Unreal Engine -> Pion WebRTC -> RTP Receiver

This is a proof of concept demonstrating pixel streaming from Unreal Engine to Pion's WebRTC, with Pion then forwarding RTP video/audio to FFPlay.

Running the proof of concept

There are number of moving pieces to run this demo. We tested this proof of concept using FFPlay 4.3.1, NodeJS 15.5.1, GoLang 1.15.6, and Unreal Engine 4.25 - we recommend using similar version numbers to reproduce our results.

1. Get the "Pixel Streaming Demo" and run that in Unreal Engine.
2. Run the Cirrus signalling server bundled with Unreal Engine by calling run.bat or sudo node cirrus.js in Samples\PixelStreaming\WebServers\SignallingWebServer.
3. Run this RTP forwarder, go run main.go in this repository.
4. Play the forwarded video/audio streams in FFPlay, run play-stream.bat or play-stream.sh

Configuring the forwarder

There are a number of flags that can be passed to main.go that may be of interest to those using this proof of concept:

// CirrusPort - The port of the Cirrus signalling server that the Pixel Streaming instance is connected to.
var CirrusPort = flag.Int("CirrusPort", 80, "The port of the Cirrus signalling server that the Pixel Streaming instance is connected to.")

// CirrusAddress - The address of the Cirrus signalling server that the Pixel Streaming instance is connected to.
var CirrusAddress = flag.String("CirrusAddress", "localhost", "The address of the Cirrus signalling server that the Pixel Streaming instance is connected to.")

// ForwardingAddress - The address to send the RTP stream to.
var ForwardingAddress = flag.String("ForwardingAddress", "127.0.0.1", "The address to send the RTP stream to.")

// RTPVideoForwardingPort - The port to use for sending the RTP video stream.
var RTPVideoForwardingPort = flag.Int("RTPVideoForwardingPort", 4002, "The port to use for sending the RTP video stream.")

// RTPAudioForwardingPort - The port to use for sending the RTP audio stream.
var RTPAudioForwardingPort = flag.Int("RTPAudioForwardingPort", 4000, "The port to use for sending the RTP audio stream.")

// RTPAudioPayloadType - The payload type of the RTP packet, 111 is OPUS.
var RTPAudioPayloadType = flag.Uint("RTPAudioPayloadType", 111, "The payload type of the RTP packet, 111 is OPUS.")

// RTPVideoPayloadType - The payload type of the RTP packet, 125 is H264 constrained baseline 2.0 in Chrome, with packetization mode of 1.
var RTPVideoPayloadType = flag.Uint("RTPVideoPayloadType", 125, "The payload type of the RTP packet, 125 is H264 constrained baseline in Chrome.")

// RTCPIntervalMs - How often (ms) to send RTCP messages (such as REMB, PLI)
var RTCPIntervalMs = flag.Int("RTCPIntervalMs", 2000, "How often (ms) to send RTCP message such as REMB, PLI.")

//Whether or not to send PLI messages on an interval.
var RTCPSendPLI = flag.Bool("RTCPSendPLI", true, "Whether or not to send PLI messages on an interval.")

//Whether or not to send REMB messages on an interval.
var RTCPSendREMB = flag.Bool("RTCPSendREMB", true, "Whether or not to send REMB messages on an interval.")

// Receiver-side estimated maximum bitrate.
var REMB = flag.Uint64("REMB", 400000000, "Receiver-side estimated maximum bitrate.")

Configuring FFPlay

You may need to download FFPlay if it is not on your system already: https://ffmpeg.org/ffplay.html Currently FFPlay is passed details about the RTP streams using the rtp-forwarder.sdp file. Additionally, FFPlay is passed the -fflags nobuffer -flags low_delay flags to reduce latency; however, these may not be suitable in all cases.