-Release notes for GStreamer RTSP Server Library 1.5.1
+Release notes for GStreamer RTSP Server Library 1.12.0
+The GStreamer team is pleased to announce the first release in the stable 1.12
+release series. The 1.12 release series is adding new features on top of the
+1.0, 1.2, 1.4, 1.6, 1.8 and 1.10 series and is part of the API and ABI-stable
+1.x release series of the GStreamer multimedia framework.
-The GStreamer team is pleased to announce the first release of the unstable
-1.5 release series. The 1.5 release series is adding new features on top of
-the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
-series of the GStreamer multimedia framework. The unstable 1.5 release series
-will lead to the stable 1.6 release series in the next weeks, and newly added
-API can still change until that point.
+Full release notes can be found here
-Binaries for Android, iOS, Mac OS X and Windows will be provided separately
-during the unstable 1.5 release series.
+Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days.
-Features of this release
-
-
-Bugs fixed in this release
-
- * 732238 : Listen on the multicast group for RTP/RTCP packets
- * 734546 : tests: Unref element after usage
- * 736041 : Protect rtsp transport data.
- * 736647 : Tunneled RTSP sessions do not always timeout as expected
- * 737110 : rtsp-client: race condition when closing client connection
- * 737631 : gst-rtsp-server deadlock while sending response over TCP
- * 737675 : media: media_unprepare() is kind of broken
- * 737690 : rtsp-client: deadlock when setting session medias to NULL
- * 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
- * 737829 : rtsp-server: deactivate media when shutting down from paused
- * 738905 : rtsp-client: add stream transport to the context
- * 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
- * 740752 : add retransmission support
- * 740845 : crash when reciving a rtcp after teardown but before client finalize.
- * 741678 : configure: add --disable-examples switch
- * 742115 : Examples: Accept a 'port' argument for running multiple instances
- * 742869 : Remove URI-escaping of RTSP session-id
- * 742954 : Crash when two treads are in handle_new_sample at the same time.
- * 743175 : Add support for RECORD
- * 743346 : When system time is increased the ongoing RTSP sessions will time out.
- * 743734 : RTCP packets not sent
- * 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
- * 745704 : Losing the first packet
- * 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
- * 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
- * 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
- * 749845 : Client have problem to find the teardown response.
-
==== Download ====
You can find source releases of gst-rtsp-server in the download
-directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
+directory: https://gstreamer.freedesktop.org/src/gst-rtsp-server/
The git repository and details how to clone it can be found at
http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
==== Homepage ====
-The project's website is http://gstreamer.freedesktop.org/
+The project's website is https://gstreamer.freedesktop.org/
==== Support and Bugs ====
subscribe to the gstreamer-devel list.
-Applications
-
Contributors to this release
- * Aleix Conchillo Flaqué
- * Alistair Buxton
- * Andreas Frisch
- * Anila Balavan
- * Arun Raghavan
- * Branko Subasic
- * Edward Hervey
- * Gregor Boirie
- * Göran Jönsson
- * Hyunjun Ko
- * Jan Schmidt
- * Kent-Inge Ingesson
- * Linus Svensson
- * Luis de Bethencourt
- * Matthew Waters
- * Nicolas Dufresne
- * Nirbheek Chauhan
- * Ognyan Tonchev
- * Olivier Crête
* Sebastian Dröge
- * Sebastian Rasmussen
- * Srimanta Panda
- * Stefan Sauer
+ * Thibault Saunier
* Tim-Philipp Müller
- * Vincent Penquerc'h
- * Wim Taymans
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