SIP to OpenAI

README.md

@@ -167,6 +167,7 @@ Several sample applications demonstrating different scenarios are available in t
 - **LocalFunctions** – showcases the local function calling feature.
 - **GetPaid** – extends local functions to simulate payment requests.
 - **GetStartedDI** – illustrates using the library with .NET Dependency Injection.
+- **GetStartedSIP** – demonstrates how to create a SIP-to-OpenAI WebRTC gateway that receives SIP (VoIP) calls and bridges to OpenAI.
 - **ASP.NET Get Started** – ASP.NET application bridging a browser WebRTC client to OpenAI.
 - **ASP.NET Local Function** – ASP.NET application that builds on the Get Started example and adds a local function to tailor OpenAI responses.
 

examples/GetStartedSIP/GetStartedSIP.csproj

@@ -0,0 +1,31 @@
+<Project Sdk="Microsoft.NET.Sdk">
+
+  <PropertyGroup>
+    <OutputType>Exe</OutputType>
+    <TargetFramework>net8.0</TargetFramework>
+    <LangVersion>12.0</LangVersion>
+    <Nullable>enable</Nullable>
+    <TreatWarningsAsErrors>true</TreatWarningsAsErrors>
+    <SuppressTfmSupportBuildWarnings>true</SuppressTfmSupportBuildWarnings>
+  </PropertyGroup>
+
+  <PropertyGroup>
+    <AllowUnsafeBlocks>true</AllowUnsafeBlocks>
+  </PropertyGroup>
+
+  <ItemGroup>
+    <PackageReference Include="LanguageExt.Core" Version="4.4.9" />
+    <PackageReference Include="Microsoft.Extensions.Logging" Version="9.0.0" />
+    <PackageReference Include="Serilog.Extensions.Logging" Version="9.0.0" />
+    <PackageReference Include="Serilog.Sinks.Console" Version="6.0.0" />
+  </ItemGroup>
+
+  <ItemGroup>
+    <ProjectReference Include="..\..\src\SIPSorcery.OpenAI.WebRTC.csproj" />
+  </ItemGroup>
+
+  <ItemGroup>
+    <Folder Include="Properties\" />
+  </ItemGroup>
+
+</Project>

examples/GetStartedSIP/Program.cs

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+//-----------------------------------------------------------------------------
+// Filename: Program.cs
+//
+// Description: An example showing how to use SIPSorcery with OpenAI's WebRTC endpoint.
+// This demo shows the concept of how you could bridge SIP calls to OpenAI, though
+// a complete implementation would require additional SIP handling logic.
+//
+// Usage:
+// set OPENAI_API_KEY=your_openai_key
+// dotnet run
+//
+// Author(s):
+// Aaron Clauson ([email protected])
+// 
+// History:
+// 09 Aug 2025	Aaron Clauson	Created, Dublin, Ireland.
+//
+// License: 
+// BSD 3-Clause "New" or "Revised" License and the additional
+// BDS BY-NC-SA restriction, see included LICENSE.md file.
+//-----------------------------------------------------------------------------
+
+using Microsoft.Extensions.Logging;
+using Serilog;
+using Serilog.Extensions.Logging;
+using SIPSorcery.Net;
+using SIPSorcery.OpenAIWebRTC;
+using SIPSorcery.OpenAIWebRTC.Models;
+using SIPSorcery.SIP;
+using SIPSorcery.SIP.App;
+using SIPSorceryMedia.Abstractions;
+using System;
+using System.Collections.Concurrent;
+using System.Net;
+using System.Threading.Tasks;
+
+namespace demo;
+
+record SIPToOpenAiCall(SIPUserAgent ua, RTPSession voip, WebRTCEndPoint? webrtc);
+
+class Program
+{
+    private static int SIP_LISTEN_PORT = 5060;
+
+    /// <summary>
+    /// Keeps track of the current active calls. It includes both received and placed calls.
+    /// </summary>
+    private static ConcurrentDictionary<string, SIPToOpenAiCall> _calls = new ConcurrentDictionary<string, SIPToOpenAiCall>();
+
+    static async Task Main()
+    {
+        Log.Logger = new LoggerConfiguration()
+            .MinimumLevel.Debug()
+            //.MinimumLevel.Verbose()
+            .Enrich.FromLogContext()
+            .WriteTo.Console()
+            .CreateLogger();
+
+        var loggerFactory = new SerilogLoggerFactory(Log.Logger);
+        SIPSorcery.LogFactory.Set(loggerFactory);
+
+        Log.Logger.Information("SIP-to-WebRTC OpenAI Demo Program");
+
+        var openAiKey = Environment.GetEnvironmentVariable("OPENAI_API_KEY");
+
+        if (string.IsNullOrWhiteSpace(openAiKey))
+        {
+            Log.Logger.Error("Please provide your OpenAI key as an environment variable. For example: set OPENAI_API_KEY=<your openai api key>");
+            return;
+        }
+
+        var logger = loggerFactory.CreateLogger<Program>();
+
+        SIPSorcery.LogFactory.Set(loggerFactory);
+        var sipTransport = new SIPTransport();
+        sipTransport.EnableTraceLogs();
+        sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
+        sipTransport.SIPTransportRequestReceived += (lep, rep, req) => OnRequest(lep, rep, req, sipTransport, openAiKey);
+
+        Console.WriteLine("Wait for ctrl-c to indicate user exit.");
+
+        var exitTcs = new TaskCompletionSource<object?>();
+        Console.CancelKeyPress += (s, e) =>
+        {
+            e.Cancel = true;
+            exitTcs.TrySetResult(null);
+        };
+
+        await exitTcs.Task;
+    }
+
+    /// <summary>
+    /// Because this is a server user agent the SIP transport must start listening for client user agents.
+    /// </summary>
+    private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest, SIPTransport sipTransport, string openAiKey)
+    {
+        try
+        {
+            if (sipRequest.Header.From != null &&
+                sipRequest.Header.From.FromTag != null &&
+                sipRequest.Header.To != null &&
+                sipRequest.Header.To.ToTag != null)
+            {
+                // This is an in-dialog request that will be handled directly by a user agent instance.
+            }
+            else if (sipRequest.Method == SIPMethodsEnum.INVITE)
+            {
+                Log.Information($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
+
+                SIPUserAgent ua = new SIPUserAgent(sipTransport, null);
+                ua.OnCallHungup += OnHangup;
+                ua.ServerCallCancelled += (uas, cancelReq) => Log.Debug("Incoming call cancelled by remote party.");
+                ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
+                ua.OnRtpEvent += (evt, hdr) => Log.Debug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
+                //ua.OnTransactionTraceMessage += (tx, msg) => Log.LogDebug($"uas tx {tx.TransactionId}: {msg}");
+                ua.ServerCallRingTimeout += (uas) =>
+                {
+                    Log.Warning($"Incoming call timed out in {uas.ClientTransaction.TransactionState} state waiting for client ACK, terminating.");
+                    ua.Hangup();
+                };
+
+                //bool wasMangled = false;
+                //sipRequest.Body = SIPPacketMangler.MangleSDP(sipRequest.Body, remoteEndPoint.Address.ToString(), out wasMangled);
+                //Log.LogDebug("INVITE was mangled=" + wasMangled + " remote=" + remoteEndPoint.Address.ToString() + ".");
+                //sipRequest.Header.ContentLength = sipRequest.Body.Length;
+
+                var uas = ua.AcceptCall(sipRequest);
+                var rtpSession = CreateRtpSession(ua);
+
+                // Insert a brief delay to allow testing of the "Ringing" progress response.
+                // Without the delay the call gets answered before it can be sent.
+                //await Task.Delay(500);
+
+                //if (!string.IsNullOrWhiteSpace(_publicIPAddress))
+                //{
+                //    await ua.Answer(uas, rtpSession, IPAddress.Parse(_publicIPAddress));
+                //}
+                //else
+                //{
+                await ua.Answer(uas, rtpSession);
+                //}
+
+                if (ua.IsCallActive)
+                {
+                    await rtpSession.Start();
+                    _calls.TryAdd(ua.Dialogue.CallId, new SIPToOpenAiCall(ua, rtpSession, null));
+
+                    Log.Information($"Call answered, call ID {ua.Dialogue.CallId}.");
+
+                    // Create a WebRTC session to OpenAI.
+                    await CreateOpenAIWebRTCSession(new SerilogLoggerFactory(Log.Logger), openAiKey, ua.Dialogue.CallId, rtpSession);
+                }
+            }
+            else if (sipRequest.Method == SIPMethodsEnum.BYE)
+            {
+                SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
+                await sipTransport.SendResponseAsync(byeResponse);
+            }
+            else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
+            {
+                SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
+                await sipTransport.SendResponseAsync(notAllowededResponse);
+            }
+            else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
+            {
+                SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
+                await sipTransport.SendResponseAsync(optionsResponse);
+            }
+        }
+        catch (Exception reqExcp)
+        {
+            Log.Warning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
+        }
+    }
+
+    /// <summary>
+    /// Example of how to create a basic RTP session object and hook up the event handlers.
+    /// </summary>
+    /// <param name="ua">The user agent the RTP session is being created for.</param>
+    /// <returns>A new RTP session object.</returns>
+    private static RTPSession CreateRtpSession(SIPUserAgent ua)
+    {
+        var rtpSession = new RTPSession(false, false, false);
+        rtpSession.addTrack(new MediaStreamTrack(AudioCommonlyUsedFormats.OpusWebRTC));
+        rtpSession.AcceptRtpFromAny = true;
+
+        // Wire up the event handler for RTP packets received from the remote party.
+        //rtpSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, ep, type, rtp);
+        rtpSession.OnTimeout += (mediaType) =>
+        {
+            if (ua?.Dialogue != null)
+            {
+                Log.Warning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
+            }
+            else
+            {
+                Log.Warning($"RTP timeout on incomplete call, closing RTP session.");
+            }
+
+            ua?.Hangup();
+        };
+
+        return rtpSession;
+    }
+
+    private static async Task CreateOpenAIWebRTCSession(ILoggerFactory loggerFactory, string openAiKey, string sipCallID, RTPSession rtpSession)
+    {
+        var logger = loggerFactory.CreateLogger<WebRTCEndPoint>();
+        var webrtcEndPoint = new WebRTCEndPoint(openAiKey, logger);
+
+        if (_calls.TryGetValue(sipCallID, out var existing))
+        {
+            var updated = existing with { webrtc = webrtcEndPoint };
+            _calls.TryUpdate(sipCallID, updated, existing);
+        }
+
+        var negotiateConnectResult = await webrtcEndPoint.StartConnect();
+
+        if (negotiateConnectResult.IsLeft)
+        {
+            Log.Logger.Error($"Failed to negotiation connection to OpenAI Realtime WebRTC endpoint: {negotiateConnectResult.LeftAsEnumerable().First()}");
+            return;
+        }
+
+        webrtcEndPoint.OnPeerConnectionConnected += () =>
+        {
+            Log.Logger.Information("WebRTC peer connection established.");
+
+            webrtcEndPoint.ConnectRTPSession(rtpSession, AudioCommonlyUsedFormats.OpusWebRTC);
+
+            var voice = RealtimeVoicesEnum.shimmer;
+
+            // Optionally send a session update message to adjust the session parameters.
+            var sessionUpdateResult = webrtcEndPoint.DataChannelMessenger.SendSessionUpdate(
+                voice,
+                "Keep it short.",
+                transcriptionModel: TranscriptionModelEnum.Whisper1);
+
+            if (sessionUpdateResult.IsLeft)
+            {
+                Log.Logger.Error($"Failed to send session update message: {sessionUpdateResult.LeftAsEnumerable().First()}");
+            }
+
+            // Trigger the conversation by sending a response create message.
+            var result = webrtcEndPoint.DataChannelMessenger.SendResponseCreate(voice, "Say Hi!");
+            if (result.IsLeft)
+            {
+                Log.Logger.Error($"Failed to send response create message: {result.LeftAsEnumerable().First()}");
+            }
+        };
+
+        webrtcEndPoint.OnDataChannelMessage += (dc, message) =>
+        {
+            var log = message switch
+            {
+                RealtimeServerEventSessionUpdated sessionUpdated => $"Session updated: {sessionUpdated.ToJson()}",
+                //RealtimeServerEventConversationItemInputAudioTranscriptionDelta inputDelta => $"ME ⌛: {inputDelta.Delta?.Trim()}",
+                RealtimeServerEventConversationItemInputAudioTranscriptionCompleted inputTranscript => $"ME ✅: {inputTranscript.Transcript?.Trim()}",
+                //RealtimeServerEventResponseAudioTranscriptDelta responseDelta => $"AI ⌛: {responseDelta.Delta?.Trim()}",
+                RealtimeServerEventResponseAudioTranscriptDone responseTranscript => $"AI ✅: {responseTranscript.Transcript?.Trim()}",
+                //_ => $"Received {message.Type} -> {message.GetType().Name}"
+                _ => string.Empty
+            };
+
+            if (log != string.Empty)
+            {
+                Log.Information(log);
+            }
+        };
+    }
+
+    /// <summary>
+    /// Event handler for receiving RTP packets.
+    /// </summary>
+    /// <param name="ua">The SIP user agent associated with the RTP session.</param>
+    /// <param name="type">The media type of the RTP packet (audio or video).</param>
+    /// <param name="rtpPacket">The RTP packet received from the remote party.</param>
+    private static void OnRtpPacketReceived(SIPUserAgent ua, IPEndPoint remoteEp, SDPMediaTypesEnum type, RTPPacket rtpPacket)
+    {
+        // The raw audio data is available in rtpPacket.Payload.
+        Log.Verbose($"OnRtpPacketReceived from {remoteEp}.");
+    }
+
+    /// <summary>
+    /// Event handler for receiving a DTMF tone.
+    /// </summary>
+    /// <param name="ua">The user agent that received the DTMF tone.</param>
+    /// <param name="key">The DTMF tone.</param>
+    /// <param name="duration">The duration in milliseconds of the tone.</param>
+    private static void OnDtmfTone(SIPUserAgent ua, byte key, int duration)
+    {
+        string callID = ua.Dialogue.CallId;
+        Log.Information($"Call {callID} received DTMF tone {key}, duration {duration}ms.");
+    }
+
+    /// <summary>
+    /// Remove call from the active calls list.
+    /// </summary>
+    /// <param name="dialogue">The dialogue that was hungup.</param>
+    private static void OnHangup(SIPDialogue dialogue)
+    {
+        if (dialogue != null)
+        {
+            string callID = dialogue.CallId;
+            if (_calls.ContainsKey(callID))
+            {
+                if (_calls.TryRemove(callID, out var call))
+                {
+                    Log.Information($"Call {callID} removed.");
+
+                    // This app only uses each SIP user agent once so here the agent is 
+                    // explicitly closed to prevent is responding to any new SIP requests.
+                    call.ua.Close();
+                    call.webrtc?.Close();
+                }
+            }
+        }
+    }
+}