如何处理JsSIP中的音频流？

  • Z时代
  • 2024-01-10
  • 分类：问答

我正在创建使用JsSIP库来应答通过VoIP SIP提供程序进行的呼叫的React应用程序。

我已经创建了一个具有两个按钮（接受和拒绝）的页面。它在SIP服务器上成功注册了SIP客户端。它也成功接听电话，我可以接听。但是接听电话时我什么都没听到。

注册JsSIP客户端（willReceiveProps因为在更改道具后我有连接信息）：

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()

处理答案按钮单击的方法：

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？

回答：

我解决了问题。

问题在于this.audioElement.play()生产线的位置。

我将其移至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()
})

现在工作正常。希望您也觉得它有用。

以上是 如何处理JsSIP中的音频流？ 的全部内容， 来源链接： utcz.com/qa/432227.html

回到顶部