-------------------------------------------------------------------------------- Fedora Update Notification FEDORA-2021-7a98d8f780 2021-04-26 00:26:25.299618 --------------------------------------------------------------------------------
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.
-------------------------------------------------------------------------------- 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 -------------------------------------------------------------------------------- ChangeLog:
* Sat Apr 24 2021 Robert Scheck robert@fedoraproject.org 1.1.0-1 - Upgrade to 1.1.0 (#1953196) - Added upstream feature patch for GTK+ call history -------------------------------------------------------------------------------- References:
[ 1 ] Bug #1953196 - baresip-1.1.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=1953196 --------------------------------------------------------------------------------
This update can be installed with the "dnf" update program. Use su -c 'dnf upgrade --advisory FEDORA-2021-7a98d8f780' at the command 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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