go-zero+WebRTC realizes real-time video communication

With the development of video communication technology, more and more application scenarios require the implementation of real-time video communication functions. WebRTC is an open source project that allows browsers and mobile applications to communicate in real time, and go-zero is a framework for quickly building high-performance Go language web services. This article will introduce how to use go-zero and WebRTC to implement real-time video communication.

1. Preliminary understanding of WebRTC

WebRTC is a Google open source project that allows real-time communication between browsers and mobile applications. It provides real-time audio and video communication and data transmission functions. WebRTC uses a series of technologies to achieve real-time communication functions, including:

1. TCP/UDP Transport Protocol
2. ICE (Interactive Connectivity Establishment) technology is used to determine the optimal path and correct The transmission protocol
3. SDP (Session Description Protocol) protocol, used to describe the session process
4. STUN (Session Traversal Utilities for NAT) protocol, used to detect and bypass NAT
5. TURN (Traversal Using Relays around NAT) protocol is used to use the relay server for transmission when both ends are unable to connect using STUN

2. Preliminary understanding of go-zero

go-zero is a framework for quickly building high-performance Go language web services. It has the following characteristics:

1. Based on RPC framework, supports multiple protocols
2. High performance, uses Sync.Pool and memory pool technology
3. Plug-in, flexible Extension
4. Support middleware
5. Support API gateway

3. Use go-zero and WebRTC to implement real-time video communication

In order to use go- To achieve real-time video communication between zero and WebRTC, we need to complete the following steps:

1. Build go-zero's Web service
2. Implement the signaling server of WebRTC
3. Implementation Video streaming transmission of WebRTC
4. Implementing the front-end page

The signaling server is a key part of WebRTC and is used to establish and maintain the video communication channel. We can use go-zero to implement the signaling server. First, we need to import the relevant go-zero dependency package, as shown below:

import (
    "bytes"
    "encoding/json"
    "github.com/creasty/defaults"
    "github.com/go-chi/chi/v5"
    "github.com/gorilla/websocket"
    "github.com/rs/zerolog"
    "github.com/rs/zerolog/log"
    "github.com/segmentio/ksuid"
    "math/rand"
    "net/http"
    "sync"
    "time"
)

Next, we can implement the WebSocket protocol server and the corresponding routing handler. The main function of the routing handler is to handle websocket connections and data transmission, and implement the basic functions of the signaling server. The code is as follows:

type SignalServer struct {
    hub *Hub
    mu  sync.Mutex
}

func NewSignalServer() *SignalServer {
    return &SignalServer{
        hub: newHub(),
    }
}

func (s *SignalServer) routes() *chi.Mux {
    r := chi.NewRouter()
    r.Handle("/ws", websocket.Handler(s.handleWebSocket))
    return r
}

func (s *SignalServer) handleWebSocket(conn *websocket.Conn) {
    sessionId := ksuid.New().String()
    client := &Client{
        id:   sessionId,
        conn: conn,
        send: make(chan []byte, 256),
        hub:  s.hub,
    }
    s.hub.register <- client
    go client.writePump()

    for {
        _, message, err := conn.ReadMessage()
        if err != nil {
            break
        }
        s.handleMessage(client, message)
    }

    s.hub.unregister <- client
    conn.Close()
}

func (s *SignalServer) handleMessage(client *Client, data []byte) {
    log.Debug().Msgf("received message: %s", data)
    message := &SignalingMessage{}
    if err := json.Unmarshal(data, message); err != nil {
        log.Error().Msgf("failed to unmarshal data: %s", err.Error())
        return
    }

    switch message.Type {
    case "register":
        s.handleRegister(client, message)
    case "offer":
        s.hub.broadcast <- &MessageData{
            SenderId: client.id,
            Type:     "offer",
            Payload:  message.Payload,
        }
    case "answer":
        s.hub.broadcast <- &MessageData{
            SenderId: client.id,
            Type:     "answer",
            Payload:  message.Payload,
        }
    case "candidate":
        s.hub.broadcast <- &MessageData{
            SenderId: client.id,
            Type:     "candidate",
            Payload:  message.Payload,
        }
    default:
        log.Error().Msgf("unknown message type: %s", message.Type)
    }
}

func (s *SignalServer) handleRegister(client *Client, message *SignalingMessage) {
    room := message.Payload
    s.mu.Lock()
    defer s.mu.Unlock()
    if _, ok := s.hub.rooms[room]; !ok {
        s.hub.rooms[room] = make(map[string]*Client)
    }
    s.hub.rooms[room][client.id] = client
    log.Debug().Msgf("client %s registered in room %s", client.id, room)
}

In the above code, we use the websocket.Handler function to handle WebSocket connection requests, where the createHub function is used to create a hub structure representing the signal server. The handleWebSocket function is used to handle read and write operations of websocket connections. The handleMessage function is used to handle different types of signaling messages. At the same time, in order to maintain connections between different clients, we created a hub structure and used pipes such as register, unregister, and broadcast to maintain the client list and respond when the client connects, disconnects, and sends messages. processing.

Next, we need to implement the video streaming function of WebRTC. In WebRTC, video streams are transmitted through PeerConnection objects. We can generate an offer signaling message and send it to another Peer by calling the CreateOffer method of RTCPeerConnection. After processing the offer message, another Peer can call the CreateAnswer method of RTCPeerConnection, generate an answer message, and send it back. Eventually, both parties will send their SDP description information to the other party through the SetLocalDescription and SetRemoteDescription methods of RTCPeerConnection to establish a video communication channel. The code is as follows:

func (p *Peer) generateOffer() (*sdp.SessionDescription, error) {
    offer, err := p.pconn.CreateOffer(nil)
    if err != nil {
        return nil, err
    }

    if err := p.pconn.SetLocalDescription(offer); err != nil {
        return nil, err
    }

    return offer, nil
}

func (p *Peer) handleOffer(payload string) (*sdp.SessionDescription, error) {
    offer, err := webrtc.NewSessionDescription(sdp.SessionDescriptionProtocolType, payload)
    if err != nil {
        return nil, err
    }
    if err := p.pconn.SetRemoteDescription(offer); err != nil {
        return nil, err
    }

    answer, err := p.pconn.CreateAnswer(nil)
    if err != nil {
        return nil, err
    }

    if err := p.pconn.SetLocalDescription(answer); err != nil {
        return nil, err
    }

    return answer, nil
}

func (p *Peer) handleAnswer(payload string) error {
    answer, err := webrtc.NewSessionDescription(sdp.SessionDescriptionProtocolType, payload)
    if err != nil {
        return err
    }
    if err := p.pconn.SetRemoteDescription(answer); err != nil {
        return err
    }
    return nil
}

In the above code, we define three processing methods: generateOffer, handleOffer and handleAnswer. generateOffer is used to generate an offer signaling message and return an object of type sdp.SessionDescription. handleOffer and handleAnswer are used to process offer and answer signaling messages respectively, and set their respective SDP description information through the SetRemoteDescription and SetLocalDescription methods.

Finally, we need to implement the front-end page and implement the video streaming function through WebRTC. I won’t go into details here.

Summary

This article introduces how to use go-zero and WebRTC to implement real-time video communication. We first introduced the basic knowledge of WebRTC, then used go-zero to implement the signaling server and video streaming functions of WebRTC, and finally implemented the front-end page. Through the above practices, we not only gained an in-depth understanding of the core technology of WebRTC, but also learned how to use go-zero for rapid Web service development.

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