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Added some SIP demos tuned for k8s. Added some extra classes to help ith NAT & ICE servers (#1418)
* Added some SIP demos tuned for k8s. Added some extra classes to help with NAT & ICE servers. * Put back namespace to help with PR comparison. * Refactored RtpIceChannel to use new IceServerResolver class. * Demo checks. * Fixed typo.
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‎examples/SIPExamples/SIPCallServer/readme-docker.txt‎

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examples/SIPExamples/SIPCallServer/Dockerfile renamed to examples/SIPExamples/SIPCloudCallServer/Dockerfile

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@@ -24,4 +24,4 @@ EXPOSE 5060/udp
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EXPOSE 5061/tcp
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# Run the server
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ENTRYPOINT ["dotnet", "SIPCallServer.dll"]
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ENTRYPOINT ["dotnet", "SIPCloudCallServer.dll"]

‎examples/SIPExamples/SIPCloudCallServer/Program.cs‎

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//-----------------------------------------------------------------------------
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// Filename: Program.cs
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//
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// Description: An example SIP server program to accept and initiate calls.
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// This example is derived from teh SIPCallServer example and is tuned to use
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// in a Kuberenetes cluster.
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//
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// Author(s):
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// Aaron Clauson (aaron@sipsorcery.com)
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//
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// History:
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// 25 May 2025 Aaron Clauson Created, Dublin, Ireland.
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//
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// License:
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// BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
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//-----------------------------------------------------------------------------
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using System;
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using System.Collections.Concurrent;
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using System.Collections.Generic;
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using System.Net;
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using System.Security.Cryptography.X509Certificates;
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using System.Threading;
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using System.Threading.Tasks;
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using Microsoft.Extensions.Logging;
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using Microsoft.Extensions.Logging.Abstractions;
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using Serilog;
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using Serilog.Extensions.Logging;
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using SIPSorcery.Media;
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using SIPSorcery.Net;
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using SIPSorcery.SIP;
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using SIPSorcery.SIP.App;
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using SIPSorceryMedia.Abstractions;
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namespace SIPSorcery
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{
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class Program
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{
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private const string SIP_CONTACT_HOST_ENV_VAR = "SIP_CONTACT_HOST";
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private const string RTP_PORT_ENV_VAR = "RTP_PORT";
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private const string STUN_URL_ENV_VAR = "STUN_URL";
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private static int SIP_LISTEN_PORT = 5060;
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private static int SIPS_LISTEN_PORT = 5061;
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private static string SIPS_CERTIFICATE_PATH = "localhost.pfx";
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private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance;
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private static SIPTransport _sipTransport;
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/// <summary>
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/// Keeps track of the current active calls. It includes both received and placed calls.
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/// </summary>
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private static ConcurrentDictionary<string, SIPUserAgent> _calls = new ConcurrentDictionary<string, SIPUserAgent>();
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private static int _rtpPort = 0;
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private static IPAddress _publicIpAddress = null;
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static void Main()
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{
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Console.WriteLine("SIPSorcery SIP Call Server example.");
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Log = AddConsoleLogger();
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// Set up a default SIP transport.
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_sipTransport = new SIPTransport();
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if (!string.IsNullOrWhiteSpace(Environment.GetEnvironmentVariable(SIP_CONTACT_HOST_ENV_VAR)))
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{
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_sipTransport.ContactHost = Environment.GetEnvironmentVariable(SIP_CONTACT_HOST_ENV_VAR);
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}
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if (!string.IsNullOrWhiteSpace(Environment.GetEnvironmentVariable(RTP_PORT_ENV_VAR)))
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{
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int.TryParse(Environment.GetEnvironmentVariable(RTP_PORT_ENV_VAR), out _rtpPort);
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}
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if (!string.IsNullOrWhiteSpace(Environment.GetEnvironmentVariable(STUN_URL_ENV_VAR)))
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{
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_publicIpAddress = STUNClient.GetPublicIPAddress(Environment.GetEnvironmentVariable(STUN_URL_ENV_VAR));
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}
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_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
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_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
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_sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
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var localhostCertificate = new X509Certificate2(SIPS_CERTIFICATE_PATH);
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_sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(IPAddress.Any, SIPS_LISTEN_PORT)));
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// If it's desired to listen on a single IP address use the equivalent of:
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//_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Parse("192.168.11.50"), SIP_LISTEN_PORT)));
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_sipTransport.EnableTraceLogs();
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_sipTransport.SIPTransportRequestReceived += OnRequest;
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// Ctrl-c will gracefully exit the call at any point.
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ManualResetEvent exitMre = new ManualResetEvent(false);
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Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
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{
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e.Cancel = true;
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exitMre.Set();
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};
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// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
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exitMre.WaitOne();
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Log.LogInformation("Exiting...");
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if (_sipTransport != null)
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{
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Log.LogInformation("Shutting down SIP transport...");
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_sipTransport.Shutdown();
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}
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}
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/// <summary>
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/// Example of how to create a basic RTP session object and hook up the event handlers.
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/// </summary>
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/// <param name="ua">The user agent the RTP session is being created for.</param>
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/// <param name="dst">THe destination specified on an incoming call. Can be used to
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/// set the audio source.</param>
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/// <returns>A new RTP session object.</returns>
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private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst, int bindPort)
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{
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List<AudioCodecsEnum> codecs = new List<AudioCodecsEnum> { AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA, AudioCodecsEnum.G722 };
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var audioSource = AudioSourcesEnum.SineWave;
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if (string.IsNullOrEmpty(dst) || !Enum.TryParse(dst, out audioSource))
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{
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audioSource = AudioSourcesEnum.Music;
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}
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Log.LogInformation($"RTP audio session source set to {audioSource}.");
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AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = audioSource });
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audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec));
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var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = audioExtrasSource }, bindPort: bindPort);
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rtpAudioSession.AcceptRtpFromAny = true;
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// Wire up the event handler for RTP packets received from the remote party.
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rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, ep, type, rtp);
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rtpAudioSession.OnTimeout += (mediaType) =>
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{
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if (ua?.Dialogue != null)
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{
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Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
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}
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else
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{
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Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
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}
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ua.Hangup();
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};
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return rtpAudioSession;
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}
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/// <summary>
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/// Event handler for receiving RTP packets.
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/// </summary>
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/// <param name="ua">The SIP user agent associated with the RTP session.</param>
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/// <param name="type">The media type of the RTP packet (audio or video).</param>
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/// <param name="rtpPacket">The RTP packet received from the remote party.</param>
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private static void OnRtpPacketReceived(SIPUserAgent ua, IPEndPoint remoteEp, SDPMediaTypesEnum type, RTPPacket rtpPacket)
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{
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// The raw audio data is available in rtpPacket.Payload.
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//Log.LogTrace($"OnRtpPacketReceived from {remoteEp}.");
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}
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/// <summary>
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/// Event handler for receiving a DTMF tone.
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/// </summary>
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/// <param name="ua">The user agent that received the DTMF tone.</param>
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/// <param name="key">The DTMF tone.</param>
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/// <param name="duration">The duration in milliseconds of the tone.</param>
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private static void OnDtmfTone(SIPUserAgent ua, byte key, int duration)
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{
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string callID = ua.Dialogue.CallId;
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Log.LogInformation($"Call {callID} received DTMF tone {key}, duration {duration}ms.");
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}
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/// <summary>
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/// Because this is a server user agent the SIP transport must start listening for client user agents.
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/// </summary>
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private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
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{
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try
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{
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if (sipRequest.Header.From != null &&
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sipRequest.Header.From.FromTag != null &&
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sipRequest.Header.To != null &&
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sipRequest.Header.To.ToTag != null)
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{
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// This is an in-dialog request that will be handled directly by a user agent instance.
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}
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else if (sipRequest.Method == SIPMethodsEnum.INVITE)
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{
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Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
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SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
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ua.OnCallHungup += OnHangup;
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ua.ServerCallCancelled += (uas, cancelReq) => Log.LogDebug("Incoming call cancelled by remote party.");
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ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
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ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
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//ua.OnTransactionTraceMessage += (tx, msg) => Log.LogDebug($"uas tx {tx.TransactionId}: {msg}");
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ua.ServerCallRingTimeout += (uas) =>
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{
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Log.LogWarning($"Incoming call timed out in {uas.ClientTransaction.TransactionState} state waiting for client ACK, terminating.");
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ua.Hangup();
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};
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var uas = ua.AcceptCall(sipRequest);
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var rtpSession = CreateRtpSession(ua, sipRequest.URI.User, _rtpPort);
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await ua.Answer(uas, rtpSession, _publicIpAddress);
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if (ua.IsCallActive)
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{
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await rtpSession.Start();
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_calls.TryAdd(ua.Dialogue.CallId, ua);
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}
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}
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else if (sipRequest.Method == SIPMethodsEnum.BYE)
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{
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SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
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await _sipTransport.SendResponseAsync(byeResponse);
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}
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else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
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{
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SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
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await _sipTransport.SendResponseAsync(notAllowededResponse);
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}
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else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
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{
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SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
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await _sipTransport.SendResponseAsync(optionsResponse);
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}
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}
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catch (Exception reqExcp)
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{
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Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
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}
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}
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/// <summary>
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/// Remove call from the active calls list.
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/// </summary>
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/// <param name="dialogue">The dialogue that was hungup.</param>
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private static void OnHangup(SIPDialogue dialogue)
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{
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if (dialogue != null)
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{
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string callID = dialogue.CallId;
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if (_calls.ContainsKey(callID))
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{
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if (_calls.TryRemove(callID, out var ua))
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{
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// This app only uses each SIP user agent once so here the agent is
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// explicitly closed to prevent is responding to any new SIP requests.
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ua.Close();
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}
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}
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}
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}
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/// <summary>
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/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
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/// </summary>
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private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
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{
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var serilogLogger = new LoggerConfiguration()
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.Enrich.FromLogContext()
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.MinimumLevel.Is(Serilog.Events.LogEventLevel.Verbose)
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.WriteTo.Console()
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.CreateLogger();
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var factory = new SerilogLoggerFactory(serilogLogger);
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SIPSorcery.LogFactory.Set(factory);
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return factory.CreateLogger<Program>();
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}
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}
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}

‎examples/SIPExamples/SIPCloudCallServer/SIPCloudCallServer.csproj‎

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<Project Sdk="Microsoft.NET.Sdk">
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<PropertyGroup>
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<OutputType>Exe</OutputType>
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<TargetFramework>net8.0</TargetFramework>
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</PropertyGroup>
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<ItemGroup>
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<Compile Remove="local-nuget\**" />
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<EmbeddedResource Remove="local-nuget\**" />
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<None Remove="local-nuget\**" />
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</ItemGroup>
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<ItemGroup>
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<None Remove="Dockerfile" />
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<None Remove="readme-docker.txt" />
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</ItemGroup>
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<ItemGroup>
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<PackageReference Include="Microsoft.Extensions.Logging" Version="9.0.0" />
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<PackageReference Include="Serilog.Extensions.Logging" Version="9.0.0" />
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<PackageReference Include="Serilog.Sinks.Console" Version="6.0.0" />
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<PackageReference Include="SIPSorcery" Version="8.0.19-pre" />
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</ItemGroup>
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<ItemGroup>
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<None Update="localhost.pfx">
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<CopyToOutputDirectory>PreserveNewest</CopyToOutputDirectory>
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</None>
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</ItemGroup>
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</Project>

‎examples/SIPExamples/SIPCloudCallServer/localhost.pfx‎

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‎examples/SIPExamples/SIPCloudCallServer/readme-docker.txt‎

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C:\dev\sipsorcery\src> dotnet pack SIPSorcery.csproj --configuration Debug --output c:\dev\local-nuget
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docker build -t sipsorcery/sipcloudcallserver:0.1 .
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docker push sipsorcery/sipcloudcallserver:0.1

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