peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{
    ICEServers: []webrtc.ICEServer{
        {
            URLs: []string{"stun:stun.l.google.com:19302"},
        },
    },
})
if err != nil {
    panic(err)
}

webrtc.NewPeerConnection创建默认配置的WebRTC标准PeerConnection。

// Open a UDP Listener for RTP Packets on port 5004
listener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
if err != nil {
    panic(err)
}
defer func() {
    if err = listener.Close(); err != nil {
        panic(err)
    }
}()

开一个UDP端口等待RTP数据传入。

// Create a video track
videoTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video", "pion")
if err != nil {
    panic(err)
}
rtpSender, err := peerConnection.AddTrack(videoTrack)
if err != nil {
    panic(err)
}

创建并添加本地的传给请求方的Track，AddTrack会根据传入的track构建。

// Read incoming RTCP packets
// Before these packets are returned they are processed by interceptors. For things
// like NACK this needs to be called.
go func() {
    rtcpBuf := make([]byte, 1500)
    for {
        if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
            return
        }
    }
}()

从注释上看，这里的读取只是为了rtpSender里面的那些操作能执行，那么也就是说如果没有rtpSender.Read发进来的包会阻塞？

这个rtpSender是一个RTPSender，Read里面是调用的interceptor.RTCPReader.Read，所以这里是为了执行绑定的interceptor里面的操作，比如返回的SenderReport等，只是因为这个程序里不需要用到这些东西所以直接忽略读出来的东西。

// 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())

    if connectionState == webrtc.ICEConnectionStateFailed {
        if closeErr := peerConnection.Close(); closeErr != nil {
            panic(closeErr)
        }
    }
})

连接状态改变时输出提示信息，不必多讲。

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

// Set the remote SessionDescription
if err = peerConnection.SetRemoteDescription(offer); 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
if err = peerConnection.SetLocalDescription(answer); 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()))

和pion-rtp-forwarder里一样的等待链接的方式，不必多讲。

// Read RTP packets forever and send them to the WebRTC Client
inboundRTPPacket := make([]byte, 1600) // UDP MTU
for {
    n, _, err := listener.ReadFrom(inboundRTPPacket)
    if err != nil {
        panic(fmt.Sprintf("error during read: %s", err))
    }

    if _, err = videoTrack.Write(inboundRTPPacket[:n]); err != nil {
        if errors.Is(err, io.ErrClosedPipe) {
            // The peerConnection has been closed.
            return
        }

        panic(err)
    }
}

和pion-rtp-forwarder里直接用select阻塞不一样，这里是一个死循环不停将UDP发来的RTP数据包写进前面创建的track里面。所以这地方应该就是主要的转发流程了。