cmake: add pulse and pulse_async module #1919
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Jul 18, 2022
Still needs more testing for all the different options and plugins. Changelog: == 2.5.0 - 2022-07-01 == What's Changed * audio: add optional decoding buffer by @cspiel1 in baresip/baresip#1842 * audio: RX filter thread needs separate sampv buffer by @cspiel1 in baresip/baresip#1879 * aufile: fix possible data race warning by @cspiel1 in baresip/baresip#1880 * audiounit,coreaudio: fix kAudioObjectPropertyElementMaster deprecation by @sreimers in baresip/baresip#1881 * av1: explicitly check for supported OBU types by @alfredh in baresip/baresip#1882 * audiounit/coreaudio: fix kAudioObjectPropertyElementMain by @sreimers in baresip/baresip#1885 * ci/build: bump macos min. sdk to 10.12 by @sreimers in baresip/baresip#1883 * ci: run only for pull requests and main branch by @sreimers in baresip/baresip#1887 * multicast: C11 mutex by @alfredh in baresip/baresip#1892 * dtls_srtp: enable ECC by default, remove RSA by @alfredh in baresip/baresip#1891 * ci/build: add ubuntu 22.04 by @sreimers in baresip/baresip#1890 * test: add check for memory leaks by @sreimers in baresip/baresip#1896 * stream,metric: RX real-time - make metric thread-safe by @cspiel1 in baresip/baresip#1895 * Cmake findre by @alfredh in baresip/baresip#1893 * test: wait for both audio and video to be established by @alfredh in baresip/baresip#1903 * docs: remove old TODO file by @alfredh in baresip/baresip#1902 * audio: fixed check for aubuf started flag by @cspiel1 in baresip/baresip#1904 * use new mutex interface by @cspiel1 in baresip/baresip#1905 * audio: make rx.filtl thread-safe by @cspiel1 in baresip/baresip#1897 * audio: allocate correct buffer size for static auplay srate by @cspiel1 in baresip/baresip#1906 * Pulseaudio Async Interface Module by @cHuberCoffee in baresip/baresip#1907 * Do not destroy register client when it is unregistered by @juha-h in baresip/baresip#1908 * Two spaces are required after email address by @juha-h in baresip/baresip#1909 * cmake: add alsa module by @alfredh in baresip/baresip#1910 * cmake: fix static openssl and thread linking by @sreimers in baresip/baresip#1911 * In start_registering, create register clients if reg list is empty by @juha-h in baresip/baresip#1913 * ctrl_dbus: use new thread and mtx interface by @cspiel1 in baresip/baresip#1916 * cmake: add pulse and pulse_async module by @cHuberCoffee in baresip/baresip#1919 * Un-subscribe mwi at un-register by @juha-h in baresip/baresip#1918 * call: update media on session progress. by @RobertMi21 in baresip/baresip#1922 * ctrl_dbus send event in main thread by @cspiel1 in baresip/baresip#1921 * uag: add timestamps to SIP trace by @cspiel1 in baresip/baresip#1914 * main: fix open timers check by @sreimers in baresip/baresip#1925 * cmake: add account module by @alfredh in baresip/baresip#1926 --- == 2.4.0 - 2022-06-01 == What's Changed * mulitcast unmute bad quality by @cspiel1 in baresip/baresip#1821 * menu ringback for parallel call by @cspiel1 in baresip/baresip#1827 * multicast: support error code EAGAIN of jbuf_get() by @cspiel1 in baresip/baresip#1832 * use RTP clock rate for timestamp calculation by @cspiel1 in baresip/baresip#1834 * av1 obu by @alfredh in baresip/baresip#1835 * av1 packetizer by @alfredh in baresip/baresip#1836 * av1: depacketizer by @alfredh in baresip/baresip#1837 * Disabled debug statement by @juha-h in baresip/baresip#1838 * h264: move from rem to re by @sreimers in baresip/baresip#1839 * ua: send new event UA_EVENT_CREATE at successful ua allocation by @cHuberCoffee in baresip/baresip#1840 * evdev: fix wrong ioctl size by @sreimers in baresip/baresip#1843 * aufile: ausrc_prm has to be copied when source is allocated by @cspiel1 in baresip/baresip#1844 * conf: missing pointer initialization found by clang analyzer by @cspiel1 in baresip/baresip#1845 * mk/modules: fix omx RPI detection by @sreimers in baresip/baresip#1847 * auconv: add auconv_to_float (fixes #1833) by @alfredh in baresip/baresip#1849 * avfilter: migrate to C11 mutex by @alfredh in baresip/baresip#1850 * avformat: C11 mutex by @alfredh in baresip/baresip#1851 * selfview: C11 mutex by @alfredh in baresip/baresip#1852 * audio: C11 mutex by @alfredh in baresip/baresip#1853 * metric: C11 mutex by @alfredh in baresip/baresip#1854 * play: C11 mutex by @alfredh in baresip/baresip#1855 * dns: add query cache by @sreimers in baresip/baresip#1848 * video: C11 mutex by @alfredh in baresip/baresip#1856 * aufile: C11 threads by @alfredh in baresip/baresip#1858 * audio: add more locking by @alfredh in baresip/baresip#1857 * aufile/play: fix run data race by @sreimers in baresip/baresip#1859 * mc: multicast receiver enable state fix by @cHuberCoffee in baresip/baresip#1861 * audio: C11 thread by @alfredh in baresip/baresip#1860 * av1: add packetize handler by @alfredh in baresip/baresip#1865 * net/net_debug: add default route hint by @sreimers in baresip/baresip#1864 * ice: fix local prio calculation by @sreimers in baresip/baresip#1863 * avformat: open codec if not passthrough by @alfredh in baresip/baresip#1866 * dtls_srtp: Minor whitespace fix by @robert-scheck in baresip/baresip#1870 * vp8: add packetize handler by @alfredh in baresip/baresip#1868 * vp9: add packetizer by @alfredh in baresip/baresip#1871 * debug_cmd: support absolute path for command aufileinfo by @cspiel1 in baresip/baresip#1875 * event: add diverter URI to UA event by @cspiel1 in baresip/baresip#1876 * aufileinfo with synchronous response by @cspiel1 in baresip/baresip#1877 **Full Changelog**: baresip/baresip@v2.3.0...v2.4.0 --- == [2.3.0] - 2022-05-01 == What's Changed * mc: multicast mute function by @cHuberCoffee in baresip/baresip#1805 * mc: reject incoming call if high prio multicast is received by @cHuberCoffee in baresip/baresip#1804 * mc: mcplayer stream fade-out and fade-in by @cHuberCoffee in baresip/baresip#1802 * clean_number now will remove all non-digit chars by @mbattista in baresip/baresip#1806 * Workflows cmakelint by @alfredh in baresip/baresip#1808 * ccheck: check all CMakeLists.txt files by @sreimers in baresip/baresip#1810 * mk: remove win32 MSVC project files by @alfredh in baresip/baresip#1811 * cmake: add modules by @sreimers in baresip/baresip#1812 * ajb,aubuf: timestamp is given in [us] by @cspiel1 in baresip/baresip#1809 * call: allow optional leading space in SIP INFO for dtmf-relay by @thomas-karl in baresip/baresip#1814 * conf: add fs_file_extension() by @alfredh in baresip/baresip#1816 * Updated debian version by @juha-h in baresip/baresip#1817 * pulse: fix timestamp integer overrun for arm by @cspiel1 in baresip/baresip#1818 * fix audio multicast artefacts by @cspiel1 in baresip/baresip#1819 * audio: flush aubuf if ssrc changes by @cspiel1 in baresip/baresip#1822 * Debian control dependency update by @juha-h in baresip/baresip#1823 * pulse: support restart of pulseaudio during stream by @cspiel1 in baresip/baresip#1824 * version 2.3.0 by @alfredh in baresip/baresip#1826 == New Contributors * @thomas-karl made their first contribution in baresip/baresip#1814 --- == [2.0.2] - 2022-04-09 == What's Changed * Added API function call_diverteruri by @juha-h in baresip/baresip#1780 * Avoid undeclared 'CLOCK_REALTIME' on RHEL/CentOS 7 (fixes #1781) by @robert-scheck in baresip/baresip#1782 * audio: add lock in audio_send_digit by @GGGO in baresip/baresip#1786 * vumeter: use new auframe_level() by @sreimers in baresip/baresip#1788 * reg.c: use already declared acc by @GGGO in baresip/baresip#1789 * aubuf adaptive jitter buffer by @cspiel1 in baresip/baresip#1784 * multicast set aubuf silence by @cspiel1 in baresip/baresip#1791 * ccheck: fix line number in error print by @cspiel1 in baresip/baresip#1793 * test: check the correct stream in UA_EVENT_CALL_MENC by @alfredh in baresip/baresip#1794 * audio: missing lock around stream_send by @GGGO in baresip/baresip#1796 * docs: remove obsolete jitter_buffer_wish from config example by @cspiel1 in baresip/baresip#1798 * Multicast jbuf and aubuf changes by @cHuberCoffee in baresip/baresip#1797 * uag: uag_hold_resume() should not return err if there is no call to hold by @cspiel1 in baresip/baresip#1799 * stream: remove mbuf_get_left check in rtp_handler by @GGGO in baresip/baresip#1801 * cmake: preliminary support by @alfredh in baresip/baresip#1800 == New Contributors * @GGGO made their first contribution in baresip/baresip#1786 --- == [2.0.1] - 2022-03-27 === What's Changed * audio: fix rx_thread (adaptive jitter buffer) by @sreimers in baresip/baresip#1769 * test: init fixture by @alfredh in baresip/baresip#1772 * test: refactoring of test_account_uri_complete by @alfredh in baresip/baresip#1773 * mk: check also if extensions/XShm.h is present by @cspiel1 in baresip/baresip#1774 * menu: support custom SIP headers by @cspiel1 in baresip/baresip#1775 * menu: use new sdp_dir_decode by @cspiel1 in baresip/baresip#1776 * menu: avoid multiple hash entries with same key by @cspiel1 in baresip/baresip#1777 * menu: support audio file config value "none" by @cspiel1 in baresip/baresip#1778 * intercom: add video preview call by @cspiel1 in baresip/baresip#1779 --- == [2.0.0] - 2022-03-11 === What's Changed * debug_cmd: use module_event() for aufileinfo events by @cspiel1 in baresip/baresip#1345 * multicast: use module_event() for sending events by @cspiel1 in baresip/baresip#1346 * ctrl_dbus: use module_event() to send exported event by @cspiel1 in baresip/baresip#1347 * ua,call: add CALL_EVENT_OUTGOING by @cspiel1 in baresip/baresip#1348 * GTK caller history by @mbattista in baresip/baresip#1350 * Convert FRITZ!Box XML phone book into Baresip contacts by @robert-scheck in baresip/baresip#1382 * menu: play ringtone on audio_alert device by @cspiel1 in baresip/baresip#1396 * menu: use str_isset() for command parameter by @cspiel1 in baresip/baresip#1397 * dtls_srtp: use elliptic curve cryptography by @cHuberCoffee in baresip/baresip#1385 * Support for s16 playback in jack. Needed for play tones by @srperens in baresip/baresip#1399 * Check that account ;sipnat param has valid value by @juha-h in baresip/baresip#1401 * Tls sipcert per acc by @cHuberCoffee in baresip/baresip#1376 * Vidsrc add packet handler by @alfredh in baresip/baresip#1402 * ToS for video and sip by @cspiel1 in baresip/baresip#1393 * account: add accounts parameter to force media address family by @cspiel1 in baresip/baresip#1395 * Selective early media by @cspiel1 in baresip/baresip#1398 * ua,uag: split ua.c and uag.c by @cspiel1 in baresip/baresip#1349 * Account media af template by @cspiel1 in baresip/baresip#1406 * account: add missing client certificate parameter to template by @cHuberCoffee in baresip/baresip#1408 * account: update answermode values in template by @cspiel1 in baresip/baresip#1405 * menu: command uafind raises UA to head by @cspiel1 in baresip/baresip#1407 * ctrl_dbus: fix possible memleak on failed initialization by @cspiel1 in baresip/baresip#1410 * video passthrough by @alfredh in baresip/baresip#1418 * menu: enable auto answer calls also for command dialdir by @cspiel1 in baresip/baresip#1412 * menu: add command for settings media local direction by @cspiel1 in baresip/baresip#1413 * Accounts addr params by @cspiel1 in baresip/baresip#1414 * Accounts example cleanup by @cspiel1 in baresip/baresip#1415 * menu,call: fix hangup for outgoing call by @cspiel1 in baresip/baresip#1417 * multicast: add source and player API calls by @cHuberCoffee in baresip/baresip#1403 * menu: add command /uareg by @alfredh in baresip/baresip#1421 * menu: return complete URI for commands dial,dialdir by @cspiel1 in baresip/baresip#1424 * menu: in command dialdir call uag_find_requri() with uri by @cspiel1 in baresip/baresip#1425 * gst: replace variable length array (buf) with mem_zalloc by @sreimers in baresip/baresip#1426 * menu: avoid possible memleaks for dial/dialdir commands by @cspiel1 in baresip/baresip#1430 * uag: use local cuser for selecting user-agent (#1433) by @cspiel1 in baresip/baresip#1434 * Work on Intercom module by @cspiel1 in baresip/baresip#1432 * Attended Transfer on GTK by @mbattista in baresip/baresip#1435 * Update README.md with configuration suggestion by @webstean in baresip/baresip#1438 * README fixes by @juha-h in baresip/baresip#1440 * Accounts examples and template by @cspiel1 in baresip/baresip#1441 * serreg: use a timer for registration restart by @cspiel1 in baresip/baresip#1445 * gst: audio playback not correct for some WAV files. by @RobertMi21 in baresip/baresip#1442 * Working on intercom (ringtone override) by @cspiel1 in baresip/baresip#1436 * Use line number 0 if user did not provide any line number by @negbie in baresip/baresip#1451 * AMR Bandwidth Efficient mode support by @srperens in baresip/baresip#1423 * Working on Intercom (menu: allow other modules to reject a call) by @cspiel1 in baresip/baresip#1437 * auframe: add samplerate and channels by @sreimers in baresip/baresip#1452 * account: comment out very basic example in template by @cspiel1 in baresip/baresip#1458 * call answer media dir by @cspiel1 in baresip/baresip#1449 * Account auto answer beep by @cspiel1 in baresip/baresip#1461 * serreg: unregister correct User-Agents on registration failure by @cspiel1 in baresip/baresip#1462 * mk: enable auto-detect of av1 module by @alfredh in baresip/baresip#1463 * ctrl dbus makefile depends by @cspiel1 in baresip/baresip#1457 * stream: check if media is present before enabling the RTP timeout by @cspiel1 in baresip/baresip#1465 * ctrl_dbus: generate dbus code and documentation in makefile by @cspiel1 in baresip/baresip#1456 * auframe: always set srate and ch by @janh in baresip/baresip#1468 * auto answer beep per alert info URI by @cspiel1 in baresip/baresip#1466 * auframe: move to rem by @sreimers in baresip/baresip#1470 * mixminus: add conference feature by @sreimers in baresip/baresip#1411 * vidbridge: check vidbridge_disp_display args fixes segfault by @sreimers in baresip/baresip#1471 * gst: fixed some memory leaks by @RobertMi21 in baresip/baresip#1476 * ua, menu: move auto answer delay handling to menu (#1474) by @cspiel1 in baresip/baresip#1475 * ua,menu: move handling of ANSWERMODE_AUTO to menu (#1474) by @cspiel1 in baresip/baresip#1478 * ausine: support for multiple samplerates by @alfredh in baresip/baresip#1479 * account: fix IPv6 only URI for account_uri_complete() by @cspiel1 in baresip/baresip#1472 * ilbc: remove deprecated module by @alfredh in baresip/baresip#1483 * aubridge/device: remove unused sampv_out (old resample code) by @sreimers in baresip/baresip#1484 * pkg-config version check by @sreimers in baresip/baresip#1481 * mk: support more locations for libre.pc and librem.pc by @cspiel1 in baresip/baresip#1486 * net: remove unused domain by @alfredh in baresip/baresip#1489 * audio: fix aufilt_setup update handling by @sreimers in baresip/baresip#1498 * SIP redirect callbackfunction by @cHuberCoffee in baresip/baresip#1495 * add secure websocket tls context by @sreimers in baresip/baresip#1499 * test: add stunuri by @alfredh in baresip/baresip#1503 * turn: refactoring, add compv by @alfredh in baresip/baresip#1505 * fmt: add string to bool function by @cspiel1 in baresip/baresip#1501 * mk: check glib-2.0 at least like in ubuntu 18.04 by @cspiel1 in baresip/baresip#1507 * registration fixes by @cspiel1 in baresip/baresip#1510 * uag,menu: add commands to enable/disable UDP/TCP/TLS by @cspiel1 in baresip/baresip#1502 * config,audio: add setting audio.telev_pt by @cspiel1 in baresip/baresip#1509 * stream: fix telephone event (#1494) by @cspiel1 in baresip/baresip#1506 * Fix I2S compile error, use auframe by @andreaswatch in baresip/baresip#1512 * ci/tools: fix pylint by @sreimers in baresip/baresip#1515 * config: not all audio config was printed by @cspiel1 in baresip/baresip#1516 * net: replace network_if_getname with net_if_getname by @sreimers in baresip/baresip#1518 * account: add setting audio payload type for telephone-event by @cspiel1 in baresip/baresip#1517 * uag,menu: simplify transport enable/disable and support also ws/wss by @cspiel1 in baresip/baresip#1514 * rst: remove deprecated module by @alfredh in baresip/baresip#1519 * turn: add TCP and TLS transports by @alfredh in baresip/baresip#1520 * speex_pp: remove deprecated module by @alfredh in baresip/baresip#1521 * call: allow video calls by only rejecting a call without any common codecs by @cHuberCoffee in baresip/baresip#1523 * multicast: add missing join for multicast addresses by @cHuberCoffee in baresip/baresip#1524 * confg,uag: rework on sip_transports setting by @cspiel1 in baresip/baresip#1525 * ua: check if peer is capable of video for early video by @cHuberCoffee in baresip/baresip#1526 * mqtt/subscribe: replace fixed command buf and increase response size by @sreimers in baresip/baresip#1527 * mqtt: add reconnect handling (lost broker connection) by @sreimers in baresip/baresip#1528 * event: increase module_event buffer size by @sreimers in baresip/baresip#1532 * mqtt/subscribe: use safe odict_string to prevent crashes by @sreimers in baresip/baresip#1534 * stream: add stream_set_label by @alfredh in baresip/baresip#1537 * Makefile dependency check improvements by @sreimers in baresip/baresip#1531 * account: add enable/disable flag for video by @cspiel1 in baresip/baresip#1536 * audio: use account specific audio telev pt correctly by @cspiel1 in baresip/baresip#1542 * net: add missing HAVE_INET6 by @cspiel1 in baresip/baresip#1543 * account: remove unused API function for video enable by @cspiel1 in baresip/baresip#1544 * gst: changed log level for end of file message by @RobertMi21 in baresip/baresip#1548 * multicast: add new configurable multicast TTL config parameter by @cHuberCoffee in baresip/baresip#1545 * call: fix early video capability check (wrong SDP direction checked) by @cHuberCoffee in baresip/baresip#1549 * audio: catch end of file message in ausrc error handler (#1539) by @RobertMi21 in baresip/baresip#1550 * menu: added stopringing command by @RobertMi21 in baresip/baresip#1551 * stream: remove obsolete rx.jbuf_started by @cspiel1 in baresip/baresip#1552 * ua: downgrade level of message "ua: using best effort AF" by @viordash in baresip/baresip#1553 * outgoing calls early callid by @cspiel1 in baresip/baresip#1547 * audio: changed log level for ausrc error handler messages by @RobertMi21 in baresip/baresip#1554 * SIP default protocol by @cspiel1 in baresip/baresip#1538 * serreg: fix server selection in case all server were unavailable by @cHuberCoffee in baresip/baresip#1557 * multicast: fix missing unlock by @alfredh in baresip/baresip#1559 * config: replace strcpy by saver re_snprintf (#1558) by @cspiel1 in baresip/baresip#1560 * multicast: fix coverity scan by @alfredh in baresip/baresip#1561 * odict: hide struct odict_entry by @sreimers in baresip/baresip#1562 * ctrl_dbus: use mqueue to trigger processing of command in remain thread by @cspiel1 in baresip/baresip#1565 * multicast,config: add separate jitter buffer configuration by @cspiel1 in baresip/baresip#1566 * ua: emit CALL_CLOSED event when user agent is deleted by @cspiel1 in baresip/baresip#1564 * core: move stream_enable_rtp_timeout to api by @sreimers in baresip/baresip#1569 * stream: add mid sdp attribute by @alfredh in baresip/baresip#1570 * rtpext: change length type to size_t by @alfredh in baresip/baresip#1573 * avcodec: remove old backwards compat wrapper by @alfredh in baresip/baresip#1575 * main: Added option (-a) to set the ua agent string. by @RobertMi21 in baresip/baresip#1576 * menu fix tones for parallel outgoing calls by @cspiel1 in baresip/baresip#1577 * Fix win32 by @viordash in baresip/baresip#1579 * Fix static analyzer warnings by @viordash in baresip/baresip#1580 * call: added auto dtmf mode by @RobertMi21 in baresip/baresip#1583 * RTP inbound telephone events should not lead to packet loss by @cspiel1 in baresip/baresip#1581 * Running tests in a win32 project by @viordash in baresip/baresip#1585 * stream: wrong media direction after setting stream to hold by @RobertMi21 in baresip/baresip#1587 * move network check to module by @cspiel1 in baresip/baresip#1584 * serreg: do not ignore returned errors of ua_register() by @cspiel1 in baresip/baresip#1589 * Bundle media mux by @alfredh in baresip/baresip#1588 * mixausrc: no warnings flood when sampc changes by @cspiel1 in baresip/baresip#1595 * ua: select laddr with route to SDP offer address by @cspiel1 in baresip/baresip#1590 * net,uag: allow incoming peer-to-peer calls with user@domain by @cspiel1 in baresip/baresip#1591 * uag: in uag_reset_transp() select laddr with route to SDP raddr by @cspiel1 in baresip/baresip#1592 * uag: exit if transport could not be added by @cspiel1 in baresip/baresip#1593 * avcodec: use const AVCodec by @alfredh in baresip/baresip#1602 * module: deprecate module_tmp by @alfredh in baresip/baresip#1600 * test: use ausine as audio source by @alfredh in baresip/baresip#1601 * Selftest fakevideo by @alfredh in baresip/baresip#1604 * When adding local address, check that it has not been added already by @juha-h in baresip/baresip#1606 * start without network by @cspiel1 in baresip/baresip#1607 * config: add netroam module by @sreimers in baresip/baresip#1608 * multicast: allow any port number for sender and receiver by @cHuberCoffee in baresip/baresip#1609 * netroam: add netlink immediate network change detection by @cspiel1 in baresip/baresip#1612 * remove uag transp rm (#1611) by @cspiel1 in baresip/baresip#1616 * net dns srv get by @cspiel1 in baresip/baresip#1615 * move calls to stream_start_rtcp to call.c by @alfredh in baresip/baresip#1617 * video: null pointer check for the display handler by @cspiel1 in baresip/baresip#1621 * audio: add lock by @alfredh in baresip/baresip#1619 * ua: select proper af and laddr for outgoing IP calls by @cspiel1 in baresip/baresip#1618 * audio: lock stream by @alfredh in baresip/baresip#1622 * test: replace mock ausrc with ausine by @alfredh in baresip/baresip#1623 * menu ringback session progress by @cspiel1 in baresip/baresip#1625 * New module providing webrtc aec mobile mode filter by @juha-h in baresip/baresip#1626 * uag: respect setting sip_listen (#1627) by @cspiel1 in baresip/baresip#1628 * select laddr for SDP with respect to net_interface by @cspiel1 in baresip/baresip#1630 * stream: do not start audio during early-video by @cspiel1 in baresip/baresip#1629 * remove struct media_ctx by @alfredh in baresip/baresip#1632 * ci: add libwebrtc-audio-processing-dev (module webrtc_aec) by @sreimers in baresip/baresip#1635 * auconv: new module for audio format conversion by @alfredh in baresip/baresip#1634 * Support for IPv6 link local address for streams by @cspiel1 in baresip/baresip#1624 * call: check if address family is valid also for video stream by @cspiel1 in baresip/baresip#1636 * audio: pass pointer to tx->ausrc_prm instead of local variable by @cspiel1 in baresip/baresip#1637 * menu: add an event for call transfer by @cspiel1 in baresip/baresip#1641 * netroam: error handling for reset transport by @cspiel1 in baresip/baresip#1642 * mk: use CC_TEST for auto detect modules by @sreimers in baresip/baresip#1647 * test: use dtls_srtp.so module instead of mock by @alfredh in baresip/baresip#1646 * stream: create jbuf only if use_rtp is set by @cspiel1 in baresip/baresip#1648 * multicast: fix memleak in player destructor by @cspiel1 in baresip/baresip#1653 * stream: split up sender/receiver by @alfredh in baresip/baresip#1654 * set sdp laddr to SIP src address by @cspiel1 in baresip/baresip#1645 * serreg fix fallback accounts by @cspiel1 in baresip/baresip#1660 * ctrl_dbus: print command with the warning by @cspiel1 in baresip/baresip#1662 * call: new transfer call state to handle transfered calls correctly by @cHuberCoffee in baresip/baresip#1658 * serreg: prevent fast register retries if offline by @cspiel1 in baresip/baresip#1663 * av1: update packetization code by @alfredh in baresip/baresip#1657 * call: magic check in sipsess_desc_handler() by @cspiel1 in baresip/baresip#1664 * alsa: use snd_pcm_drop instead of snd_pcm_drain by @sreimers in baresip/baresip#1669 * Increased debian compat level to 10 by @juha-h in baresip/baresip#1667 * conf: fix conf_configure_buf() config parse by @sreimers in baresip/baresip#1666 * stream flush rtp socket by @cspiel1 in baresip/baresip#1671 * Transfer like rfc5589 by @cHuberCoffee in baresip/baresip#1678 * GTK: mem_derefer call earlier by @mbattista in baresip/baresip#1682 * netroam: add fail counter and event by @cspiel1 in baresip/baresip#1685 * Added API functions stream_metric_get_(tx|rx)_bitrate by @juha-h in baresip/baresip#1686 * Multicast new functions by @cHuberCoffee in baresip/baresip#1687 * avcodec: Enable pass-through for more codecs by @abrodkin in baresip/baresip#1692 * menu: filter for the correct call state in menu_selcall by @cHuberCoffee in baresip/baresip#1693 * test: fix warning on mingw32 by @alfredh in baresip/baresip#1696 * menu: Play ringback in play device by @myrkr in baresip/baresip#1698 * sip: add optional TCP source port by @cspiel1 in baresip/baresip#1695 * rtpext: change id unsigned -> uint8_t by @alfredh in baresip/baresip#1701 * ci: add mingw build test by @sreimers in baresip/baresip#1700 * test: use mediaenc srtp instead of mock by @alfredh in baresip/baresip#1702 * test: remove mock mediaenc by @alfredh in baresip/baresip#1704 * descr: add session_description by @alfredh in baresip/baresip#1706 * use fs_isfile() by @alfredh in baresip/baresip#1709 * stream: only call rtp_clear for audio by @alfredh in baresip/baresip#1710 * checks if call is available before calling call, closes #1708 by @mbattista in baresip/baresip#1712 * conf: add conf_loadfile by @alfredh in baresip/baresip#1713 * ice: remove ice_mode by @sreimers in baresip/baresip#1714 * audio: use auframe in encode_rtp_send, ref #1699 by @alfredh in baresip/baresip#1715 * Increased account's max video codec count from four to eight by @juha-h in baresip/baresip#1717 * gtk: Avoid duplicate call_timer registration by @myrkr in baresip/baresip#1719 * Attended call transfer by @cHuberCoffee in baresip/baresip#1718 * menu: exclude given call when searching for active call by @cspiel1 in baresip/baresip#1721 * menu: play call waiting tone on audio_player device by @cspiel1 in baresip/baresip#1722 * ci/build/macos: link ffmpeg@4 by @sreimers in baresip/baresip#1725 * module auresamp by @cspiel1 in baresip/baresip#1705 * test: remove h264 testcode, already in retest by @alfredh in baresip/baresip#1726 * h265: move from avcodec to rem by @alfredh in baresip/baresip#1728 * mc: send more details at receiver - timeout event by @cHuberCoffee in baresip/baresip#1731 * h265: move packetizer from avcodec to rem by @alfredh in baresip/baresip#1732 * FFmpeg 5 by @sreimers in baresip/baresip#1734 * Fixing clang ThreadSanitizer warnings by @sreimers in baresip/baresip#1730 * auresamp: replace anonymous union for pre C11 compilers by @cspiel1 in baresip/baresip#1738 * aufile: align naming of alloc handlers by @sreimers in baresip/baresip#1739 * auresamp fixes by @cspiel1 in baresip/baresip#1741 * mc: new priority handling with multicast state by @cHuberCoffee in baresip/baresip#1740 * remove support for Solaris platform by @alfredh in baresip/baresip#1745 * Allow hanging up call that has not been ACKed yet by @juha-h in baresip/baresip#1747 * Multicast identical condition and fmt string fix by @cHuberCoffee in baresip/baresip#1751 * audio: allocate aubuf before ausrc_alloc (fixes data race) by @sreimers in baresip/baresip#1748 * call: send supported header for 200 answering/ok by @cHuberCoffee in baresip/baresip#1752 * event: check if media line is present for encoding audio/video dir by @cspiel1 in baresip/baresip#1754 * Removed unused variable in modules/webrtc_aec/aec.cpp by @juha-h in baresip/baresip#1756 * audio use module auconv by @cspiel1 in baresip/baresip#1742 * test: use aufile module by @alfredh in baresip/baresip#1757 * x11grab: remove module, use avformat.so instead by @alfredh in baresip/baresip#1758 * audio: declare iterator inside for-loop (C99) by @alfredh in baresip/baresip#1759 * aufile: set run=true before write thread starts (#1727) by @cspiel1 in baresip/baresip#1762 * Added new API function call_supported() and used it in menu module by @juha-h in baresip/baresip#1761 * aufile: separate aufile_src.c from aufile.c by @cspiel1 in baresip/baresip#1765 * ctrl_dbus: fix possible data race (#1727) by @cspiel1 in baresip/baresip#1764 * menu select other call on hangup by @cspiel1 in baresip/baresip#1763 * event: encode also combined media direction by @cspiel1 in baresip/baresip#1766 == New Contributors * @srperens made their first contribution in baresip/baresip#1399 * @negbie made their first contribution in baresip/baresip#1451 * @andreaswatch made their first contribution in baresip/baresip#1512 * @viordash made their first contribution in baresip/baresip#1553 * @abrodkin made their first contribution in baresip/baresip#1692 * @myrkr made their first contribution in baresip/baresip#1698 ---
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