The following Fedora EPEL 7 Security updates need testing:
Age URL
6
https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-b169dce5bc
chromium-99.0.4844.51-1.el7
4
https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-1f3ec359c3
cobbler-2.8.5-5.el7
4
https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-db09048bde
nbd-3.24-1.el7
2
https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-e1430e72de
wordpress-5.1.13-1.el7
The following builds have been pushed to Fedora EPEL 7 updates-testing
abcm2ps-8.14.13-1.el7
baresip-2.0.0-1.el7
globus-gssapi-gsi-14.17-4.el7
libre-2.1.1-1.el7
librem-2.0.0-1.el7
zabbix40-4.0.39-1.el7
zabbix50-5.0.21-1.el7
zchunk-1.2.1-1.el7
Details about builds:
================================================================================
abcm2ps-8.14.13-1.el7 (FEDORA-EPEL-2022-d009c17be8)
A program to typeset ABC tunes into Postscript
--------------------------------------------------------------------------------
Update Information:
New upstream bugfix release.
--------------------------------------------------------------------------------
ChangeLog:
* Sat Mar 12 2022 Stuart Gathman <stuart(a)gathman.org> - 8.14.13-1
- New upstream release
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #2063269 - CVE-2021-32434 CVE-2021-32435 CVE-2021-32436 abcm2ps: multiple
security vulnerabilities [epel-all]
https://bugzilla.redhat.com/show_bug.cgi?id=2063269
--------------------------------------------------------------------------------
================================================================================
baresip-2.0.0-1.el7 (FEDORA-EPEL-2022-c9b3f4b951)
Modular SIP user-agent with audio and video support
--------------------------------------------------------------------------------
Update Information:
# 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
--------------------------------------------------------------------------------
ChangeLog:
* Sun Mar 13 2022 Robert Scheck <robert(a)fedoraproject.org> 2.0.0-1
- Upgrade to 2.0.0 (#2063451)
* Thu Jan 27 2022 Tom Callaway <spot(a)fedoraproject.org> - 1.1.0-8
- rebuild for libvpx
* Wed Jan 19 2022 Fedora Release Engineering <releng(a)fedoraproject.org> - 1.1.0-7
- Rebuilt for
https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild
* Sun Dec 5 2021 Richard Shaw <hobbes1069(a)gmail.com> - 1.1.0-6
- Rebuild for codec2 1.0.1.
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #2019879 - [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV
https://bugzilla.redhat.com/show_bug.cgi?id=2019879
[ 2 ] Bug #2063340 - libre-2.1.1 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063340
[ 3 ] Bug #2063450 - librem-2.0.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063450
[ 4 ] Bug #2063451 - baresip-2.0.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063451
--------------------------------------------------------------------------------
================================================================================
globus-gssapi-gsi-14.17-4.el7 (FEDORA-EPEL-2022-9fd87e9670)
Grid Community Toolkit - GSSAPI library
--------------------------------------------------------------------------------
Update Information:
Fix TLS 1.3 interoperability with dCache gridftp server.
--------------------------------------------------------------------------------
ChangeLog:
* Sun Mar 6 2022 Mattias Ellert <mattias.ellert(a)physics.uu.se> - 14.17-4
- Better logic for TLS 1.3 special handling
- Use sha256 hash when generating test certificates
- Don't test TLS 1.0 and 1.1 when using openssl 3.0.1 or later
* Thu Jan 20 2022 Fedora Release Engineering <releng(a)fedoraproject.org> - 14.17-3
- Rebuilt for
https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild
* Tue Sep 14 2021 Sahana Prasad <sahana(a)redhat.com> - 14.17-2
- Rebuilt with OpenSSL 3.0.0
--------------------------------------------------------------------------------
================================================================================
libre-2.1.1-1.el7 (FEDORA-EPEL-2022-c9b3f4b951)
Library for real-time communications and SIP stack
--------------------------------------------------------------------------------
Update Information:
# 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
--------------------------------------------------------------------------------
ChangeLog:
* Sun Mar 13 2022 Robert Scheck <robert(a)fedoraproject.org> 2.1.1-1
- Upgrade to 2.1.1 (#2063340)
* Fri Mar 11 2022 Robert Scheck <robert(a)fedoraproject.org> 2.1.0-1
- Upgrade to 2.1.0 (#2063340)
* Thu Jan 20 2022 Fedora Release Engineering <releng(a)fedoraproject.org> - 2.0.1-4
- Rebuilt for
https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild
* Tue Sep 14 2021 Sahana Prasad <sahana(a)redhat.com> - 2.0.1-3
- Rebuilt with OpenSSL 3.0.0
* Thu Jul 22 2021 Fedora Release Engineering <releng(a)fedoraproject.org> - 2.0.1-2
- Rebuilt for
https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #2019879 - [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV
https://bugzilla.redhat.com/show_bug.cgi?id=2019879
[ 2 ] Bug #2063340 - libre-2.1.1 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063340
[ 3 ] Bug #2063450 - librem-2.0.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063450
[ 4 ] Bug #2063451 - baresip-2.0.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063451
--------------------------------------------------------------------------------
================================================================================
librem-2.0.0-1.el7 (FEDORA-EPEL-2022-c9b3f4b951)
Library for real-time audio and video processing
--------------------------------------------------------------------------------
Update Information:
# 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
--------------------------------------------------------------------------------
ChangeLog:
* Sun Mar 13 2022 Robert Scheck <robert(a)fedoraproject.org> 2.0.0-1
- Upgrade to 2.0.0 (#2063450)
* Thu Jan 20 2022 Fedora Release Engineering <releng(a)fedoraproject.org> - 1.0.0-3
- Rebuilt for
https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild
* Thu Jul 22 2021 Fedora Release Engineering <releng(a)fedoraproject.org> - 1.0.0-2
- Rebuilt for
https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #2019879 - [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV
https://bugzilla.redhat.com/show_bug.cgi?id=2019879
[ 2 ] Bug #2063340 - libre-2.1.1 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063340
[ 3 ] Bug #2063450 - librem-2.0.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063450
[ 4 ] Bug #2063451 - baresip-2.0.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=2063451
--------------------------------------------------------------------------------
================================================================================
zabbix40-4.0.39-1.el7 (FEDORA-EPEL-2022-bd2c412d62)
Open-source monitoring solution for your IT infrastructure
--------------------------------------------------------------------------------
Update Information:
Security fix for CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919
--------------------------------------------------------------------------------
ChangeLog:
* Sat Mar 12 2022 Orion Poplawski <orion(a)nwra.com> - 4.0.39-1
- Update to 4.0.39
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #2063280 - CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919
zabbix40: zabbix: Multiple security vulnerabilities [epel-all]
https://bugzilla.redhat.com/show_bug.cgi?id=2063280
--------------------------------------------------------------------------------
================================================================================
zabbix50-5.0.21-1.el7 (FEDORA-EPEL-2022-54fdcd70bd)
Open-source monitoring solution for your IT infrastructure
--------------------------------------------------------------------------------
Update Information:
Security fix for CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919
--------------------------------------------------------------------------------
ChangeLog:
* Sat Mar 12 2022 Orion Poplawski <orion(a)nwra.com> - 5.0.21-1
- Update to 5.0.21
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #2063282 - CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919
zabbix50: zabbix: Multiple security vulnerabilities [epel-all]
https://bugzilla.redhat.com/show_bug.cgi?id=2063282
--------------------------------------------------------------------------------
================================================================================
zchunk-1.2.1-1.el7 (FEDORA-EPEL-2022-0d8982b43c)
Compressed file format that allows easy deltas
--------------------------------------------------------------------------------
Update Information:
* Fix bug that prevented creating a zchunk file from a source that was larger
than 2GB * Fix memory leak
--------------------------------------------------------------------------------
ChangeLog:
* Sat Mar 12 2022 Jonathan Dieter <jdieter(a)gmail.com> - 1.2.1-1
- Fixed bug that limited size of file that could be compressed using zchunk to 2GB
- Fixed memory leak
--------------------------------------------------------------------------------