GNOME Bugzilla – Bug 760556
rtspsrc: interleaved data and srtp don't play well together
Last modified: 2016-01-18 14:12:19 UTC
When using RTSP with interleaved data (protocols=tcp) in conjunction with SRTP, rtspsrc doesn't work (see at the end). This could be solved by just disabling SRTP. If you have SRTP it probably means you also encrypt the RTSP channel, so there's no point on having additional encryption. But, in any case, this should still work. ---- $ gst-launch-1.0 rtspsrc location=rtsps://localhost:8554/webcam latency=300 protocols=tcp tls-validation-flags=0 ! decodebin ! fakesink silent=false -v Gives these error message: ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Internal data flow error. Additional debug info: gstrtspsrc.c(5483): gst_rtspsrc_loop (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: streaming task paused, reason not-negotiated (-4)
Created attachment 318925 [details] [review] follow stream profile when setting rtcp caps
commit 665d14a2a0e741c5b6153fce0b0a03ad93ea7968 Author: Aleix Conchillo Flaqué <aconchillo@gmail.com> Date: Tue Jan 12 14:01:21 2016 -0800 rtspsrc: handle rtcp/srtcp caps properly when using interleaved data We check the stream profile and use the proper RTCP caps: application/x-srtcp if we are using a secure profile and application/x-rtcp otherwise. https://bugzilla.gnome.org/show_bug.cgi?id=760556