Target	SIPSorcery	Examples (Windows Only)	Softphone (Windows Only)
net46
netstandard2.0
dotnetcore3.1	Windows MacOS Ubuntu

This fully C# library can be used to add Real-time Communications, typically audio and video calls, to .NET Core applications.

The diagram below is a high level overview of a Real-time audio and video call between Alice and Bob. It illustrates where the SIPSorcery library can help.

Supports both VoIP (get started) and WebRTC (get started).

Some of the protocols supported:

• Session Initiation Protocol (SIP),
• Real-time Transport Protocol (RTP),
• Web Real-time Communications (WebRTC),
• Interactive Connectivity Establishment (ICE),
• And more.

Caveats:

• This library does not provide access to audio and video devices. Some Windows specific examples of using 3rd party libraries are provided,
• This library provides only a small number of audio and video codecs (G711, G722 and MJPEG). Additional codecs, particularly video ones, require C++ libraries.

The library is compliant with .NET Standard 2.0, .Net Core 3.1 and .NET Framework 4.6. It is available via NuGet.

For .NET Core:

dotnet add package SIPSorcery -v 4.0.60-pre

With Visual Studio Package Manager Console (or search for SIPSorcery on NuGet):

Install-Package SIPSorcery -v 4.0.60-pre

Class reference documentation and articles explaining common usage are available at https://sipsorcery.github.io/sipsorcery/.

The simplest possible example to place an audio-only SIP call is shown below. This example relies on the Windows specific SIPSorceryMedia library to play the received audio and only works on Windows (due to lack of .NET Core audio device support on non-Windows platforms).

dotnet new console --name SIPGetStarted
cd SIPGetStarted
dotnet add package SIPSorcery -v 4.0.60-pre
dotnet add package SIPSorceryMedia -v 4.0.60-pre
code . # If you have Visual Studio Code https://code.visualstudio.com installed.
# edit Program.cs and paste in the contents below.
dotnet run
# if successful you will hear the current time read out.
ctrl-c

using System;
using System.Threading.Tasks;
using SIPSorcery.SIP;
using SIPSorcery.SIP.App;
using SIPSorcery.Media;

namespace SIPGetStarted
{
    class Program
    {
         private static string DESTINATION = "time@sipsorcery.com";
        
        static async Task Main()
        {
            Console.WriteLine("SIP Get Started");
            
            var sipTransport = new SIPTransport();
            var userAgent = new SIPUserAgent(sipTransport, null);
            var rtpSession = new RtpAVSession(new AudioOptions { AudioSource = AudioSourcesEnum.CaptureDevice }, null);

            // Place the call and wait for the result.
            bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
            Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");

            Console.WriteLine("Press any key to hangup and exit.");
            Console.ReadLine();
        }
    }
}

The GetStarted example contains the full source and project file for the example above.

The three key classes in the above example are described in dedicated articles:

• SIPTransport,
• SIPUserAgent,
• RTPSession.

The examples folder contains sample code to demonstrate other common SIP/VoIP cases.

The core of the code required to establish a WebRTC connection is demonstrated below. The code shown will build but will not establish a connection due to no mechanism to exchange the SDP offer and answer between peers. A full working example with a web socket signalling mechanism is available in the WebRTCTestPatternServer example.

If you are familiar with the WebRTC javascript API the API in this project aims to be as close to it as possible.

dotnet new console --name WebRTCGetStarted
cd WebRTCGetStarted
dotnet add package SIPSorcery -v 4.0.60-pre
dotnet add package SIPSorceryMedia -v 4.0.60-pre
code . # If you have Visual Studio Code (https://code.visualstudio.com) installed
# edit Program.cs and paste in the contents below.
dotnet run

using System;
using System.Collections.Generic;
using System.Threading.Tasks;
using SIPSorcery.Net;

namespace WebRTCGetStarted
{
    class Program
    {
        static async Task Main()
        {
            Console.WriteLine("Get Started WebRTC");
            
            var pc = new RTCPeerConnection(null);

            MediaStreamTrack videoTrack = new MediaStreamTrack(
              SDPMediaTypesEnum.video, 
              false, 
              new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, 
              MediaStreamStatusEnum.SendOnly);
            pc.addTrack(videoTrack);

            pc.oniceconnectionstatechange += (state) => Console.WriteLine($"ICE connection state change to {state}.");
            pc.onconnectionstatechange += (state) => Console.WriteLine($"Peer connection state change to {state}.");

            var offerSdp = pc.createOffer(null);
            await pc.setLocalDescription(offerSdp);

            Console.ReadLine();
        }
    }
}

The examples folder contains sample code to demonstrate other common WebRTC cases.