Null reference bug fix in SDP packet mangler. (sipsorcery-org#1417)

Author: sipsorcery

examples/SIPExamples/SIPCallServer/Dockerfile

@@ -0,0 +1,27 @@
+# Stage 1: build
+FROM mcr.microsoft.com/dotnet/sdk:8.0 AS build
+WORKDIR /src
+
+# Copy csproj and restore as distinct layers for caching
+COPY *.csproj ./
+COPY local-nuget/ ./local-nuget/
+
+RUN dotnet restore --source ./local-nuget --source https://api.nuget.org/v3/index.json 
+
+# Copy rest of source and publish
+COPY . .
+RUN dotnet publish -c Release -o /app/publish /p:UseAppHost=false
+
+# Stage 2: runtime
+FROM mcr.microsoft.com/dotnet/runtime:8.0 AS runtime
+WORKDIR /app
+
+# Copy published app
+COPY --from=build /app/publish ./
+
+# Expose SIP ports
+EXPOSE 5060/udp
+EXPOSE 5061/tcp
+
+# Run the server
+ENTRYPOINT ["dotnet", "SIPCallServer.dll"]

examples/SIPExamples/SIPCallServer/Program.cs

@@ -64,6 +64,10 @@ public SendSilenceJob(Timer timer, SIPUserAgent ua)
 
     class Program
     {
+        private const string PUBLIC_IP_ADDRESS_ENV_VAR = "PUBLIC_IP_ADDRESS";
+        private const string RTP_PORT_ENV_VAR = "RTP_PORT";
+        private const string IS_HEADLESS_ENV_VAR = "IS_HEADLESS";
+
         //private static string DEFAULT_CALL_DESTINATION = "sip:*61@192.168.0.48";
         private static string DEFAULT_CALL_DESTINATION = "sip:aaron@127.0.0.1:7060;transport=tcp";
         private static string DEFAULT_TRANSFER_DESTINATION = "sip:*61@192.168.0.48";
@@ -93,38 +97,85 @@ class Program
         /// </summary>
         private static ConcurrentDictionary<string, SIPRegistrationUserAgent> _registrations = new ConcurrentDictionary<string, SIPRegistrationUserAgent>();
 
+        private static string _publicIPAddress = null;
+        private static int _rtpPort = 0;
+
+        /// <summary>
+        /// If running on K8S or similar there is no console window to receive key presses.
+        /// </summary>
+        private static bool _isHeadless = false;
+
         static async Task Main()
         {
             Console.WriteLine("SIPSorcery SIP Call Server example.");
-            Console.WriteLine("Press 'c' to place a call to the default destination.");
-            Console.WriteLine("Press 'd' to send a random DTMF tone to the newest call.");
-            Console.WriteLine("Press 'h' to hangup the oldest call.");
-            Console.WriteLine("Press 'H' to hangup all calls.");
-            Console.WriteLine("Press 'l' to list current calls.");
-            Console.WriteLine("Press 'r' to list current registrations.");
-            Console.WriteLine("Press 't' to transfer the newest call to the default destination.");
-            Console.WriteLine("Press 'q' to quit.");
+
+            if (!_isHeadless)
+            {
+                Console.WriteLine("Press 'c' to place a call to the default destination.");
+                Console.WriteLine("Press 'd' to send a random DTMF tone to the newest call.");
+                Console.WriteLine("Press 'h' to hangup the oldest call.");
+                Console.WriteLine("Press 'H' to hangup all calls.");
+                Console.WriteLine("Press 'l' to list current calls.");
+                Console.WriteLine("Press 'r' to list current registrations.");
+                Console.WriteLine("Press 't' to transfer the newest call to the default destination.");
+                Console.WriteLine("Press 'q' to quit.");
+            }
 
             Log = AddConsoleLogger();
 
             // Set up a default SIP transport.
             _sipTransport = new SIPTransport();
+
+            if (!string.IsNullOrWhiteSpace(Environment.GetEnvironmentVariable(PUBLIC_IP_ADDRESS_ENV_VAR)))
+            {
+                _publicIPAddress = Environment.GetEnvironmentVariable(PUBLIC_IP_ADDRESS_ENV_VAR);
+                _sipTransport.ContactHost = _publicIPAddress;
+            }
+
+            if (!string.IsNullOrWhiteSpace(Environment.GetEnvironmentVariable(RTP_PORT_ENV_VAR)))
+            {
+                int.TryParse(Environment.GetEnvironmentVariable(RTP_PORT_ENV_VAR), out _rtpPort);
+            }
+
+            if (!string.IsNullOrWhiteSpace(Environment.GetEnvironmentVariable(IS_HEADLESS_ENV_VAR)))
+            {
+                bool.TryParse(Environment.GetEnvironmentVariable(IS_HEADLESS_ENV_VAR), out _isHeadless);
+            }
+
             _sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
             _sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
             _sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
             var localhostCertificate = new X509Certificate2(SIPS_CERTIFICATE_PATH);
             _sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(IPAddress.Any, SIPS_LISTEN_PORT)));
             // If it's desired to listen on a single IP address use the equivalent of:
             //_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Parse("192.168.11.50"), SIP_LISTEN_PORT)));
-            _sipTransport.EnableTraceLogs();
+            //_sipTransport.EnableTraceLogs();
 
             _sipTransport.SIPTransportRequestReceived += OnRequest;
 
             // Uncomment to enable registrations.
             //StartRegistrations(_sipTransport, _sipAccounts);
 
-            CancellationTokenSource exitCts = new CancellationTokenSource();
-            await Task.Run(() => OnKeyPress(exitCts.Token));
+            if (!_isHeadless)
+            {
+                CancellationTokenSource exitCts = new CancellationTokenSource();
+                await Task.Run(() => OnKeyPress(exitCts.Token));
+
+                Console.WriteLine("Press ctrl-c to exit.");
+            }
+            else
+            {
+                // Ctrl-c will gracefully exit the call at any point.
+                ManualResetEvent exitMre = new ManualResetEvent(false);
+                Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
+                {
+                    e.Cancel = true;
+                    exitMre.Set();
+                };
+
+                // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
+                exitMre.WaitOne();
+            }
 
             Log.LogInformation("Exiting...");
 
@@ -159,7 +210,7 @@ private static async Task OnKeyPress(CancellationToken exit)
                         ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
                         ua.OnCallHungup += OnHangup;
 
-                        var rtpSession = CreateRtpSession(ua, null);
+                        var rtpSession = CreateRtpSession(ua, null, _rtpPort);
                         var callResult = await ua.Call(DEFAULT_CALL_DESTINATION, null, null, rtpSession);
 
                         if (callResult)
@@ -297,7 +348,7 @@ private static async Task OnKeyPress(CancellationToken exit)
         /// <param name="dst">THe destination specified on an incoming call. Can be used to
         /// set the audio source.</param>
         /// <returns>A new RTP session object.</returns>
-        private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
+        private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst, int bindPort)
         {
             List<AudioCodecsEnum> codecs = new List<AudioCodecsEnum> { AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA, AudioCodecsEnum.G722 };
 
@@ -310,12 +361,12 @@ private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
             Log.LogInformation($"RTP audio session source set to {audioSource}.");
 
             AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = audioSource });
-            audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec) );
-            var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = audioExtrasSource });
+            audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec));
+            var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = audioExtrasSource }, bindPort: bindPort);
             rtpAudioSession.AcceptRtpFromAny = true;
 
             // Wire up the event handler for RTP packets received from the remote party.
-            rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
+            rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, ep, type, rtp);
             rtpAudioSession.OnTimeout += (mediaType) =>
             {
                 if (ua?.Dialogue != null)
@@ -339,9 +390,10 @@ private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
         /// <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, SDPMediaTypesEnum type, RTPPacket rtpPacket)
+        private static void OnRtpPacketReceived(SIPUserAgent ua, IPEndPoint remoteEp, SDPMediaTypesEnum type, RTPPacket rtpPacket)
         {
             // The raw audio data is available in rtpPacket.Payload.
+            //Log.LogTrace($"OnRtpPacketReceived from {remoteEp}.");
         }
 
         /// <summary>
@@ -386,14 +438,26 @@ private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint re
                         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, sipRequest.URI.User);
+                    var rtpSession = CreateRtpSession(ua, sipRequest.URI.User, _rtpPort);
 
                     // 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);
 
-                    await ua.Answer(uas, rtpSession);
+                    if (!string.IsNullOrWhiteSpace(_publicIPAddress))
+                    {
+                        await ua.Answer(uas, rtpSession, IPAddress.Parse(_publicIPAddress));
+                    }
+                    else
+                    {
+                        await ua.Answer(uas, rtpSession);
+                    }
 
                     if (ua.IsCallActive)
                     {
@@ -434,7 +498,7 @@ private static void OnHangup(SIPDialogue dialogue)
                 string callID = dialogue.CallId;
                 if (_calls.ContainsKey(callID))
                 {
-                    if(_calls.TryRemove(callID, out var ua))
+                    if (_calls.TryRemove(callID, out var ua))
                     {
                         // 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.
@@ -475,7 +539,7 @@ private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
         {
             var serilogLogger = new LoggerConfiguration()
                 .Enrich.FromLogContext()
-                .MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
+                .MinimumLevel.Is(Serilog.Events.LogEventLevel.Verbose)
                 .WriteTo.Console()
                 .CreateLogger();
             var factory = new SerilogLoggerFactory(serilogLogger);

examples/SIPExamples/SIPCallServer/SIPCallServer.csproj

@@ -9,7 +9,6 @@
     <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" />
-    <PackageReference Include="System.Net.Security" Version="4.3.2" />
   </ItemGroup>
 
     <ItemGroup>

examples/SIPExamples/SIPCallServer/readme-docker.txt

@@ -0,0 +1,4 @@
+docker build -t sipsorcery/sipcallserver:0.5 .
+docker push sipsorcery/sipcallserver:0.5
+
+docker run -it --rm --name sip-callserver -p 5060:5060/udp -p 5061:5061/tcp sipsorcery/sipcallserver:0.2

src/app/SIPPacketMangler.cs

@@ -38,9 +38,16 @@ public static string MangleSDP(string sdpBody, string publicIPAddress, out bool
             {
                 if (sdpBody != null && publicIPAddress != null)
                 {
-                    IPAddress addr = SDP.GetSDPRTPEndPoint(sdpBody).Address;
+                    var sdpEndPoint = SDP.GetSDPRTPEndPoint(sdpBody);
+                    if(sdpEndPoint == null)
+                    {
+                        logger.LogWarning("SDP mangling failed to find a valid RTP endpoint in the SDP body.");
+                        return sdpBody;
+                    }
+
                     //rj2: need to consider publicAddress and IPv6 for mangling
                     IPAddress pubaddr = IPAddress.Parse(publicIPAddress);
+                    var addr = sdpEndPoint.Address;
                     string sdpAddress = addr.ToString();
 
                     // Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.

src/net/SDP/SDP.cs

@@ -901,6 +901,10 @@ public IPEndPoint GetSDPRTPEndPoint()
             {
                 return new IPEndPoint(IPAddress.Parse(sessionConnection.ConnectionAddress), firstMediaOffer.Port);
             }
+            else if(firstMediaOffer != null && firstMediaOffer.Connection != null)
+            {
+                return new IPEndPoint(IPAddress.Parse(firstMediaOffer.Connection.ConnectionAddress), firstMediaOffer.Port);
+            }
             else
             {
                 return null;