如何在JsSIP中处理音频流？

Question

我正在创建使用JsSIP库来响应通过VoIP SIP发出的调用的React应用程序。

我已经创建了一个有两个按钮的页面(接受和拒绝)。成功地在SIP服务器上注册了SIP客户端.它也成功地接到了电话，我可以接听。但我接电话的时候什么都没听到。

注册JsSIP客户端(在willReceiveProps中，因为我有道具更改后连接的信息)：

const socketHost = 'wss://' + contactCenter.host + ':' + contactCenter.port
const socket = new JsSIP.WebSocketInterface(socketHost)
const configuration = {
    sockets: [socket],
    uri: 'sip:' + contactCenter.login + '@' + contactCenter.host,
    password: contactCenter.password,
    socketHost: socketHost,
}

const coolPhone = new JsSIP.UA(configuration)

coolPhone.on('connected', (e: any) => {
    const messages = ServiceContainer.get<MessageManagerInterface>(ServiceTypes.Messages)
    messages.addSuccess('SIP connected')
})

coolPhone.on('newRTCSession', (e: any) => {
    const messages = ServiceContainer.get<MessageManagerInterface>(ServiceTypes.Messages)
    messages.addAlert('New call')

    const session = e.session

    session.on('failed', this.resetLocalState)
    session.on('ended', this.resetLocalState)

    const numberRegexp = /\"(\d+)\"/
    const fromNumber = (numberRegexp.exec(e.request.headers.From[0].raw))[1]
    const toNumber = (numberRegexp.exec(e.request.headers.Contact[0].raw))[1].slice(1)

    this.setState({
        callReceived: true,
        callSession: session,
        fromNumber: fromNumber,
        toNumber: toNumber,
    })
})

coolPhone.start()

方法处理应答按钮，单击：

private answerCall = () => {
    const messages = ServiceContainer.get<MessageManagerInterface>(ServiceTypes.Messages)
    messages.addSuccess('Call answered')

    const callOptions = {
        mediaConstraints: {
            audio: true, // only audio calls
            video: false
        },
        pcConfig: {
            iceServers: [
                { urls: ["stun:stun.l.google.com:19302"] }
            ],
            iceTransportPolicy: "all",
            rtcpMuxPolicy: "negotiate"
        }
    }

    this.state.callSession.answer(callOptions)

    this.state.callSession.connection.addEventListener('addstream', (event: any) => {
        console.log(event)
        this.audioElement.srcObject = event.stream
    })

    this.audioElement.play()

    this.setState({
        callAnswered: true,
        callReceived: false,
    })
}

我做错什么了？

Answer 1

我解决了问题。

问题出在this.audioElement.play()线的位置。

我将其移到addstream事件的回调中：

this.state.callSession.connection.addEventListener('addstream', (event: any) => {
    console.log(event)
    this.audioElement.srcObject = event.stream
    this.audioElement.play()
})

现在它很好用。希望你也发现它有用。

Answer 2

您可以使用react-sip npm库，它简化了在React应用程序中的jssip的使用：https://www.npmjs.com/package/react-sip

您只需要将连接设置作为道具传递给<SipProvider/>，它将位于您的响应树顶部附近的某个位置。这将允许您执行基本的开始/停止/应答操作，并在上下文中查看调用的状态！