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Release notes for GStreamer RTSP Server Library 1.5.1

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.

Binaries for Android, iOS, Mac OS X and Windows will be provided separately during the unstable 1.5 release series.

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 : fails due to autopoint erroring out due to missing gettext version in
    • 749845 : Client have problem to find the teardown response.


    You can find source releases of gst-rtsp-server in the gst-rtsp-server download directory.

    The git repository and details how to clone it can be found at .


    The project's website is

    Support and Bugs

    We use GNOME's bugzilla for bug reports and feature requests.

    Please submit patches via bugzilla as well.

    For help and support, please subscribe to and send questions to the gstreamer-devel mailing list (see below for details).

    Find us on IRC at #gstreamer.


    Git is hosted on You can browse the gst-rtsp-server repository.

    All code is in Git and can be checked out from there.

    Interested developers of the core library, plugins, and applications should subscribe to the gstreamer-devel list.


    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
    • Tim-Philipp Müller
    • Vincent Penquerc'h
    • Wim Taymans

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