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

This repository contains the source for a C# .NET library with full support for the Session Initiation Protocol (SIP) and the Real-time Transport Protocol (RTP).

This library does NOT provide any media (audio and video) handling. For Windows the companion SIPSorceryMedia library provides audio & video functions for rendering & capture. This project can be used for SIP signalling and to send and receive RTP packets.

NEW (Feb/Mar 2020): Pre-release support for Web Real-Time Communication (WebRTC) for early adopters. See Getting Started WebRTC.

Installation

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

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

Install-Package SIPSorcery

SIPSorceryMedia Install

.NET Core does not provide any audio or video capture capabilities nor any audio rendering function (UWP does and there are some tricks to get at its API but it's fragile). A lot of the uses for SIP and WebRTC revolve around such capabilities and functions. A companion Windows specific library which fills this gap is maintained at SIPSorceryMedia. For non-Windows applications there is no known .NET Core library that provides audio and video functions.

The RtpAVSession class from the SIPSorceryMedia library wraps the audio and video functions and integrates with the SIPUserAgent class for ease of use.

Note that the RtpAVSession class is only available as a pre-release in version 4.0.28-pre and greater.

The SIPSorceryMedia library is compliant with .NET Core 3.1. It is available via NuGet:

For .NET Core:

dotnet add package SIPSorceryMedia -v 4.0.28-pre

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

Install-Package SIPSorceryMedia -v 4.0.28-pre

Documentation

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

Getting Started SIP/VoIP

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.

dotnet new console --name SIPGetStarted
cd SIPGetStarted
dotnet add package SIPSorcery -v 4.0.28-pre
dotnet add package SIPSorceryMedia -v 4.0.28-pre
edit Program.cs and paste in the contents below
dotnet run

using System;
using System.Net.Sockets;
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(string[] args)
        {
            Console.WriteLine("SIP Get Started");
            
            var sipTransport = new SIPTransport();
            var userAgent = new SIPUserAgent(sipTransport, null);
            var rtpSession = new RtpAVSession(AddressFamily.InterNetwork, new AudioOptions { AudioSource = AudioSourcesEnum.Microphone }, 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")}.");
        }
    }
}

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 base class for RtpAVSession.

The examples folder contains sample code to demonstrate other common SIP use cases such as:

• Attended Transfers,
• Blind Transfers,
• Call Hold,
• Sending DTMF tones.

The full list of SIP examples available is:

• Get Started: Simplest example. Demonstrates how to place a SIP call.
• Get Started Video: Adds video to the Get Started example.
• SIP Proxy: Very rudimentary example for a SIP Proxy and SIP Registrar.
• Registration Client: Demonstrates how to use the SIPRegistrationUserAgent class to register with a SIP Registrar server.
• SIP Call Client: Demonstrates how to use SIPClientUserAgent class to place a call to a SIP server user agent.
• SIP Call Server: Demonstrates how to use the SIPServerUserAgent class to receive a call from a SIP client user agent.
• SoftPhone: A very rudimentary SIP softphone implementation.
• Get Started Web Socket: An example of how to create a web socket listener to send and receive SIP messages. An explanation of the example is available here.
• STUN Server: An example of how to create a basic STUN (RFC3849) server. An explanation of the example is available here.
• Call Hold and Blind Transfer: An example of how to place a call on hold and perform a blind transfer using a REFER request as specified in RFC3515. An explanation of the example is available here.
• Call Attended Transfer: An example of how to perform an attended transfer. An explanation of the example is available here.
• Send DTMF (as RTP events): An example of how to send DTMF tones using RTP events as specified in RFC2833. An explanation of the example is available here.

Getting Started WebRTC

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.

dotnet new console --name WebRTCGetStarted
cd WebRTCGetStarted
dotnet add package SIPSorcery -v 4.0.28-pre
dotnet add package SIPSorceryMedia -v 4.0.28-pre
edit Program.cs and paste in the contents below
dotnet run

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

namespace WebRTCGetStarted
{
    class Program
    {
        private const string DTLS_CERTIFICATE_PATH = "certs/localhost.pem";
        private const string DTLS_KEY_PATH = "certs/localhost_key.pem";
        private const string DTLS_CERTIFICATE_FINGERPRINT = "sha-256 C6:ED:8C:9D:06:50:77:23:0A:4A:D8:42:68:29:D0:70:2F:BB:C7:72:EC:98:5C:62:07:1B:0C:5D:CB:CE:BE:CD";
        
        static async Task Main(string[] args)
        {
            Console.WriteLine("Get Started WebRTC");
            
            var webRtcSession = new WebRtcSession(
                AddressFamily.InterNetwork,
                DTLS_CERTIFICATE_FINGERPRINT,
                null,
                null);

            MediaStreamTrack videoTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) });
            webRtcSession.addTrack(videoTrack);
            
            var offerSdp = await webRtcSession.createOffer(null);
            webRtcSession.setLocalDescription(new RTCSessionDescription { sdp = offerSdp, type = RTCSdpType.offer });
            
            // At this point the SDP offer and answer need to be exchanged with the remote peer.
            
            var answerSdp = SDP.ParseSDPDescription(sdpAnswer);
            webRtcSession.setRemoteDescription(new RTCSessionDescription { sdp = answerSdp, type = RTCSdpType.answer }); 
            
            var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
            webRtcSession.OnClose += (reason) => dtls.Shutdown();
            
            dtls.DoHandshakeAsServer((ulong)webRtcSession.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle);

            if (dtls.IsHandshakeComplete())
            {
                var srtpSendContext = new Srtp(dtls, false);
                var srtpReceiveContext = new Srtp(dtls, true);

                webRtcSession.SetSecurityContext(
                    srtpSendContext.ProtectRTP,
                    srtpReceiveContext.UnprotectRTP,
                    srtpSendContext.ProtectRTCP,
                    srtpReceiveContext.UnprotectRTCP);

                Console.WriteLine("DTLS handshake completed.");
            }
            else
            {
               Console.WriteLine("DTLS handshake failed.");
            }
            
            // If the DTLS key exchange succeeded then secure RTP packets can now be exchanged between the peers.
        }
    }
}

The key class for using WebRTC is described in this article:

• WebRTCSession

The full list of WebRTC examples available is:

• WebRTCTestPatternServer: The simplest example. This program serves up a test pattern video stream to a WebRTC peer.
• WebRTCServer: This example extends the test pattern example and can act as a media source for a peer. It has two source options:
  • An mp4 file.
  • Capture devices (webcam and microphone). The example includes an html file which runs in a Browser and will connect to a sample program running on the same machine.
• WebRTCReceiver: A receive only example. It attempts to connect to a WebRTC peer and display the video stream that it receives.

The WebRtcSession class and all WebRTC functionality in this library are still under heavy development. There are large blocks of functionality still missing, particularly ICE and codec support. All issues and PR's are very welcome.