baresip-2.0.0-1.fc36, libre-2.1.1-1.fc36, & 1 more

FEDORA-2022-7814b1e12f created by robert a year ago for Fedora 36

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 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 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 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 (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

How to install

Updates may require up to 24 hours to propagate to mirrors. If the following command doesn't work, please retry later:

sudo dnf upgrade --refresh --advisory=FEDORA-2022-7814b1e12f

This update has been submitted for testing by robert.

a year ago

This update's test gating status has been changed to 'ignored'.

a year ago

robert edited this update.

a year ago

This update has been pushed to testing.

a year ago

This update has been submitted for stable by bodhi.

a year ago

This update has been pushed to stable.

a year ago

Please login to add feedback.

Content Type
Test Gating
Unstable by Karma
Stable by Karma
Stable by Time
3 days
a year ago
in testing
a year ago
in stable
a year ago
a year ago
BZ#2019879 [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV
BZ#2063340 libre-2.1.1 is available
BZ#2063450 librem-2.0.0 is available
BZ#2063451 baresip-2.0.0 is available
BZ#2063502 F36FailsToInstall: Multiple packages built from baresip

Automated Test Results