rtp-to-webrtc

rtp-to-webrtc

rtp-to-webrtc demonstrates how to consume a RTP stream video UDP, and then send to a WebRTC client.

With this example we have pre-made GStreamer and ffmpeg pipelines, but you can use any tool you like!

Instructions

Download rtp-to-webrtc

export GO111MODULE=on
go get github.com/guoguov/webrtc/examples/rtp-to-webrtc

Open jsfiddle example page

jsfiddle.net you should see two text-areas and a 'Start Session' button

Run rtp-to-webrtc with your browsers SessionDescription as stdin

In the jsfiddle the top textarea is your browser's SessionDescription, copy that and:

Linux/macOS

Run echo $BROWSER_SDP | rtp-to-webrtc

Windows

1. Paste the SessionDescription into a file.
2. Run rtp-to-webrtc < my_file

Send RTP to listening socket

You can use any software to send VP8 packets to port 5004. We also have the pre made examples below

GStreamer

gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,format=I420 ! vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 ! rtpvp8pay ! udpsink host=127.0.0.1 port=5004

ffmpeg

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -vcodec libvpx -cpu-used 5 -deadline 1 -g 10 -error-resilient 1 -auto-alt-ref 1 -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'

If you wish to send audio replace all occurrences of vp8 with Opus in main.go then run

ffmpeg -f lavfi -i 'sine=frequency=1000' -c:a libopus -b:a 48000 -sample_fmt s16p -ssrc 1 -payload_type 111 -f rtp -max_delay 0 -application lowdelay 'rtp://127.0.0.1:5004?pkt_size=1200'

If you wish to send H264 instead of VP8 replace all occurrences of vp8 with H264 in main.go then run

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -pix_fmt yuv420p -c:v libx264 -g 10 -preset ultrafast -tune zerolatency -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'

Input rtp-to-webrtc's SessionDescription into your browser

Copy the text that rtp-to-webrtc just emitted and copy into second text area

Hit 'Start Session' in jsfiddle, enjoy your video!

A video should start playing in your browser above the input boxes.

Congrats, you have used Pion WebRTC! Now start building something cool

Dealing with broken/lossy inputs

Pion WebRTC also provides a SampleBuilder. This consumes RTP packets and returns samples. It can be used to re-order and delay for lossy streams. You can see its usage in this example in daf27b.

Currently it isn't working with H264, but is useful for VP8 and Opus. See #1652 for the status of fixing for H264.