--------------------------------------------------------------------------------
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(a)fedoraproject.org> 1.0.0-1
- Upgrade to 1.0.0 (#1887059)
* Mon Jul 27 2020 Fedora Release Engineering <releng(a)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\
/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
--------------------------------------------------------------------------------