Fedora EPEL 8 updates-testing report
by updates@fedoraproject.org
The following Fedora EPEL 8 Security updates need testing:
Age URL
16 https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-17ae719cb2 syncthing-1.18.6-3.el8
6 https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-52a1bafe29 chromium-99.0.4844.51-1.el8
The following builds have been pushed to Fedora EPEL 8 updates-testing
abcm2ps-8.14.13-1.el8
baresip-2.0.0-1.el8
devilspie2-0.44-1.el8
globus-gssapi-gsi-14.17-4.el8
libre-2.1.1-1.el8
librem-2.0.0-1.el8
python-paramiko-2.4.3-2.el8
zabbix40-4.0.39-1.el8
zchunk-1.2.1-1.el8
Details about builds:
================================================================================
abcm2ps-8.14.13-1.el8 (FEDORA-EPEL-2022-0dca326d43)
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.el8 (FEDORA-EPEL-2022-c601dc0ed1)
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
--------------------------------------------------------------------------------
================================================================================
devilspie2-0.44-1.el8 (FEDORA-EPEL-2022-d1a278412c)
A window-matching utility
--------------------------------------------------------------------------------
Update Information:
EPEL8 init at 0.44
--------------------------------------------------------------------------------
ChangeLog:
* Fri Mar 11 2022 Michal Min���� <miminar(a)redhat.com> - 0.44-1
- EPEL8 init at 0.44
--------------------------------------------------------------------------------
References:
[ 1 ] Bug #1852502 - [EPEL8] Please build devilspie2 for EPEL8
https://bugzilla.redhat.com/show_bug.cgi?id=1852502
[ 2 ] Bug #2042469 - Request to add devilspie2 package to epel 8
https://bugzilla.redhat.com/show_bug.cgi?id=2042469
--------------------------------------------------------------------------------
================================================================================
globus-gssapi-gsi-14.17-4.el8 (FEDORA-EPEL-2022-e45f3f52bf)
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.el8 (FEDORA-EPEL-2022-c601dc0ed1)
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.el8 (FEDORA-EPEL-2022-c601dc0ed1)
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
--------------------------------------------------------------------------------
================================================================================
python-paramiko-2.4.3-2.el8 (FEDORA-EPEL-2022-ad126686cf)
SSH2 protocol library for python
--------------------------------------------------------------------------------
Update Information:
CVE-2022-24302: Creation of new private key files using `~paramiko.pkey.PKey`
subclasses was subject to a race condition between file creation and mode
modification, which could be exploited by an attacker with knowledge of where
the Paramiko-using code would write out such files; this has been patched by
using `os.open` and `os.fdopen` to ensure new files are opened with the correct
mode immediately (we've left the subsequent explicit 'chmod' in place to
minimize any possible disruption).
--------------------------------------------------------------------------------
ChangeLog:
* Sun Mar 13 2022 Paul Howarth <paul(a)city-fan.org> - 2.4.3-2
- Security fix backported from 2.10.1
- CVE-2022-24302: Creation of new private key files using
'~paramiko.pkey.PKey' subclasses was subject to a race condition between
file creation and mode modification, which could be exploited by an
attacker with knowledge of where the Paramiko-using code would write out
such files; this has been patched by using 'os.open' and 'os.fdopen' to
ensure new files are opened with the correct mode immediately (we've left
the subsequent explicit 'chmod' in place to minimize any possible
disruption, though it may get removed in future backwards-incompatible
updates)
--------------------------------------------------------------------------------
================================================================================
zabbix40-4.0.39-1.el8 (FEDORA-EPEL-2022-d431be322b)
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
* Wed Feb 23 2022 Orion Poplawski <orion(a)cora.nwra.com> - 4.0.38-1
- Update to 4.0.38 (CVE-2022-23132, CVE-2022-23133, CVE-2022-23134)
- Fix up bundled provides
--------------------------------------------------------------------------------
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
--------------------------------------------------------------------------------
================================================================================
zchunk-1.2.1-1.el8 (FEDORA-EPEL-2022-77842bdac8)
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
--------------------------------------------------------------------------------