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资源说明：A RTSP handling library
Sharp RTSP
==========
[](https://ci.appveyor.com/project/ngraziano/sharprtsp)
[](https://coveralls.io/github/ngraziano/SharpRTSP?branch=master)
A C# library to build RTSP Clients, RTSP Servers and handle RTP data streams. The library has several examples.
* RTSP Client Example - will connect to a RTSP server and receive Video and Audio in H264, H265/HEVC, G711, AAC and AMR formats. UDP, TCP and Multicast are supported. The data received is written to files.
* RTSP Camera Server Example - A YUV Image Generator and a very simple H264 Encoder generate H264 NALs which are then delivered via a RTSP Server to clients
* RTP Receiver - will recieve RTP and RTCP packets and pass them to a transport handler
* RTSP Server - will accept RTSP connections and talk to clients
* RTP Sender - will send RTP packets to clients
* Transport Handler - Transport hanlders for H264, H265/HEVC, G711 and AMR are provided.
**:warning: : This library does not handle the decoding of the video or audio (eg converting H264 into a bitmap). SharpRTSP is limited to the transport layer and generates the raw data that you need to feed into a video decoder or audio decoder. Many people use FFMPEG or use Hardware Accelerated Operating System APIs to do the decoding.**
Walkthrough of the RTSP Client Example
======================================
This is a walkthrough of an **old version** of the RTSP Client Example which highlights the main way to use the library.
* STEP 1 - Open TCP Socket connection to the RTSP Server
```C#
// Connect to a RTSP Server
tcp_socket = new Rtsp.RtspTcpTransport(host,port);
if (tcp_socket.Connected == false)
{
Console.WriteLine("Error - did not connect");
return;
}
```
This opens a connection for a 'TCP' mode RTSP/RTP session where RTP packets are set in the RTSP socket.
* STEP 2 - Create a RTSP Listener and attach it to the RTSP TCP Socket
```C#
// Connect a RTSP Listener to the TCP Socket to send messages and listen for replies
rtsp_client = new Rtsp.RtspListener(tcp_socket);
rtsp_client.MessageReceived += Rtsp_client_MessageReceived;
rtsp_client.DataReceived += Rtsp_client_DataReceived;
rtsp_client.Start(); // start reading messages from the server
```
The RTSP Listener class lets you SEND messages to the RTSP Server (see below).
The RTSP Listner class has a worker thread that listens for replies from the RTSP Server.
When replies are received the MessageReceived Event is fired.
When RTP packets are received the DataReceived Event is fired.
* STEP 3 - Send Messages to the RTSP Server
The samples below show how to send messages.
Send OPTIONS with this code :
```C#
Rtsp.Messages.RtspRequest options_message = new Rtsp. Messages.RtspRequestOptions();
options_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(options_message);
```
Send DESCRIBE with this code :
```C#
// send the Describe
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
// The reply will include the SDP data
```
Send SETUP with this code :
```C#
// the value of 'control' comes from parsing the SDP for the desired video or audio sub-stream
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");
rtsp_client.SendMessage(setup_message);
// The reply will include the Session
```
Send PLAY with this code :
```C#
// the value of 'session' comes from the reply of the SETUP command
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
```
* STEP 4 - Handle Replies when the MessageReceived event is fired
This example assumes the main program sends an OPTIONS Command.
It looks for a reply from the server for OPTIONS and then sends DESCRIBE.
It looks for a reply from the server for DESCRIBE and then sends SETUP (for the video stream)
It looks for a reply from the server for SETUP and then sends PLAY.
Once PLAY has been sent the video, in the form of RTP packets, will be received.
```C#
private void Rtsp_client_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
// send the DESCRIBE
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.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].GetMediaType() == Rtsp.Sdp.Media.MediaType.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(' ');
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");
rtsp_client.SendMessage(setup_message);
}
}
}
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.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
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
Console.WriteLine("Got reply from Play " + message.Command);
}
}
```
* STEP 5 - Handle RTP Video
This code handles each incoming RTP packet, combining RTP packets that are all part of the same frame of vdeo (using the Marker Bit).
Once a full frame is received it can be passed to a De-packetiser to get the compressed video data
```C#
List temporary_rtp_payloads = new List();
private void Rtsp_client_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
// RTP Packet Header
// 0 - Version, P, X, CC, M, PT and Sequence Number
//32 - Timestamp
//64 - SSRC
//96 - CSRCs (optional)
//nn - Extension ID and Length
//nn - Extension header
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_payload_type != video_payload)
{
Console.WriteLine("Ignoring this RTP payload");
return; // ignore this data
}
// 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
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
temporary_rtp_payloads.Add(rtp_payload);
if (rtp_marker == 1)
{
// Process the RTP frame
Process_RTP_Frame(temporary_rtp_payloads);
temporary_rtp_payloads.Clear();
}
}
```
* STEP 6 - Process RTP frame
An RTP frame consists of 1 or more RTP packets
H264 video is packed into one or more RTP packets and this sample extracts Normal Packing and
Fragmented Unit type A packing (the common two)
This example writes the video to a .264 file which can be played with FFPLAY
```C#
FileStream fs = null;
byte[] nal_header = new byte[]{ 0x00, 0x00, 0x00, 0x01 };
int norm, fu_a, fu_b, stap_a, stap_b, mtap16, mtap24 = 0; // stats counters
public void Process_RTP_Frame(Listrtp_payloads)
{
Console.WriteLine("RTP Data comprised of " + rtp_payloads.Count + " rtp packets");
if (fs == null)
{
// Create the file
String filename = "rtsp_capture_" + DateTime.Now.ToString("yyyyMMdd_HHmmss") + ".h264";
fs = new FileStream(filename, FileMode.Create);
// TODO. Get SPS and PPS from the SDP Attributes (the fmtp attribute) and write to the file
// for IP cameras that only out the SPS and PPS out-of-band
}
for (int payload_index = 0; payload_index < rtp_payloads.Count; payload_index++) {
// Examine the first rtp_payload and the first byte (the NAL header)
int nal_header_f_bit = (rtp_payloads[payload_index][0] >> 7) & 0x01;
int nal_header_nri = (rtp_payloads[payload_index][0] >> 5) & 0x03;
int nal_header_type = (rtp_payloads[payload_index][0] >> 0) & 0x1F;
// If the NAL Header Type is in the range 1..23 this is a normal NAL (not fragmented)
// So write the NAL to the file
if (nal_header_type >= 1 && nal_header_type <= 23)
{
Console.WriteLine("Normal NAL");
norm++;
fs.Write(nal_header, 0, nal_header.Length);
fs.Write(rtp_payloads[payload_index], 0, rtp_payloads[payload_index].Length);
}
else if (nal_header_type == 24)
{
// There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
Console.WriteLine("Agg STAP-A not supported");
stap_a++;
}
else if (nal_header_type == 25)
{
// There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
Console.WriteLine("Agg STAP-B not supported");
stap_b++;
}
else if (nal_header_type == 26)
{
// There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
Console.WriteLine("Agg MTAP16 not supported");
mtap16++;
}
else if (nal_header_type == 27)
{
// There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
Console.WriteLine("Agg MTAP24 not supported");
mtap24++;
}
else if (nal_header_type == 28)
{
Console.WriteLine("Fragmented Packet Type FU-A");
fu_a++;
// Parse Fragmentation Unit Header
int fu_header_s = (rtp_payloads[payload_index][1] >> 7) & 0x01; // start marker
int fu_header_e = (rtp_payloads[payload_index][1] >> 6) & 0x01; // end marker
int fu_header_r = (rtp_payloads[payload_index][1] >> 5) & 0x01; // reserved. should be 0
int fu_header_type = (rtp_payloads[payload_index][1] >> 0) & 0x1F; // Original NAL unit header
Console.WriteLine("Frag FU-A s="+fu_header_s + "e="+fu_header_e);
// Start Flag set
if (fu_header_s == 1)
{
// Write 00 00 00 01 header
fs.Write(nal_header, 0, nal_header.Length); // 0x00 0x00 0x00 0x01
// Modify the NAL Header that was at the start of the RTP packet
// Keep the F and NRI flags but substitute the type field with the fu_header_type
byte reconstructed_nal_type = (byte)((nal_header_nri << 5) + fu_header_type);
fs.WriteByte(reconstructed_nal_type); // NAL Unit Type
fs.Write(rtp_payloads[payload_index], 2, rtp_payloads[payload_index].Length - 2); // start after NAL Unit Type and FU Header byte
}
if (fu_header_s == 0)
{
// append this payload to the output NAL stream
// Data starts after the NAL Unit Type byte and the FU Header byte
fs.Write(rtp_payloads[payload_index], 2, rtp_payloads[payload_index].Length-2); // start after NAL Unit Type and FU Header byte
}
}
else if (nal_header_type == 29)
{
Console.WriteLine("Fragmented Packet FU-B not supported");
fu_b++;
}
else
{
Console.WriteLine("Unknown NAL header " + nal_header_type);
}
}
// ensure video is written to disk
fs.Flush(true);
// Print totals
Console.WriteLine("Norm=" + norm + " ST-A=" + stap_a + " ST-B=" + stap_b + " M16=" + mtap16 + " M24=" + mtap24 + " FU-A=" + fu_a + " FU-B=" + fu_b);
}
```
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