Golang development: Implementing a video call application based on WebRTC-Golang-php.cn

Golang development: Implementing a video call application based on WebRTC

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Release： 2023-09-20 15:22:50

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Golang development: Implementing a video call application based on WebRTC

Abstract:
WebRTC (Web Real-Time Communication) is an open standard for real-time audio and video Communication technology can be used to build applications such as audio and video calls, conferences, and real-time live broadcasts. This article will introduce how to use Golang to develop a video call application based on WebRTC, and provide some specific code examples to help readers have a deeper understanding and mastery of related technologies.

1. Background
WebRTC is an open real-time audio and video communication standard that can be used on various web browsers and operating systems. Through WebRTC, we can achieve real-time audio and video calls between browsers. Due to its openness and compatibility, WebRTC has become the preferred technology for building audio and video communication applications.

2. Environment preparation
Before starting development, we need to install the Go language development environment and WebRTC-related libraries.

1. Install the Golang development environment:

Visit the official website https://golang.org/dl/, download and install the Golang version suitable for your operating system.

2. Install the WebRTC library:

The Golang implementation of WebRTC is developed and maintained by Pion. They provide a Golang library that can be used directly. Use the following command to install the Pion library:
```
go get github.com/pion/webrtc/v3
```
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3. Create a video call application
The following is a simple example that demonstrates how to create a video call application using Golang and the WebRTC library.

package main

import (
    "encoding/json"
    "fmt"
    "log"
    "net/http"

    "github.com/gorilla/websocket"
    "github.com/pion/webrtc/v3"
)

var (
    upgrader = websocket.Upgrader{}
    peers    = map[*websocket.Conn]*webrtc.PeerConnection{}
)

func main() {
    http.HandleFunc("/offer", handleOffer)
    http.HandleFunc("/answer", handleAnswer)
    http.HandleFunc("/ws", handleWebSocket)
    log.Fatal(http.ListenAndServe(":8080", nil))
}

func handleOffer(w http.ResponseWriter, r *http.Request) {
    conn, err := upgrader.Upgrade(w, r, nil)
    if err != nil {
        log.Println(err)
        return
    }
    defer conn.Close()

    pc, err := webrtc.NewPeerConnection(webrtc.Configuration{})
    if err != nil {
        log.Println(err)
        return
    }

    pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
        if candidate != nil {
            jsonCandidate, _ := json.Marshal(candidate.ToJSON())
            conn.WriteMessage(websocket.TextMessage, jsonCandidate)
        }
    })

    pc.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
        go func() {
            for {
                _, _, _ = track.ReadRTP()
            }
        }()
    })

    pc.AddTransceiver(webrtc.RTPCodecTypeVideo)

    peers[conn] = pc

    for {
        _, message, err := conn.ReadMessage()
        if err != nil {
            log.Println(err)
            delete(peers, conn)
            return
        }
        pc.AddICECandidate(webrtc.ICECandidateInit{
            Candidate: string(message),
        })
    }
}

// handleAnswer 处理answer请求
func handleAnswer(w http.ResponseWriter, r *http.Request) {
    var answer struct {
        Candidate string `json:"candidate"`
    }
    err := json.NewDecoder(r.Body).Decode(&answer)
    if err != nil {
        http.Error(w, err.Error(), http.StatusBadRequest)
        return
    }

    conn, err := upgrader.Upgrade(w, r, nil)
    if err != nil {
        log.Println(err)
        return
    }
    defer conn.Close()

    pc := peers[conn]
    pc.AddICECandidate(webrtc.ICECandidateInit{
        Candidate: answer.Candidate,
    })
}

// handleWebSocket 处理WebSocket连接请求
func handleWebSocket(w http.ResponseWriter, r *http.Request) {
    conn, err := upgrader.Upgrade(w, r, nil)
    if err != nil {
        log.Println(err)
        return
    }
    defer conn.Close()

    pc, err := webrtc.NewPeerConnection(webrtc.Configuration{})
    if err != nil {
        log.Println(err)
        return
    }

    pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
        if candidate != nil {
            jsonCandidate, _ := json.Marshal(candidate.ToJSON())
            conn.WriteMessage(websocket.TextMessage, jsonCandidate)
        }
    })

    pc.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
        go func() {
            for {
                _, _, _ = track.ReadRTP()
            }
        }()
    })

    offer, err := pc.CreateOffer(nil)
    if err != nil {
        log.Println(err)
        return
    }

    err = pc.SetLocalDescription(offer)
    if err != nil {
        log.Println(err)
        return
    }

    conn.WriteJSON(offer)

    peers[conn] = pc
}

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In the above code, we used Golang’s net/http package to create a simple HTTP server. In the server, we use websocket.Upgrader to handle WebSocket connection requests, and the github.com/pion/webrtc/v3 library to handle WebRTC related operations. peers is a map used to hold connections and peer connection instances.

4. Compile and run
1. Execute the go mod init command in the project root directory to initialize the go module.

2. Execute the following command to compile and run the code:

go build -o video-call-app
./video-call-app

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5. Summary
Through the above sample code, we demonstrate how to use Golang and WebRTC libraries to create a WebRTC-based Video calling app. Readers can carry out secondary development and functional expansion according to their own needs. I hope this article can provide some help to beginners and developers in implementing WebRTC-based audio and video call applications in Golang development.

Reference materials:

Pion official documentation: https://pkg.go.dev/github.com/pion/webrtc/v3
WebRTC official documentation ：https://webrtc.org/

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