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GStreamer multimedia framework
2894640cc5
When tunneling RTP over RTSP the stream transports are stored in a hash table in the GstRTSPClientPrivate struct. They are used for, among other things, mapping channel id to stream transports when receiving data from the client. The stream tranports are created and added to the hash table in handle_setup_request(), but unfortuately they are not removed in handle_teardown_request(). This means that if the client sends data on the RTSP connection after it has sent the TEARDOWN, which is often the case when audio backchannel is enabled, handle_data() will still be able to map the channel to a session transport and pass the data along to it. Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp() because the stream is no longer joined to a bin. We avoid this by removing the stream transports from the hash table when we handle the TEARDOWN request. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184> |
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| docs | ||
| examples | ||
| gst | ||
| hooks | ||
| scripts | ||
| tests | ||
| .gitignore | ||
| .gitlab-ci.yml | ||
| AUTHORS | ||
| ChangeLog | ||
| COPYING | ||
| COPYING.LIB | ||
| gst-rtsp-server.doap | ||
| meson.build | ||
| meson_options.txt | ||
| NEWS | ||
| README | ||
| RELEASE | ||
| REQUIREMENTS | ||
| TODO | ||
gst-rtsp-server is a library on top of GStreamer for building an RTSP server There are some examples in the examples/ directory and more comprehensive documentation in docs/README.