7:27 p.m.
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Fedora Update Notification
FEDORA-2021-7a98d8f780
2021-04-26 00:26:25.299618
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Name : baresip
Product : Fedora 34
Version : 1.1.0
Release : 1.fc34
URL : https://github.com/baresip/baresip
Summary : Modular SIP user-agent with audio and video support
Description :
A modular SIP user-agent with support for audio and video, and many IETF
standards such as SIP, RTP, STUN, TURN, and ICE for both, IPv4 and IPv6.
Additional modules provide support for audio codecs like G.711, G.722,
G.726, GSM, L16, MPA, and Opus, audio drivers like ALSA, GStreamer, JACK
Audio Connection Kit, Portaudio, and PulseAudio, video codecs like VP8 or
VP9, video sources like Video4Linux and X11 grabber, video outputs like
SDL2 or X11, NAT traversal via STUN, TURN, ICE, NATBD, and NAT-PMP, media
encryption via SRTP or DTLS-SRTP, management features like embedded web-
server with HTTP interface, command-line console and interface, and MQTT.
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Update Information:
# baresip 1.1.0 * config: enable sip_verify_server if sip_cafile or
sip_capath is set (#1392) * Update README.md * config: add capath for
freebsd (#1389) * audio: prevent potential mutex deadlock (#1381) (#1383) *
ua: less restrictions for local account in uag_find_requri() (#1375) (#1384) *
avformat: use av_packet_alloc() instead of deprecated function (fixes #1372) *
avcodec: use av_packet_alloc() instead of deprecated function (#1380) * gtk:
check strncmp input (clang analyzer) * aufile: fix clang analyzer warnings
(#1378) * account: allow also ip;transport=tcp as valid dial URI (#1377) *
httpreq: fix clang analyzer warnings (#1368) (#1379) * multicast: remove clang
static analyzer warnings (#1374) * aac: remove unused code * avcapture: fix
macos 10.15 devicesWithMediaType deprecation warning (#1371) * ci: add extra
checks * ctrl_dbus: ignore return value from conf_get() * ausine: check
re_regex, fix formatting * httpeq: fix potential NULL dereference *
opus_multistream: fix clang warning * config: add empty newline between blocks
* coreaudio: check return value of AudioObjectGetPropertyDataSize * opus: no
need to increment pointer in last block * snapshot: return errorcode from
png_save_vidframe * vidloop: allocate frame before the filter loop *
avformat: register codecs for old versions of ffmpeg (fix #1363) (#1365) *
should fix #1364 * menu: fix wrong use of sizeof in clean_number() * menu:
check error in cmd_dialdir * menu: remove redundant code * video: check
input parameter in video_update() * gtk: fix wrong sizeof argument * ui: fix
gcc warning ingoring return value of fgets (#1361) (#1362) * ctrl_dbus: ensure
save synchronization (#1358) (#1360) * menu: print useful responses for
tlsissuer and tlssubject (#1357) (#1359) * evdev: init struct input_event to
zero * reg: fix potential NULL deref * gst: Use GstBus sync handler instead
of GMainLoop (#1356) * ua: fix warning when compiling with HAVE_INET6= *
call: fix clang warning (ref #1353) * ua: fix err is never read warning (make
clang) * reg: fix possible null pointer dereference (clang check) * event:
fix deadstore clang warning * snapshot: fix unknown warning option on OSX *
test: fix a warning from clang analyzer * call: remove tool attribute from SDP
(#1354) * audio: fix errors reported by make clang * account: align debug
variables * Document and enable rtcpsummary and snapshot modules (fixes #1341)
(#1343) * video: remove redundant code * changelog: update * gst:
formatting, two empty lines between functions * misc/include/objc/cpp/mk:
update copyright * cons: emulate key-release -- ref #1329 * Correct reverse
domain name notation (#1342) #1342 * gtk with account_uri_complete (#1339)
#1339 * bump version to 1.1.0 -- ref #1333 * ui: fix leaking of cmd_ctx
(#1338) #1338 * DTMF tones for A B C D (#1340) #1340 * account: use a fixed
username for the template * contact: update contacts template * config:
disable ctrl_dbus in config template * Module event (#1335) #1335 * add
event UA_EVENT_MODULE to tell to app when snapshot has been written (#1330)
#1330 * ringtone: generated busy and ringback tone (#1332) #1332 * audio:
prevent restart of rx_thread on call termination (#1331) #1331 * modules:
update auplay/ausrc modules * Auplay remove inheritance (#1328) #1328 *
h264: add doxygen comment * vidloop: add VIDEO_SRATE * vidloop: check error
* vidloop: add vidframe_clear * vidloop: split enable_codec into
encoder/decoder * Ausrc remove inheritance (#1326) #1326 * ua: remove prev
call (#1323) #1323 * sndfile: get number of bytes from auframe * plc: check
format of struct auframe * speex_pp: check format of struct auframe *
webrtc_aec: use format from struct auframe * README: update codecs and RFCs
* menu: use uri complete for command dialdir (#1321) #1321 * video: check for
video display before calling handler * Changed name and made public (#1319)
#1319 * menu: return call-id for dial and dialdir (#1320) #1320 * Fixes for
account uri complete (#1318) #1318 * Avoid compiler warnings: * Avoid
compiler warnings (I haven't found anything wrong with the code) * vidfilt:
fix warning * vidfilt: split parameters into encode/decode * snapshot: fix
warnings * video: group functions from vidutil.c * avfilter: fix warnings
* vumeter: use format from audio frame * replaced ua_uri_complete with
account_uri_complete (#1317) #1317 * aulevel: move to librem * omx: fix
warning * vidisp: remove inheritance (#1316) #1316 * docs: change video
settings to match the default values (#1315) #1315 * menu: select call in
cmd_find_call() (#1314) #1314 * menu: use menu_stop_play() (#1311) #1311 *
main: unload app modules in signal handler (#1310) #1310 * avformat: replace
const double with double * avformat: clean up ifdefs (#1313) #1313 * ci:
drop ubuntu 16.04 support - end of life * avformat: proper code formatting *
avcodec: add avcodec prefix to log messages * avcodec: check length of H265
packet * x11grab: remove vidsrc inheritance * v4l2: remove vs inheritance
* vidsrc: remove concept of baseclass/inheritance * ua,menu: remove
uag_find_call_state (#1304) #1304 * Updated homepage * sdl: correct aspect-
ratio in fullscreen mode * vidloop: add vidisp parameters * auloop: use
auframe_size * audio: use auframe_size * Auplay use auframe (#1305) #1305
* Docs examples config (#1302) #1302 * Serreg fixes (#1301) #1301 * Update
config.c #1303 * contact: use uag_find_requri() * ua: use new tls function
to set cafile and path #1300 * config: add sip_capath config line * Call
event answered fixes alsa issue (#1299) #1299 * ctrl_dbus: send DBUS signal
when dbus interface is ready (#1296) #1296 * Multicast call priority (#1291)
#1291 * Menu fixes for play tones2 (#1294) #1294 * gst: add missing include
unistd.h #1297 * multicast: cleanup function description and fix doxygen
warning (#1292) #1292 * menu: remove call resume for command hangup (#1289)
#1289 * ua: add a generic filter API for calls (#1293) #1293 * Merge pull
request #1288 from cspiel1/remove_call_resume_on_termination #1288 * menu:
remove call resume on termination * multicast: fix build error when using
HAVE_PTHREAD= * alsa_play.c add suggestion to use dmix (#1283) #1283 *
readme.md: added multicast module (#1282) #1282 * audiounit: fix typo *
update copyright year (#1287) #1287 * config cleanup (#1286) #1286 * update
copyright year (#1285) #1285 * conf: add call_hold_other_calls config option
(#1280) #1280 * config.c: added rtmp to config template (#1284) #1284 *
main.c: update year #1281 * The avformat_decoder should be optional (#1277)
#1277 * src/audio: set started false with audio_stop (#1278) #1278 * readme:
update baresip fork links * ausine: mono support and stereo_left/right option
#1274 * menu: fix incoming calls are not selected on call termination (#1271)
#1271 * test: remove mock_aucodec, using g711 instead * opengl: remove
deprecated module (#1268) #1268 * Added account_dtmfmode and
account_set_dtmfmode API functions (#1269) #1269 * avcodec: remove support for
MPEG4 codec * call: start streams asynchronously (issue #1261) (#1267) #1267
* audio: remove special handling of Comfort Noise * multicast: fix one doxygen
warning * menu: update doxygen comment * menu: correct hangupall command for
parallel call feature (#1264) #1264 * menu: on call termination select another
active call (#1260) #1260 * ua: correct doxygen of uag_hold_resume() #1262 *
menu: simplify cmd_hangupall() (#1259) #1259 * support for sending of DTMF
INFO (#1258) #1258 * Menu optional call parameter (#1254) #1254 * cleanup
tabs and spaces #1256 * ua: correct doxygen for uag_hold_others() * ua: add
doxygen for call find functions * menu: add doxygen to cmd_hangup(),
cmd_hold(), cmd_resume() * menu: command accept searches all User-Agents for
an incoming call * ua: add function uag_find_call_state() * menu: print
correct warning for hangup, accept, hold, resume * menu: add optional
parameter call-id to cmd_call_resume() * menu: add optional parameter call-id
to cmd_call_hold() * menu: add optional parameter call-id to cmd_hangup() *
menu: add optional parameter call-id to cmd_answerdir() * menu: add utility
function that decodes complex command parameters * menu: use SDP_SENDRECV for
cmd_answerdir() as fallback * menu: add optional parameter call-id to
cmd_answer() * ua: add call find per call-id function * call: call_info()
prints also the call-id * ua: in ua_print_calls() print User-Agent info in
header * menu: ua NULL check for answer command * replace spaces with tab
#1249 * removed newline * undid httpreq spacing * fixed line too long *
moved multicast template to end of config template * ua: fix uag_hold_others
use of wrong list element #1253 * added multicast enabled message (#1251)
#1251 * updated date and added multicast to signaling (#1252) #1252 * Merge
pull request #1248 from webstean/patch-2 #1248 * Added newline to multicast
comment * Menu ensure only one established call (#1247) #1247 * Call resume
on hangup (#1246) #1246 * menu: for call answer search all UAs for calls to
put on hold * ua: ua_answer() should answer same call like ua_hold_answer()
* ua: make ua_find_call_state() global usable * Add multicast_listener to
config template (#1245) #1245 * Update config template to include multicast
module (#1244) #1244 * menu: if a call becomes established then put others on
hold * ua: add uag_hold_others() * Fix multiple resumed calls (#1242) #1242
* Merge pull request #1241 from cHuberCoffee/cmd_hangupall #1241 * RFC: Make
avformat decode mjpeg v4l2 with vaapi (#1216) #1216 * ua: add doxygen for new
uag_hold_resume() * menu: fix missing callid of menu at call closed * menu:
use uag_hold_resume to ensure only one active call * ua: on call resume check
for other active calls * menu: new hangupall command with direction parameter
* readme: update supported compilers and ssl libs * menu: fix redial * Fix
spaces * Multicast module (#1231) #1231 * menu: use print backend pointer pf
correctly (#1222) #1222 * menu: start ringback only once for parallel calls
(#1238) #1238 * jack: support port pattern in config file (#1237) #1237 *
config: disables server verification if sip_verify_server is missing (#1236)
#1236 * ua: for UA selection allow arbitrary aor for regint=0 accounts (#1234)
#1234 * Ctrl dbus synchronize (#1232) #1232 * event: encode also remote
audio direction (#1227) #1227 * Merge pull request #1235 from
cspiel1/event_add_string_for_UA_EVENT_CUSTOM #1235 * event: add string for
UA_EVENT_CUSTOM * Mimic ifdef on avutil version for hwcontext * Fix to tabs
and improve checks * src/config: show sip_cafile warning only if
sip_verify_server is enabled * Avoid compiler warnings using casts #1228 *
test: disable SIP TLS server verification #1224 * config,ua: add config flag
disable SIP TLS server verification * alsa/play: snd_pcm_writei error codes
are negative * alsa: fix clang warnings "conversion loses integer precision"
#1223 * Intelligent call answer (#1218) #1218 * Remove uag next (#1207)
#1207 * Merge pull request #1219 from cspiel1/message_reply_once #1219 *
menu: update switch_audio_player * Make vaapi/mjpeg options of avformat *
src/config: no sip_cafile wording * message: reply only once * src/ua: only
warn if tls_add_ca fails, same as undefined cafile #1214 * src/config: add
sip_cafile warning and enable by default * ua: change log message from warning
to info * video: fix video payload text * Make avformat decode mjpeg v4l2
with vaapi * ua: improve UA selection for incoming calls (#1206) #1206 * ua:
limit account matches for incoming calls to non-registrar accounts * ua: check
for NULL parameter in uag_find_msg() * ua: early exit for AF_UNSPEC in
uri_match_af() * ua: use sip_transp_decode() in uri_match_transport() * ua:
use arrays in uri_host_local() * test: add test for deny UDP peer-to-peer call
* ua: improve UA selection for incoming calls * Sip message to application
(#1201) #1201 * opus: Ensure (re)init of fmtp strings (#1209) #1209 *
ctrl_dbus: generate dbus interface during build (#1208) #1208 * mod_gtk:
switch to gtk 3 (#1203) #1203 * menu: set_answer_mode: apply all uas * menu:
find_call: search all user-agents * menu: fix usage of ua * isac: remove
deprecated module (#1204) #1204 * menu: cmd_print_calls: print all uas * Fix
interaction between CLI menu and GTK menu (#1202) #1202 * menu: rename
menu_current() to menu_uacur() * webrtc_aec: fix compilation with gcc 4.9 (fix
#1193) * win32: add cons module, fixes #1197 * ua: remove ua_aor() -- use
account_aor() instead * gtk: use account_aor() * menu: use account_aor() *
presence: use account_aor() * modules: use account_aor() * account: fix
video codes decode (#1196) #1196 * core: use account_aor() * Merge pull
request #1198 from baresip/av1 #1198 * Avoid unused parameter warning *
debug_cmd: add UA_EVENT_CUSTOM (#1194) #1194 * fix decoder changed debug text
* cairo: minor debug tuning * menu: add uadelall to delete all user agents
#1195 * use account_aor() * mctrl: remove support for media-control
(deprecated) * update doxygen comments * ua: minor cleanup * ua: split
struct uag from instance * README: add RFC 5373 * menu: fix segfault on last
account deletion (#1192) #1192 * call: extend SIP auto answer support for
incoming calls (#1191) #1191 * Sip auto answer caller (#1188) #1188 * win32:
remove timer.c * ua: give a nice name to 'global' struct * ua: remove ua_cur
* move uag_current to menu module * menu: pass ua from mqtt to menu via opaque
data * Sip autoanswer callee (#1187) #1187 * ua: for answer-mode early also
send INCOMING event (#1185) #1185 * gst: The error handler call for end of
stream is now (#1182) #1182 * mk: also detect mqtt.so in SYSROOT_ALT *
contact: add ua_lookup_domain * video: minor tuning of pipeline text * gst:
playback of read only audio files failed (#1183) #1183 * gtk: make a local
pointer to current ua * menu: clean up usage of uag_current() * call:
correction of remote video direction info at SDP-offer (#1181) #1181 *
debug_cmd: print all user-agents * presence: one command with status as
argument * ua: rename presence status to pstat * ua: remove
LIBRE_HAVE_SIPTRACE check, always enabled * update doxygen comments * mk:
update doxygen config file * menu: initialize menu with zeros (#1179) #1179
* Re mk cross build2 (#1161) #1161 * net: make fallback DNS ignored message
debug only * mixausrc: improve logging #1176 * mixausrc: fix
shorten-64-to-32 warnings * config: template for osx/ios * Supressed clang
zero length array warning * Added ctx param to video_stop/video_stop_source
and set ctx to null (#1173) #1173 * avformat: add empty line after base class
* Make macos warnings into errors (#1171) #1171 * disable mixausrc until
warnings are fixed * clang shorten-64-to-32 warnings (#1170) #1170 *
Mixausrc (#1159) #1159 * aufile: fix warning on OSX * alsa: print warning if
running, fixed #1162 * Don't default stunuser/pass to account authuser/pass
(#1164) #1164 * Audio file info (#1157) #1157 * gitignore: clangd cache,
compile_commands.json and cleanup * Merge pull request #1167 from
baresip/video_display #1167 * Reordered video_stop_display * Expose
video_stop_display() to API * Video dir rename (#1158) #1158 * ci: use
baresip/rem repo * stream: add function to send a RTP dummy packet (#1156)
#1156 * Play aufile extended support (#1155) #1155 * video: move video
related start/stop/update into video file (#1151) #1151 * aufile: add audio
player to write speaker data to wav file (#1153) #1153 * Fix compiler warnings
(#1152) #1152 * play: fix warning * play ausrc (#1147) #1147 * README: add
more status badges * README: replace travis status badge * menu: fix
uint16_t scode #1149 * config: revert dirent.h changes * audio: fix
HAVE_PTHREAD audio_destructor * gst ready for file play (#1148) #1148 *
debug_cmd: mem_deref of player fixes segfault (#1146) #1146 * net: remove
deprecated net_domain() * update contact examples * fix freeze on hangup
(#1135) (#1145) #1145 * menu: make audio files configurable (#1144) #1144 *
aptx: declare variable outside for-loop * fix warnings on openbsd * jack:
declare variable outside for loop * account: declare variable outside for loop
* coreaudio: declare variable outside for loop * menu: initialize menu.play
fixes segfault (#1143) #1143 * ausine: declare variable outside for loop *
timer: remove tmr_jiffies_usec (replaced by libre) (#1141) #1141 * Adaptive
jbuf (#1112) #1112 * Update build.yml (#1140) #1140 * mqtt: allow to
separate pub from sub topic base (#1139) #1139 * video: fix warning * mqtt:
fix printing port and add tls support (#1138) #1138 * httpreq: in cmd_setauth
check if parameter was given (#1134) #1134 * Merge pull request #1132 from
baresip/pr-dependency-action #1132 * ci: add pull request dependency checkouts
* audio: remove redundant union * menu: use menu_ as prefix for global symbols
* menu: use menu_ as prefix for global symbols * ci: add apt-get update *
menu: module refactoring (#1129) #1129 * audio, video, stream: check payload
type before put to jbuf (#1128) #1128 * Cmd dialdir (#1126) #1126 * Cmd
acceptdir (#1125) #1125 * event: add register fallback to event string and
class name (#1124) #1124 * avformat: use %u for unsigned * modify event type
and check if peeruri null (#1119) #1119 * event: move code from ua.c (#1118)
#1118 * Valgrind ci (#1117) #1117 * h264 cleanup, second part (#1115) #1115
* h264 cleanup (#1114) #1114 * Merge pull request #1113 from baresip/github-
actions-v2 #1113 * ci: remove travis * ci: add github actions - replaces
travisci * qtcapture: remove deprecated module (#1107) #1107 * test: prepare
for dualstack * test: add mock dns_server_add_aaaa * make EXTRA_MODULES
last, not first (#1106) #1106 * httpreq: fix cmd_settimeout * test: bind
network to localhost, a fix for #1090 * modules/webrtc_aec: link flags fixes
(#1105) #1105 * menu: commands in alphabetical order * httpreq: fix warning
about unused args * serreg: fix warnings about unused argument * menu: fix
warnings about unused argument * Add a HTTP request module with authorization
(#1099) #1099 * Menu: corrections for ring tones and call status by means of a
global call counter (#1102) #1102 * mk: remove dirent.h * Updating .vcxproj
file for windows builds (#1097) #1097 * ccheck: change license to BSD license
* Merge pull request #1095 from baresip/websocket #1095 * Serial registration
(#1083) #1083 * Ctrl dbus (#1085) #1085 * README: remove references to
creytiv.com * Branch of baresip that includes Alfred's sip websocket patch *
Merge pull request #1091 from baresip/debian #1091 * ua, menu: new command to
print certificate issuer and subject (#1078) #1078 * .gitignore: add ctags and
Vim swp files (#1084) #1084
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ChangeLog:
* Sat Apr 24 2021 Robert Scheck <robert(a)fedoraproject.org> 1.1.0-1
- Upgrade to 1.1.0 (#1953196)
- Added upstream feature patch for GTK+ call history
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References:
[ 1 ] Bug #1953196 - baresip-1.1.0 is available
https://bugzilla.redhat.com/show_bug.cgi?id=1953196
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line. For more information, refer to the dnf documentation available at
http://dnf.readthedocs.io/en/latest/command_ref.html#upgrade-command-label
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