webrtc/examples/stats/main.go

// SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
// SPDX-License-Identifier: MIT

//go:build !js
// +build !js

// stats demonstrates how to use the webrtc-stats implementation provided by Pion WebRTC.
package main

import (
	"fmt"
	"time"

	"github.com/pion/interceptor"
	"github.com/pion/interceptor/pkg/stats"
	"github.com/pion/webrtc/v3"
	"github.com/pion/webrtc/v3/examples/internal/signal"
)

// nolint:gocognit
func main() {
	// Everything below is the Pion WebRTC API! Thanks for using it ❤️.

	// Create a MediaEngine object to configure the supported codec
	m := &webrtc.MediaEngine{}

	if err := m.RegisterDefaultCodecs(); err != nil {
		panic(err)
	}

	// Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
	// This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
	// this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
	// for each PeerConnection.
	i := &interceptor.Registry{}

	statsInterceptorFactory, err := stats.NewInterceptor()
	if err != nil {
		panic(err)
	}

	var statsGetter stats.Getter
	statsInterceptorFactory.OnNewPeerConnection(func(_ string, g stats.Getter) {
		statsGetter = g
	})
	i.Add(statsInterceptorFactory)

	// Use the default set of Interceptors
	if err = webrtc.RegisterDefaultInterceptors(m, i); err != nil {
		panic(err)
	}

	// Create the API object with the MediaEngine
	api := webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i))

	// Prepare the configuration
	config := webrtc.Configuration{
		ICEServers: []webrtc.ICEServer{
			{
				URLs: []string{"stun:stun.l.google.com:19302"},
			},
		},
	}

	// Create a new RTCPeerConnection
	peerConnection, err := api.NewPeerConnection(config)
	if err != nil {
		panic(err)
	}

	// Allow us to receive 1 audio track, and 1 video track
	if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio); err != nil {
		panic(err)
	} else if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo); err != nil {
		panic(err)
	}

	// Set a handler for when a new remote track starts. We read the incoming packets, but then
	// immediately discard them
	peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
		fmt.Printf("New incoming track with codec: %s\n", track.Codec().MimeType)

		go func() {
			for {
				stats := statsGetter.Get(uint32(track.SSRC()))

				fmt.Printf("Stats for: %s\n", track.Codec().MimeType)
				fmt.Println(stats.InboundRTPStreamStats)

				time.Sleep(time.Second * 5)
			}
		}()

		rtpBuff := make([]byte, 1500)
		for {
			_, _, readErr := track.Read(rtpBuff)
			if readErr != nil {
				panic(readErr)
			}
		}
	})

	// Set the handler for ICE connection state
	// This will notify you when the peer has connected/disconnected
	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
		fmt.Printf("Connection State has changed %s \n", connectionState.String())
	})

	// Wait for the offer to be pasted
	offer := webrtc.SessionDescription{}
	signal.Decode(signal.MustReadStdin(), &offer)

	// Set the remote SessionDescription
	err = peerConnection.SetRemoteDescription(offer)
	if err != nil {
		panic(err)
	}

	// Create answer
	answer, err := peerConnection.CreateAnswer(nil)
	if err != nil {
		panic(err)
	}

	// Create channel that is blocked until ICE Gathering is complete
	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)

	// Sets the LocalDescription, and starts our UDP listeners
	err = peerConnection.SetLocalDescription(answer)
	if err != nil {
		panic(err)
	}

	// Block until ICE Gathering is complete, disabling trickle ICE
	// we do this because we only can exchange one signaling message
	// in a production application you should exchange ICE Candidates via OnICECandidate
	<-gatherComplete

	// Output the answer in base64 so we can paste it in browser
	fmt.Println(signal.Encode(*peerConnection.LocalDescription()))

	// Block forever
	select {}
}