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jimm98yjimm98y
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F Added nugets, API improvements
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Lines changed: 89 additions & 51 deletions

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‎README.md‎

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@@ -1 +1,8 @@
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# SharpRealTimeStreaming
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# SharpRealTimeStreaming
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This is just a thin wrapper around the fantastic SharpRTSP, mostly made of their sample code with some API enhancements to make it easier to use.
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## SharpRTSPClient
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Simple RTSP client that supports H264, H265 for video and AAC for audio.
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## SharpRTSPServer
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Simple RTSP server that supports H264, H265 for video and AAC, PCMU and PCMA for audio.

‎src/RTSPClientApp/Program.cs‎

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@@ -1,5 +1,4 @@
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using SharpRTSPClient;
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using SharpRTSPClient;
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using System;
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const string rtspUri = "rtsp://127.0.0.1:8554";
@@ -13,7 +12,7 @@
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client.NewAudioStream += (sender, e) => Console.WriteLine(e.ToString());
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client.ReceivedAudioData += (sender, e) => Console.Write("+");
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16-
client.Connect(rtspUri, RTSPClient.RtpTransport.TCP, userName, password);
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client.Connect(rtspUri, RTPTransport.TCP, userName, password);
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Console.WriteLine("Press any key to exit");
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while (!Console.KeyAvailable)

‎src/RTSPServerApp/Program.cs‎

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@@ -1,5 +1,4 @@
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using SharpMp4;
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using SharpMp4;
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using SharpRTSPServer;
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using System;
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using System.Collections.Generic;

‎src/SharpRTSPClient/H264StreamConfigurationData.cs‎

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@@ -1,8 +1,6 @@
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using System.Collections.Generic;
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namespace SharpRTSPClient
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namespace SharpRTSPClient
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{
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public class H264StreamConfigurationData : IVideoStreamConfigurationData
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public class H264StreamConfigurationData : IStreamConfigurationData
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{
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public byte[] SPS { get; set; }
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public byte[] PPS { get; set; }
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{
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return $"SPS: {Utilities.ToHexString(SPS)}\r\nPPS: {Utilities.ToHexString(PPS)}";
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}
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public IEnumerable<byte[]> GetNALUs()
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{
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return new byte[][] { SPS, PPS };
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}
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}
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}

‎src/SharpRTSPClient/H265StreamConfigurationData.cs‎

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@@ -1,8 +1,6 @@
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using System.Collections.Generic;
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namespace SharpRTSPClient
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namespace SharpRTSPClient
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{
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public class H265StreamConfigurationData : IVideoStreamConfigurationData
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public class H265StreamConfigurationData : IStreamConfigurationData
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{
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public byte[] VPS { get; set; }
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public byte[] SPS { get; set; }
@@ -22,10 +20,5 @@ public override string ToString()
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{
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return $"VPS: {Utilities.ToHexString(VPS)}\r\nSPS: {Utilities.ToHexString(SPS)}\r\nPPS: {Utilities.ToHexString(PPS)}";
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}
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public IEnumerable<byte[]> GetNALUs()
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{
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return new byte[][] { VPS, SPS, PPS };
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}
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}
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}

‎src/SharpRTSPClient/RTSPClient.cs‎

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@@ -14,6 +14,20 @@
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namespace SharpRTSPClient
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{
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public enum RTPTransport
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{
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UDP,
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TCP,
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MULTICAST
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}
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public enum MediaRequest
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{
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VIDEO_ONLY,
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AUDIO_ONLY,
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VIDEO_AND_AUDIO
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}
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/// <summary>
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/// RTSP client.
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/// </summary>
@@ -40,28 +54,14 @@ public class RTSPClient : IDisposable
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public bool AutoPlay { get; set; } = true;
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public enum RtpTransport
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{
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UDP,
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TCP,
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MULTICAST
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};
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public enum MediaRequest
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{
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VIDEO_ONLY,
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AUDIO_ONLY,
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VIDEO_AND_AUDIO
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};
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private enum RtspStatus { WaitingToConnect, Connecting, ConnectFailed, Connected };
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private IRtspTransport _rtspSocket; // RTSP connection
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private RtspStatus _rtspSocketStatus = RtspStatus.WaitingToConnect;
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// this wraps around a the RTSP tcpSocket stream
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private RtspListener _rtspClient;
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private RtpTransport _rtpTransport = RtpTransport.UDP; // Mode, either RTP over UDP or RTP over TCP using the RTSP socket
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private RTPTransport _rtpTransport = RTPTransport.UDP; // Mode, either RTP over UDP or RTP over TCP using the RTSP socket
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private IRtpTransport _videoRtpTransport;
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private IRtpTransport _audioRtpTransport;
@@ -131,12 +131,12 @@ public RTSPClient(ILoggerFactory loggerFactory)
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/// Connect.
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/// </summary>
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/// <param name="url">URL to connect to.</param>
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/// <param name="rtpTransport">Type of the RTP transport <see cref="RtpTransport"/>.</param>
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/// <param name="rtpTransport">Type of the RTP transport <see cref="RTPTransport"/>.</param>
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/// <param name="username">User name.</param>
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/// <param name="password">Password.</param>
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/// <param name="mediaRequest">Media request type <see cref="MediaRequest>."/></param>
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/// <param name="playbackSession">Playback session.</param>
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public void Connect(string url, RtpTransport rtpTransport, string username = null, string password = null, MediaRequest mediaRequest = MediaRequest.VIDEO_AND_AUDIO, bool playbackSession = false)
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public void Connect(string url, RTPTransport rtpTransport, string username = null, string password = null, MediaRequest mediaRequest = MediaRequest.VIDEO_AND_AUDIO, bool playbackSession = false)
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{
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RtspUtils.RegisterUri();
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@@ -213,14 +213,14 @@ public void Connect(string url, RtpTransport rtpTransport, string username = nul
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// If the RTP transport is MULTICAST, we have to wait for the SETUP message to get the Multicast Address from the RTSP server
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this._rtpTransport = rtpTransport;
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216-
if (rtpTransport == RtpTransport.UDP)
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if (rtpTransport == RTPTransport.UDP)
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{
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// give a range of 500 pairs (1000 addresses) to try incase some address are in use
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_videoRtpTransport = new UDPSocket(50000, 51000);
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_audioRtpTransport = new UDPSocket(50000, 51000);
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}
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223-
if (rtpTransport == RtpTransport.TCP)
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if (rtpTransport == RTPTransport.TCP)
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{
225225
int nextFreeRtpChannel = 0;
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_videoRtpTransport = new RtpTcpTransport(_rtspClient)
@@ -1118,14 +1118,14 @@ private RtspTransport CalculateTransport(IRtpTransport transport)
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{
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// Server interleaves the RTP packets over the RTSP connection
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// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
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case RtpTransport.TCP:
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case RTPTransport.TCP:
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return new RtspTransport()
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{
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LowerTransport = RtspTransport.LowerTransportType.TCP,
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// Eg Channel 0 for RTP video data. Channel 1 for RTCP status reports
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Interleaved = (transport as RtpTcpTransport)?.Channels ?? throw new ApplicationException("TCP transport asked and no tcp channel allocated"),
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};
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case RtpTransport.UDP:
1128+
case RTPTransport.UDP:
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return new RtspTransport()
11301130
{
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LowerTransport = RtspTransport.LowerTransportType.UDP,
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// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
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// using Multicast Address and Ports that are in the reply to the SETUP message
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// Example for MULTICAST mode Transport: RTP/AVP;multicast
1138-
case RtpTransport.MULTICAST:
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case RTPTransport.MULTICAST:
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return new RtspTransport()
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{
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LowerTransport = RtspTransport.LowerTransportType.UDP,
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public interface IStreamConfigurationData
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{ }
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1225-
public interface IVideoStreamConfigurationData : IStreamConfigurationData
1226-
{
1227-
IEnumerable<byte[]> GetNALUs();
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}
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public class SimpleDataEventArgs : EventArgs
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{
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public SimpleDataEventArgs(IEnumerable<ReadOnlyMemory<byte>> data, DateTime timeStamp)

‎src/SharpRTSPClient/SharpRTSPClient.csproj‎

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<PackageReference Include="SharpRTSP" Version="1.4.3" />
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</ItemGroup>
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<PropertyGroup>
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<GeneratePackageOnBuild>True</GeneratePackageOnBuild>
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<AllowUnsafeBlocks>True</AllowUnsafeBlocks>
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<Nullable>disable</Nullable>
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<Title>$(ProjectName)</Title>
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<Version>0.0.1</Version>
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<Authors>Lukas Volf</Authors>
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<Copyright>MIT</Copyright>
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<PackageProjectUrl>https://github.com/jimm98y/SharpRealTimeStreaming</PackageProjectUrl>
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<PackageReadmeFile>README.md</PackageReadmeFile>
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<RepositoryUrl>https://github.com/jimm98y/SharpRealTimeStreaming</RepositoryUrl>
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<RepositoryType>git</RepositoryType>
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<PackageTags>rtsp;rtp;video;streaming;h264;h265;aac;netstandard20</PackageTags>
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<NeutralLanguage>en-US</NeutralLanguage>
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<AssemblyVersion>1.0.0</AssemblyVersion>
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<FileVersion>$(Version)</FileVersion>
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<PackageLicenseExpression>MIT</PackageLicenseExpression>
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<Description>Simple RTSP client. Supports H264, H265 and AAC.</Description>
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</PropertyGroup>
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<ItemGroup>
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<None Include="..\..\README.md">
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<Pack>True</Pack>
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<PackagePath>\</PackagePath>
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</None>
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</ItemGroup>
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</Project>

‎src/SharpRTSPServer/SharpRTSPServer.csproj‎

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<PackageReference Include="SharpRTSP" Version="1.4.3" />
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</ItemGroup>
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<PropertyGroup>
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<GeneratePackageOnBuild>True</GeneratePackageOnBuild>
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<AllowUnsafeBlocks>True</AllowUnsafeBlocks>
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<Nullable>disable</Nullable>
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<Title>$(ProjectName)</Title>
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<Version>0.0.1</Version>
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<Authors>Lukas Volf</Authors>
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<Copyright>MIT</Copyright>
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<PackageProjectUrl>https://github.com/jimm98y/SharpRealTimeStreaming</PackageProjectUrl>
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<PackageReadmeFile>README.md</PackageReadmeFile>
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<RepositoryUrl>https://github.com/jimm98y/SharpRealTimeStreaming</RepositoryUrl>
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<RepositoryType>git</RepositoryType>
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<PackageTags>rtsp;rtp;video;streaming;h264;h265;aac;netstandard20</PackageTags>
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<NeutralLanguage>en-US</NeutralLanguage>
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<AssemblyVersion>1.0.0</AssemblyVersion>
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<FileVersion>$(Version)</FileVersion>
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<PackageLicenseExpression>MIT</PackageLicenseExpression>
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<Description>Simple RTSP server. Supports H264, H265 and AAC.</Description>
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</PropertyGroup>
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<ItemGroup>
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<None Include="..\..\README.md">
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<Pack>True</Pack>
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<PackagePath>\</PackagePath>
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</None>
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</ItemGroup>
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</Project>

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