我的网络摄像机支持音频输出（音频反向通道）。我想通过 RTSP 实时传输 PC 麦克风音频，以便可以通过摄像头扬声器听到。 ONVIF 流媒体规范告诉我必须通过提供的 RTSP URL 发送音频数据。我的相机支持 ONVIF 配置文件 T。
我试过了：
public static RtspClient rtspClient;
public static IWaveIn sourceStream;
private static void CallAudio()
{
string CameraIp = "192.168.1.69";
string UserName = "admin";
string Password = "admin123";
var ClientMessageInspector = new ClientMessageInspector(UserName, Password);
//Call Device Url and get Services.
string DeviceServiceUrl = "http://" + CameraIp + "/onvif/device_service";
var deviceClient = new DeviceClient("DeviceBinding", new EndpointAddress(DeviceServiceUrl));
deviceClient.Endpoint.Behaviors.Add(ClientMessageInspector);
var getServices = deviceClient.GetServices(false);
//Call media2 getStreamingUri.
string url = "http://" + CameraIp + "/onvif/media2_service";
var Media2Client = new Media2Client("Media2Binding", new EndpointAddress(url));
Media2Client.Endpoint.Behaviors.Add(ClientMessageInspector);
var media2GetProfiles = Media2Client.GetProfiles(null, null);
var resp = Media2Client.GetAudioDecoderConfigurationOptions(null, null);
var responseGetAudioStreamUri = Media2Client.GetStreamUri("tcp", profiles[0].token); //This gets rtsp url of media from camera.
rtspClient = new RtspClient(responseGetAudioStreamUri, UserName, Password);
sourceStream = new WaveInEvent();
sourceStream.WaveFormat = new WaveFormat(64, 8, 1); //8000 16
sourceStream.DataAvailable += new EventHandler<WaveInEventArgs>(SourceStream_DataAvailable);
sourceStream.StartRecording();
Console.ReadKey();
}
//This method gets data from PC microphone and enocodes it into Mu-Law G711 and send to rtsp url.
private static void SourceStream_DataAvailable(object sender, WaveInEventArgs e)
{
byte[] encoded = TwoWayAudio_Encode_MuLaw(e.Buffer, 0, e.BytesRecorded);
rtspClient.SendData(encoded, encoded.Length, 3);
}
private static byte[] TwoWayAudio_Encode_MuLaw(byte[] data, int offset, int length)
{
byte[] encoded = new byte[length / 2];
int outIndex = 0;
for (int n = 0; n < length; n += 2)
{
encoded[outIndex++] = MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(data, offset + n));
}
return encoded;
}
我项目中的RTSP客户端来自这里。
Rtspclient.cs
using Rtsp;
using Rtsp.Messages;
using Rtsp.Sdp;
using System;
using System.Collections.Generic;
using System.Diagnostics;
using System.IO;
using System.Security.Cryptography;
using System.Text;
using System.Text.RegularExpressions;
namespace Rtsp
{
public class RtspClient
{
private RtspListener rtsp_client;
private RtspTcpTransport tcp_socket;
public string url;
public bool canPlay = false;
public string username;
public string password;
public ushort seqNo = 0;
public event EventHandler<string> RtspError;
public event EventHandler<byte[]> RtpDataReceived;
public Stopwatch stopwatch { get; private set; }
public RtspClient(string _url, string _username, string _password)
{
url = _url;
username = _username;
password = _password;
var uri = new Uri(_url);
tcp_socket = new RtspTcpTransport(uri.Host, 554); // 554);
if (tcp_socket.Connected == false)
{
Console.WriteLine("Error - did not connect");
return;
}
// Connect a RTSP Listener to the TCP Socket to send messages and listen for replies
rtsp_client = new RtspListener(tcp_socket);
rtsp_client.MessageReceived += Rtsp_client_MessageReceived;
rtsp_client.DataReceived += DataReceived;
rtsp_client.Start(); // start reading messages from the server
rtsp_client.AutoReconnect = true;
RtspRequest describe_message = new RtspRequestDescribe();
describe_message.RtspUri = uri;
describe_message.AddHeader("Accept: application/sdp");
describe_message.AddHeader("Require: www.onvif.org/ver20/backchannel");
rtsp_client.SendMessage(describe_message);
stopwatch = new Stopwatch();
stopwatch.Start();
}
private void DataReceived(object sender, RtspChunkEventArgs e)
{
int rtp_version = (e.Message.Data[0] >> 6);
int rtp_padding = (e.Message.Data[0] >> 5) & 0x01;
int rtp_extension = (e.Message.Data[0] >> 4) & 0x01;
int rtp_csrc_count = (e.Message.Data[0] >> 0) & 0x0F;
int rtp_marker = (e.Message.Data[1] >> 7) & 0x01;
int rtp_payload_type = (e.Message.Data[1] >> 0) & 0x7F;
uint rtp_sequence_number = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtp_timestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtp_ssrc = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtp_payload_start = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtp_csrc_count); // zero or more csrcs
uint rtp_extension_id = 0;
uint rtp_extension_size = 0;
if (rtp_extension == 1)
{
rtp_extension_id = ((uint)e.Message.Data[rtp_payload_start + 0] << 8) + (uint)(e.Message.Data[rtp_payload_start + 1] << 0);
rtp_extension_size = ((uint)e.Message.Data[rtp_payload_start + 2] << 8) + (uint)(e.Message.Data[rtp_payload_start + 3] << 0);
rtp_payload_start += 4 + (int)rtp_extension_size; // extension header and extension payload
}
Console.WriteLine("RTP Data"
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
+ " Seq=" + rtp_sequence_number
+ " Time=" + rtp_timestamp
+ " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length);
// If rtp_marker is '1' then this is the final transmission for this packet.
// If rtp_marker is '0' we need to accumulate data with the same timestamp
// ToDo - Check Timestamp matches
// Add to the tempoary_rtp List
if (rtp_payload_type == 98 || rtp_payload_type == 0)
{
byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
RtpDataReceived?.Invoke(null, rtp_payload);
}
}
public bool SendData(byte[] data, int count, int channel)
{
byte[] rtp_packet = new byte[12 + data.Length];
int rtp_version = 2;
int rtp_padding = 0;
int rtp_extension = 0;
int rtp_csrc_count = 0;
int rtp_marker = 1; // set to 1 if the last NAL in the array
//int rtp_payload_type = 98;
int rtp_payload_type = 0;
RTPPacketUtil.WriteHeader(rtp_packet, rtp_version, rtp_padding, rtp_extension, rtp_csrc_count, rtp_marker, rtp_payload_type);
RTPPacketUtil.WriteSequenceNumber(rtp_packet, seqNo);
seqNo++;
RTPPacketUtil.WriteTS(rtp_packet, (uint)stopwatch.ElapsedMilliseconds);
UInt32 empty_ssrc = 1293847657;
RTPPacketUtil.WriteSSRC(rtp_packet, empty_ssrc);
// Now append the raw NAL
System.Array.Copy(data, 0, rtp_packet, 12, data.Length);
if (canPlay)
{
rtsp_client.SendData(channel, rtp_packet);
return true;
}
else return false;
}
private void Rtsp_client_MessageReceived(object sender, RtspChunkEventArgs e)
{
RtspResponse message = e.Message as RtspResponse;
if (message.ReturnCode == 500)
{
RtspError?.Invoke(this, "Internal Server Error");
}
if (message.ReturnCode == 401)
{
Rtsp.Messages.RtspRequest msg = null;
switch (message.OriginalRequest.Method)
{
case "DESCRIBE":
msg = new RtspRequestDescribe();
break;
case "SETUP":
msg = new RtspRequestSetup();
break;
default:
break;
}
msg.RtspUri = new Uri(url);
var header = message.Headers["WWW-Authenticate"];
var _realm = GrabHeaderVar("realm", header);
var _nonce = GrabHeaderVar("nonce", header);
var ha1 = CalculateMd5Hash(string.Format("{0}:{1}:{2}", username, _realm, password));
var ha2 = CalculateMd5Hash(string.Format("{0}:{1}", message.OriginalRequest.Method, url));
var digestResponse = CalculateMd5Hash(string.Format("{0}:{1}:{2}", ha1, _nonce, ha2));
var digest = string.Format("Digest username=\"{0}\", realm=\"{1}\", nonce=\"{2}\", uri=\"{3}\", response=\"{4}\" ",
username, _realm, _nonce, url, digestResponse);
msg.AddHeader("Authorization: " + digest);
msg.AddHeader("Accept: application/sdp");
rtsp_client.SendMessage(msg);
return;
}
Console.WriteLine("Received " + message.OriginalRequest.ToString());
if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// Examine the SDP
Console.Write(Encoding.UTF8.GetString(message.Data));
SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP.
// If the attribute is for Video, then send a SETUP
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio || sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video)
{
// seach the atributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
String fmtp = ""; // holds SPS and PPS
String rtpmap = ""; // holds the Payload format, 96 is often used with H264
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control")) control = attrib.Value;
if (attrib.Key.Equals("fmtp")) fmtp = attrib.Value;
if (attrib.Key.Equals("rtpmap")) rtpmap = attrib.Value;
}
// Get the Payload format number for the Video Stream
String[] split_rtpmap = rtpmap.Split(' ');
var video_payload = 0;
bool result = Int32.TryParse(split_rtpmap[0], out video_payload);
// Send SETUP for the Video Stream
// using Interleaved mode (RTP frames over the RTSP socket)
Rtsp.Messages.RtspRequest setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(url + "/" + control);
//setup_message.AddHeader("Transport: RTP/AVP/TCP;interleaved=0");
setup_message.AddHeader("Require: www.onvif.org/ver20/backchannel");
rtsp_client.SendMessage(setup_message);
}
}
}
if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestSetup)
{
// Got Reply to SETUP
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
String session = message.Session; // Session value used with Play, Pause, Teardown
// Send PLAY
RtspRequest play_message = new RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
play_message.AddHeader("Require: www.onvif.org/ver20/backchannel");
rtsp_client.SendMessage(play_message);
}
if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestPlay)
{
// Got Reply to PLAY
Console.WriteLine("Got reply from Play " + message.Command);
canPlay = true;
}
}
private static string GrabHeaderVar(string varName, string header)
{
var regHeader = new Regex(string.Format(@"{0}=""([^""]*)""", varName));
var matchHeader = regHeader.Match(header);
if (matchHeader.Success)
return matchHeader.Groups[1].Value;
throw new ApplicationException(string.Format("Header {0} not found", varName));
}
private static string CalculateMd5Hash(string input)
{
var inputBytes = Encoding.ASCII.GetBytes(input);
var hash = MD5.Create().ComputeHash(inputBytes);
var sb = new StringBuilder();
foreach (var b in hash)
sb.Append(b.ToString("x2"));
return sb.ToString();
}
public void Dispose()
{
rtsp_client.Stop();
rtsp_client.Dispose();
}
}
在调用任何 RTSP 方法之前，我添加了 RTSP backchannel 用于检查 AudioBack 通道的相机支持。调用“描述”、“设置”和“播放”后得到的输出正常：
Received Rtsp.Messages.RtspRequestDescribe
v=0
o=- 0 0 IN IP4 192.168.1.69
s=LIVE VIEW
c=IN IP4 0.0.0.0
t=0 0
a=control:rtsp://192.168.1.69/rtsp_tunnel?p=0&h26x=4&aon=1&aud=0
m=video 0 RTP/AVP 35
a=rtpmap:35 H264/90000
a=control:rtsp://192.168.1.69/rtsp_tunnel?p=0&h26x=4&aon=1&aud=0&stream=video
a=recvonly
a=fmtp:35 packetization-mode=1;profile-level-id=4d0029;sprop-parameter-
sets=Z00AKZpkA8ARPy4C1BQEFAg=,aO48gA==
m=audio 0 RTP/AVP 96
a=rtpmap:96 mpeg4-generic/16000/1
a=fmtp:96 streamtype=5; profile-level-id=5; mode=AAC-hbr; config=1408; SizeLength=13; IndexLength=3;
IndexDeltaLength=3
a=control:rtsp://192.168.1.69/rtsp_tunnel?p=0&h26x=4&aon=1&aud=0&stream=audio
a=recvonly
m=audio 0 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=control:rtsp://192.168.1.69/rtsp_tunnel?p=0&h26x=4&aon=1&aud=0&stream=backchannel
a=sendonly
Received Rtsp.Messages.RtspRequestSetup
Got reply from Setup. Session is 12346e9856840dc
Received Rtsp.Messages.RtspRequestSetup
Got reply from Setup. Session is 12346e9856840dc
Received Rtsp.Messages.RtspRequestSetup
Got reply from Setup. Session is 12346e9856840dc
Received Rtsp.Messages.RtspRequestPlay
Got reply from Play RTSP/1.0 200 OK
Received Rtsp.Messages.RtspRequestPlay
Got reply from Play RTSP/1.0 200 OK
Received Rtsp.Messages.RtspRequestPlay
Got reply from Play RTSP/1.0 200 OK
Play 方法响应后，我开始使用 RTSP 客户端中的 send 方法发送编码数据。但相机端听不到音频。
我明白了我做错了什么。实际上上面提到的步骤很好并且相机返回了好的，我做错的地方在上面的代码中：
sourceStream.WaveFormat = new WaveFormat(64, 8, 1); //8000 16
而不是 64 和 8 参数，它应该是：
sourceStream.WaveFormat = new WaveFormat(8000, 16, 1); //8000 16
都是语音采样率导致语音发送听不到。谢谢你！
可以提供一下这个项目的源码吗？或者一些库/文档来实现这个？我正在尝试做同样的事情 谢谢你