How create remote audio stream with webrtc transport with js-libp2p?

I have libp2p node
This node connects to the relay and has the ability to connect to a remote user.

this.libp2p = await createLibp2p({
    peerId: peerObject.PeerId,
    peerStore: PersistentPeerStore,
    addresses: {
        listen: [
            '/webrtc-direct',
            '/webrtc'
        ]
    },
    transports: [
        webRTCDirect(),
        webSockets({
            filter: filters.all
        }),
        webRTC(),
        circuitRelayTransport({
            discoverRelays: 2
        })
    ],
    peerDiscovery: boot,
    connectionEncryption: [noise()],
    streamMuxers: [yamux()],
    services: {
        identify: identify(),
        identifyPush: identifyPush(),
        pubsub: gossipsub(),
        dcutr: dcutr(),
        ping: ping(),
        dht: isDht ? kadDHT({
            kBucketSize: 12,
            kBucketSplitThreshold: `kBucketSize`,
            prefixLength: 32,
            clientMode: false,
            querySelfInterval: 5000,
            initialQuerySelfInterval: 1000,
            allowQueryWithZeroPeers: false,
            protocol: DhtProtocol,
            logPrefix: "libp2p:kad-dht",
            pingTimeout: 10000,
            pingConcurrency: 10,
            maxInboundStreams: 32,
            maxOutboundStreams: 64,
            peerInfoMapper: publicAddressesMapper,
        }) : () => {
        }
    },
    connectionManager: {
        minConnections: 20
    },
    transportManager: {
        faultTolerance: FaultTolerance.NO_FATAL
    },
    connectionGater: {
        denyDialPeer: (currentPeerId) => {
            return false
        },
    }
})

For transet text data.
Add listener for streams

await this.libp2p.handle(protoText, self.handler);
await this.libp2p.handle(protoAudio, self.handler);

handler
There I accept incoming data.

 handler: {
        value: async function ({ connection, stream }) {
            const protocol = stream.protocol
            if(stream.protocol === protoText) {
                const lp = lpStream(stream)

                const res = await lp.read()

                const output = new TextDecoder().decode(res.subarray())

                this._message = output
                this.printSmbl()
            }

            if(stream.protocol === protoAudio) {
                console.log('----------- stream --------------', '???')
               
            }
        },
        writable: true
    },

sender
Here I send data text and audio.
And I can’t figure out what to put here correctly.
It’s clear with the text, but not how to make an audio stream.

if(type === 'text') {
    const signal = AbortSignal.timeout(5000)

    this.DOM.input.textContent = ''
    const stream = await self.libp2p.dialProtocol(ma, proto, {
        signal
    });

    const lp = lpStream(stream)

    await lp.write(new TextEncoder().encode(msg))
}

if(type === 'audio') {
    const signal = AbortSignal.timeout(5000)
    this.DOM.input.textContent = ''
    
    const stream = await self.libp2p.dialProtocol(ma, protoAudio, {
        signal
    });
 
    ????

    return true
}

I can’t stream audio remotely.
I can’t find examples of how to transfer a audio stream.
I found examples of remote connection for webrtc but not for the libp2p library.