Stream Microphone in Unity over RTP

After some investigations finally I managed to implement a working script that basically captures the RAW PCM audio from the Microphone and stream it over RTP/UDP. RTP headers are populated by some hard-coded values, so probably you should adapt the code to your needs. Remember to add an "AudioSource" component to your gameobject.

To test the reception you can just go and use ffmpeg player (ffplay), for example:

 ffplay rtp://0.0.0.0:your_port

Following is the C# script:

using System;
using System.Linq;
using System.Net;
using System.Net.Sockets;
using UnityEngine;

public static class RtpPacket
{
    public static void WriteHeader(byte[] rtpPacket
        , int rtpVersion
        , int rtpPadding
        , int rtpExtension
        , int rtpSrcCount
        , int rtpMarker
        , int rtpPayload)
    {
        rtpPacket[0] = (byte)((rtpVersion << 6) | (rtpPadding << 5) | (rtpExtension << 4) | rtpSrcCount);
        rtpPacket[1] = (byte)((rtpMarker << 7) | (rtpPayload & 0x7F));
    }

    public static void WriteSequenceNumber(byte[] rtpPacket, uint emptySeqId)
    {
        rtpPacket[2] = ((byte)((emptySeqId >> 8) & 0xFF));
        rtpPacket[3] = ((byte)((emptySeqId >> 0) & 0xFF));
    }

    public static void WriteTS(byte[] rtpPacket, uint ts)
    {
        rtpPacket[4] = ((byte)((ts >> 24) & 0xFF));
        rtpPacket[5] = ((byte)((ts >> 16) & 0xFF));
        rtpPacket[6] = ((byte)((ts >> 8) & 0xFF));
        rtpPacket[7] = ((byte)((ts >> 0) & 0xFF));
    }

    public static void WriteSSRC(byte[] rtpPacket, uint ssrc)
    {
        rtpPacket[8] = ((byte)((ssrc >> 24) & 0xFF));
        rtpPacket[9] = ((byte)((ssrc >> 16) & 0xFF));
        rtpPacket[10] = ((byte)((ssrc >> 8) & 0xFF));
        rtpPacket[11] = ((byte)((ssrc >> 0) & 0xFF));
    }
}

public class AudioStreamer : MonoBehaviour
{
    // Audio control variables
    AudioClip mic;
    int lastPos, pos;

    // UDP Socket variables
    private Socket socket;
    private IPEndPoint RemoteEndPoint;
    private UInt32 sequenecId = 0;

    void SetRtpHeader(byte[] rtpPacket)
    {
        // Populate RTP Packet Header
        // 0  - Version, P, X, CC, M, PT and Sequence Number
        // 32 - Timestamp. H264 uses a 90kHz clock
        // 64 - SSRC
        // 96 - CSRCs (optional)
        // nn - Extension ID and Length
        // nn - Extension header
        RtpPacket.WriteHeader(rtpPacket
            , 2    // version
            , 0    // padding
            , 0    // extension
            , 0    // csrc_count
            , 1    // marker, set to one for last packet
            , 11); // payload_type PCM 16bits BE signed
        RtpPacket.WriteSequenceNumber(rtpPacket, sequenecId);
        RtpPacket.WriteTS(rtpPacket, Convert.ToUInt32(DateTime.Now.Millisecond * 90));
        RtpPacket.WriteSSRC(rtpPacket, 0);
        sequenecId++;
    }

    void SendToServer(float[] samples)
    {
        const int RTP_HEADER_LEN = 12;
        if (socket == null) return;
        if (samples == null || samples.Length == 0) return;

        // Convert audio from float to signed 16 bit PCM BigEndian and copy it to the byte array
        var byteArray = new byte[samples.Length * sizeof(Int16)]; // to convert each sample float to Int16
        int i = 0;
        int j = 0;
        while (i < samples.Length)
        {
            Int16 sample = Convert.ToInt16((samples[i] * Int16.MaxValue) / 100);
            byteArray[j] = (byte)(sample & 0xFF);
            byteArray[j + 1] = (byte)((sample >> 8) & 0xFF);
            i = i + 1;
            j = j + 2;
        }

        var dataToSend = byteArray.Length;
        int maxEthMTU = 1400;
        int offset = 0;
        while (dataToSend > 0)
        {
            var bodyLen = Math.Min(dataToSend, maxEthMTU);
            var rtpAudioData = new byte[RTP_HEADER_LEN + bodyLen];
            SetRtpHeader(rtpAudioData);
            System.Array.Copy(byteArray, offset, rtpAudioData, RTP_HEADER_LEN, bodyLen);
            int dataSent = socket.SendTo(rtpAudioData, 0, rtpAudioData.Length, SocketFlags.None, RemoteEndPoint);
            dataToSend = dataToSend - dataSent;
            offset = offset + dataSent;
        }
    }

    void Start()
    {
            RemoteEndPoint = new IPEndPoint(IPAddress.Parse("your_server_ip"), your_server_port);
            socket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, ProtocolType.Udp);
            mic = Microphone.Start(null, true, 1, 44100); // Mono
    }

    private void Update()
    {
        if ((pos = Microphone.GetPosition(null)) > 0)
        {
            if (lastPos > pos) lastPos = 0;

            if (pos - lastPos > 0)
            {
                // Allocate the space for the new sample.
                int len = (pos - lastPos) * mic.channels;
                float[] samples = new float[len];
                mic.GetData(samples, lastPos);
                SendToServer(samples);
                lastPos = pos;
            }
        }
    }

    void OnDestroy()
    {
        Microphone.End(null);
    }
}