2 Release notes for GStreamer RTSP Server Library 1.5.1
5 The GStreamer team is pleased to announce the first release of the unstable
6 1.5 release series. The 1.5 release series is adding new features on top of
7 the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
8 series of the GStreamer multimedia framework. The unstable 1.5 release series
9 will lead to the stable 1.6 release series in the next weeks, and newly added
10 API can still change until that point.
13 Binaries for Android, iOS, Mac OS X and Windows will be provided separately
14 during the unstable 1.5 release series.
19 Features of this release
22 Bugs fixed in this release
24 * 732238 : Listen on the multicast group for RTP/RTCP packets
25 * 734546 : tests: Unref element after usage
26 * 736041 : Protect rtsp transport data.
27 * 736647 : Tunneled RTSP sessions do not always timeout as expected
28 * 737110 : rtsp-client: race condition when closing client connection
29 * 737631 : gst-rtsp-server deadlock while sending response over TCP
30 * 737675 : media: media_unprepare() is kind of broken
31 * 737690 : rtsp-client: deadlock when setting session medias to NULL
32 * 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
33 * 737829 : rtsp-server: deactivate media when shutting down from paused
34 * 738905 : rtsp-client: add stream transport to the context
35 * 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
36 * 740752 : add retransmission support
37 * 740845 : crash when reciving a rtcp after teardown but before client finalize.
38 * 741678 : configure: add --disable-examples switch
39 * 742115 : Examples: Accept a 'port' argument for running multiple instances
40 * 742869 : Remove URI-escaping of RTSP session-id
41 * 742954 : Crash when two treads are in handle_new_sample at the same time.
42 * 743175 : Add support for RECORD
43 * 743346 : When system time is increased the ongoing RTSP sessions will time out.
44 * 743734 : RTCP packets not sent
45 * 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
46 * 745704 : Losing the first packet
47 * 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
48 * 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
49 * 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
50 * 749845 : Client have problem to find the teardown response.
54 You can find source releases of gst-rtsp-server in the download
55 directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
57 The git repository and details how to clone it can be found at
58 http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
62 The project's website is http://gstreamer.freedesktop.org/
64 ==== Support and Bugs ====
66 We use GNOME's bugzilla for bug reports and feature requests:
67 http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
69 Please submit patches via bugzilla as well.
71 For help and support, please subscribe to and send questions to the
72 gstreamer-devel mailing list (see below for details).
74 There is also a #gstreamer IRC channel on the Freenode IRC network.
78 GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
79 from there (see link above).
81 Interested developers of the core library, plugins, and applications should
82 subscribe to the gstreamer-devel list.
87 Contributors to this release
89 * Aleix Conchillo Flaqué
102 * Luis de Bethencourt
109 * Sebastian Rasmussen