Accepted baresip 0.5.4-1 (source) into unstable
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Hash: SHA256
Format: 1.8
Date: Mon, 03 Jul 2017 12:57:00 +0200
Source: baresip
Binary: baresip baresip-core baresip-gtk baresip-ffmpeg baresip-gstreamer baresip-x11
Architecture: source
Version: 0.5.4-1
Distribution: unstable
Urgency: medium
Maintainer: Debian VoIP Team <pkg-voip-maintainers@lists.alioth.debian.org>
Changed-By: Jonas Smedegaard <dr@jones.dk>
Description:
baresip - portable and modular SIP user-agent - metapackage
baresip-core - portable and modular SIP user-agent - core parts
baresip-ffmpeg - portable and modular SIP user-agent - FFmpeg codecs and formats
baresip-gstreamer - portable and modular SIP user-agent - GStreamer pipelines
baresip-gtk - portable and modular SIP user-agent - GTK+ front-end
baresip-x11 - portable and modular SIP user-agent - X11 features
Changes:
baresip (0.5.4-1) unstable; urgency=medium
.
[ upstream ]
* New release(s).
+ Config:
- Enable audio level RTP extension.
+ baresip-core:
- Support Client-to-Mixer Audio Level Indication (RFC 6464).
- Support RTP Header Extensions (RFC 5285).
- module: dont load same static module twice.
- ua: add ua_progress().
- ua: check for Accept header in incoming OPTIONS request.
- use a dummy RTP port for incoming OPTIONS.
- vidcodec: make the API re-entrant.
- vidfilt: make the API re-entrant.
- vidisp: make the API re-entrant.
- vidsrc: make the API re-entrant.
+ selftest:
- add test for audio level indication in call.
- add test for call progress.
+ Update all video modules with API-changes.
+ zrtp: check for RTP packet in send handler.
- registered zrtp_log function with zrtp engine.
- improved info message on how to verify remote peer.
- improved setting and printing of zrtp cache file.
.
[ Jonas Smedegaard ]
* Update package relations:
+ Tighten to build-depend on recent librem-dev.
+ Tighten build-dependency on libre-dev.
+ Drop obsolete temporary breaks/replaces for older baresip.
+ Fix limit build-dependency on libdirectfb-dev to supported archs.
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Files:
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=lGT8
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