Gstreamer: udpsink/udpsrc versus tcpserversink/tcpclientsrc [closed]

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