GNOME Bugzilla – Bug 793264
RTSP server for 1.4 doesn't allow new appsrc pipeline connections
Last modified: 2018-11-03 15:41:38 UTC
Hello, After few play/stop connections from one unique client, rtsp server doesn't allow any new ones. Other behavior: multiple clients connect to server, new client connections will always work if at least one client is still connected but when all clients are disconnected, no new connection is possible. RTSP logs: https://gist.github.com/Mezzano/d18f7ab278e2174adc9c045d5bdea307 Code is adapted from test-appsrc.c: https://gist.github.com/Mezzano/43789624d4983d70bcd9610ecaddbb30 Thanks for helping
This is with gst-rtsp-server 1.4.x ?
It is indeed.
Please test with a newer version, there were lots of bugfixes and other improvements since then. 1.4 is already very old
Hello Sebastian and Time-Philipp, I've tested with Gstreamer 1.8.3 and still have problems. Here are some logs: https://gist.github.com/Mezzano/8a0843f909b0cccecbd0b61359efb6be and later on: https://gist.github.com/Mezzano/77f25685308fd87984321bfc0ace3058
Current stable version is. 1.12. Please test on that before reopening. Note that the erreur seams to be cause by some NV omx components.
Hello Nicolas, I've finally installed Gstreamer 1.13.90 and here are the logs I have when trying to reconnect after all clients are gone: Server side: https://gist.github.com/Mezzano/12d550685d27d3c65647c12fd1cae549 Client side: https://gist.github.com/Mezzano/433eb2b833cbe396329a69aedab0bdf9 Maybe you have a simple video stream example with appsrc appsink that works with rtsp server. The default appsrc example is not enough. Thanks for helping me with this.
Can someone look into this? With last version of gstreamer, I can't reconnect after all clients disconnected. When last client disconnects, I get the message "no sockets assigned for UDP" when trying to reconnect: 0:01:40.411599422 4683 0x69901860 INFO rtspsession rtsp-session.c:249:gst_rtsp_session_manage_media: manage new media 0x1156c70 in session 0x69904200 0:01:40.411722754 4683 0x69901860 DEBUG rtspstream rtsp-stream.c:1530:gst_rtsp_stream_allocate_udp_sockets:<GstRTSPStream@0x6a319168> Allocated already 0:01:40.421363998 4683 0x69901860 INFO rtspclient rtsp-client.c:3456:handle_request: client 0x1163a58: received a request PLAY rtsp://10.1.63.236:8554/colour/ 1.0 0:01:40.421567996 4683 0x69901860 DEBUG rtspmedia rtsp-media.c:4263:gst_rtsp_media_complete_pipeline:<GstRTSPMedia@0x1156c70> complete pipeline 0:01:40.421650996 4683 0x69901860 DEBUG rtspstream rtsp-stream.c:4539:gst_rtsp_stream_complete_stream:<GstRTSPStream@0x6a319168> complete stream 0:01:40.421720662 4683 0x69901860 DEBUG rtspstream rtsp-stream.c:2922:create_receiver_part:<GstRTSPStream@0x6a319168> create receiver part 0:01:40.422480321 4683 0x69901860 DEBUG rtspstream rtsp-stream.c:2778:create_sender_part:<GstRTSPStream@0x6a319168> create sender part 0:01:40.422563320 4683 0x69901860 DEBUG rtspstream rtsp-stream.c:2790:create_sender_part:<GstRTSPStream@0x6a319168> tcp: 0, udp: 1, mcast: 0 (ttl: 0) 0:01:40.422645653 4683 0x69901860 WARN rtspstream rtsp-stream.c:2793:create_sender_part:<GstRTSPStream@0x6a319168> no sockets assigned for UDP 0:01:40.422704652 4683 0x69901860 ERROR rtspclient rtsp-client.c:1735:handle_play_request: client 0x1163a58: failed to configure the pipeline 0:01:40.462821282 4683 0x69901860 INFO rtspclient rtsp-client.c:4309:closed: client 0x1163a58: connection closed 0:01:40.462930281 4683 0x69901860 INFO rtspclient rtsp-client.c:4541:client_watch_notify: client 0x1163a58: watch destroyed https://gist.github.com/Mezzano/43789624d4983d70bcd9610ecaddbb30 Thanks.
Hi, Having no answer, the best solution I've come to is to run locally a fake client with another gstreamer chain using rtspsrc. This makes that there is always one client connected and we never meet any latencies or reconnection problems. But it is really not the good way to go.
-- GitLab Migration Automatic Message -- This bug has been migrated to freedesktop.org's GitLab instance and has been closed from further activity. You can subscribe and participate further through the new bug through this link to our GitLab instance: https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/37.