre version check and pre-release
• au/fmt: add AUFMT_RAW
• auframe: use enum aufmt for format
• auframe: move from baresip
• h264: add functions from baresip
• debian: fixes soname pkg build
• mk: add abi versioning