-------------------------------------------------------------------------------- Fedora EPEL Update Notification FEDORA-EPEL-2020-33382d341f 2020-10-25 01:31:23.076238 --------------------------------------------------------------------------------
Name : baresip Product : Fedora EPEL 8 Version : 1.0.0 Release : 1.el8 URL : http://www.creytiv.com/baresip.html 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 1.0.0 ============= Added ----- * aac: add `AAC_STREAMTYPE_AUDIO` enum value * aac: add `AAC_ prefix` * Video mode param to `call_answer()`, `ua_answer()` and `ua_hold_answer` * `video_stop_display()` API function * module: add path to `module_load()` function * conf: add `conf_configure_buf` * test: add usage of `g711.so` module * JSON initial codec state command and response * `account_set_video_codecs()` API function * net: add fallback dns nameserver * gtk: show `call_peername` in notify title * call: Added `call_state()` API function that returns enum state of the call * `account_set_stun_user()` and `account_set_stun_pass()` API functions * API functions `account_stun_uri` and `account_set_stun_uri` * ausine: Audio sine wave input module * gtk/menu: replace spaces from uri * jack: allowing jack client name to be specified in the config file * snapshot: Add snapshot_send and snapshot_recv commands #1029 * webrtc_aec: `extended_filter` config option * avfilter: FFmpeg filter graphs integration * reg: view proxy expiry value in `reg_status` * account: add parameter `rwait` for re-register interval * call, stream, menu: add cmd to set the direction of video stream * Added `AMRWBENC_PATH` env var to amr module `module.mk` Changed ------- * Using baresip/re fork now * audio: move calculation to `audio_jb_current_value` * avformat: clean up docs * gzrtp: update docs * account: increased size of audio codec list to 16 * video: make `video_sdp_attr_decode public` * config: Derive default audio driver from default audio device * jack: modifying info message on jack client creation * call: when video stream is disabled, stop also video display * dtls_srtp: use `tls_set_selfsigned_rsa` with keysize 2048 * rst: use a min ptime of 20ms * aac: change ptime to 4ms Fixed ----- * avcodec: fix H.264 interop with Firefox * winwave: `waveInGetPosition` is no longer supported for use as of Windows Vista * avcodec: call `av_hwdevice_ctx_create` before if-statement * account: use single quote instead of backtick * ice: fix segfault in `connh` * call: Update `call->got_offer` when re-INVITE or answer to re-INVITE is received * mk: Test also for `/usr/lib64/libspeexdsp.so` to cover Fedora/RHEL/CentOS * config: Allow distribution specific CA trust bundle locations * config: Allow distribution specific default audio device * mqtt: fix `err` is never read * avcodec: fix `err` is never read * gtk: notification buttons do not work on Systems * gtk: fix `dtmf_tone` and add tones as feedback * pulse: drain pulse buffers before freeing * jack: `jack_play` connect all physical ports * Makefile: do not try to install modules if build is static * gzrtp: `media_alloc` function is missing * call: when updating video, check if video stream has been disabled * amr: fix length check, fixes * modules: fix search path for `avdevice.h` * gtk: declare variables C89 style * config: init newly added member * menu: fix segfault in `ua_event_handler` * debug_cmd: fix OpenSSL no-deprecated #1065 * aac: handle missing bitrate parameter in SDP format * av1: properly configure encoder * call: When terminating outgoing call, terminate also possible refer subscription * menu: fix segfault in `/aubitrate` command * amr: should check if file (instead of directory) exists Removed ------- * ice: remove support for ICE-lite * ice: remove `ice_debug`, use log level `DEBUG` instead * ice: make stun server optional * config: remove `ice_debug option` (unused) * opengles: remove module (not working) -------------------------------------------------------------------------------- ChangeLog:
* Sat Oct 10 2020 Robert Scheck robert@fedoraproject.org 1.0.0-1 - Upgrade to 1.0.0 (#1887059) * Mon Jul 27 2020 Fedora Release Engineering releng@fedoraproject.org - 0.6.6-3 - Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild -------------------------------------------------------------------------------- References:
[ 1 ] Bug #1887059 - baresip-1.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=1887059 --------------------------------------------------------------------------------
This update can be installed with the "yum" update programs. Use su -c 'yum update baresip' at the command line. For more information, refer to "YUM", available at https://access.redhat.com/documentation/en-US/Red_Hat_Enterprise_Linux/7%5C /html/System_Administrators_Guide/ch-yum.html
All packages are signed with the Fedora EPEL GPG key. More details on the GPG keys used by the Fedora Project can be found at https://fedoraproject.org/keys --------------------------------------------------------------------------------