Gstreamer: udpsink/udpsrc versus tcpserversink/tcpclientsrc - gstreamer

Question

I'm very new to gstreamer but after a lot of research I've now managed to create my own working pipeline streaming a webcam over a network from a Raspberry PI Zero to a PC via a UDP transport. I am pleased with my progress! :)
But I'm struggling to create a TCP transport...
This pipeline works perfectly over UDP:
(note: simplified using a test video source and JPEG encoding):
Server UDP (192.168.2.1):
gst-launch-1.0 videotestsrc is-live=true ! jpegenc ! rtpjpegpay ! udpsink host=192.168.2.13 port=7001
Client UDP (192.168.2.13):
gst-launch-1.0 udpsrc port=7001 ! application/x-rtp,encoding-name=JPEG ! rtpjpegdepay ! jpegdec ! autovideosink
...but when I use a TCP sink/source with exactly the same elements I receive nothing but errors.
The modified pipeline using tcpserversink and tcpclientsrc:
Server TCP (192.168.2.1):
gst-launch-1.0 videotestsrc is-live=true ! jpegenc ! rtpjpegpay ! tcpserversink port=7001
Client TCP (192.168.2.13):
gst-launch-1.0 tcpclientsrc host=192.168.2.1 port=7001 ! application/x-rtp,encoding-name=JPEG ! rtpjpegdepay ! jpegdec ! autovideosink
Attempt 1: tcpserversink port=7001
ERROR: Failed to connect to host '192.168.2.1:7001': No connection could be made because the target machine actively refused it.
Attempt 2: tcpserversink host=localhost port=7001
ERROR: from element /GstPipeline:pipeline0/GstTCPClientSrc:tcpclientsrc0: Could not open resource for reading.
Attempt 3: tcpserversink host=127.0.0.1 port=7001
ERROR: from element /GstPipeline:pipeline0/GstTCPClientSrc:tcpclientsrc0: Could not open resource for reading.
Attempt 4: tcpserversink host=192.168.2.1 port=7001
ERROR: from element /GstPipeline:pipeline0/GstTCPClientSrc:tcpclientsrc0: Internal data stream error.
Attempt 5: tcpserversink host=0.0.0.0 port=7001
ERROR: from element /GstPipeline:pipeline0/GstTCPClientSrc:tcpclientsrc0: Internal data stream error.
I figured I should be able to replace src & sink elements without the pipeline breaking so I must just be missing something.
I would be grateful for any light you could shed on this.

Answer 1

You can solve it one of two ways (at least). The first is add the rtpstreampay element after the rtp payloader for your media type.
https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good/html/gst-plugins-good-plugins-rtpstreampay.html
server:
gst-launch-1.0 videotestsrc is-live=true \
! jpegenc \
! rtpjpegpay \
! rtpstreampay \
! tcpserversink port=7001
client:
gst-launch-1.0 tcpclientsrc port=7001 \
! application/x-rtp-stream,encoding-name=JPEG \
! rtpstreamdepay \
! rtpjpegdepay \
! jpegdec \
! autovideosink
The second way is to use a muxer rather than an rtp payloader, something like matroskamux which is pretty generic.
server:
gst-launch-1.0 videotestsrc is-live=true \
! jpegenc \
! matroskamux \
! tcpserversink port=7001
client:
gst-launch-1.0 tcpclientsrc port=7001 \
! matroskademux \
! jpegdec \
! autovideosink
You might also want to look into the GstRtspServer if you're wanting to do client/server rtp connections. A simple Python script like this will act as the server.
rtspserver.py
import gi
gi.require_version('Gst','1.0')
gi.require_version('GstRtspServer','1.0')
from gi.repository import Gst, GObject, GstRtspServer
Gst.init(None)
mainloop = GObject.MainLoop()
server = GstRtspServer.RTSPServer()
factory = GstRtspServer.RTSPMediaFactory()
factory.set_launch((
'videotestsrc is-live=true '
'! jpegenc '
'! rtpjpegpay name=pay0 pt=26'
))
# allow multiple connections
factory.set_shared(True)
mounts = server.get_mount_points()
mounts.add_factory('/live', factory)
server.attach(None)
mainloop.run()
And you can use a pipeline like this to view the output.
gst-launch-1.0 \
rtspsrc location=rtsp://localhost:8554/live latency=100 \
! rtpjpegdepay \
! jpegdec \
! autovideosink