Added RTP session to user agent. Messed around with the softphone UI.

Author: sipsorcery

.editorconfig

@@ -20,7 +20,7 @@ dotnet_separate_import_directive_groups = false
 csharp_space_after_keywords_in_control_flow_statements = true
 
 # Wrapping preferences
-csharp_preserve_single_line_statements = false
+csharp_preserve_single_line_statements = true
 csharp_preserve_single_line_blocks = true
 
 # Don't allow single line if/else blocks without braces.

examples/CallHoldAndTransfer/Program.cs

@@ -58,10 +58,10 @@ static void Main()
             var sipTransport = new SIPTransport();
             sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
 
-            EnableTraceLogs(sipTransport);
+            //EnableTraceLogs(sipTransport);
 
-            // Initialise an RTP session to receive the RTP packets from the remote SIP server.
-            var rtpSession = new RTPSession((int)SDPMediaFormatsEnum.PCMU, null, null, true);
+            // Get the default speaker.
+            var (audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
 
             // Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
             var userAgent = new SIPUserAgent(sipTransport, null);
@@ -79,17 +79,13 @@ static void Main()
                 if (resp.Status == SIPResponseStatusCodesEnum.Ok)
                 {
                     Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
-
-                    // Only set the remote RTP end point if there hasn't already been a packet received on it.
-                    if (rtpSession.DestinationEndPoint == null)
-                    {
-                        rtpSession.DestinationEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
-                        Log.LogDebug($"Remote RTP socket {rtpSession.DestinationEndPoint}.");
-                    }
+                    PlayRemoteMedia(userAgent.RtpSession, audioOutProvider);
                 }
                 else
                 {
                     Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
+                    hasCallFailed = true;
+                    exitCts.Cancel();
                 }
             };
             userAgent.OnCallHungup += () =>
@@ -100,19 +96,6 @@ static void Main()
             userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
             userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
             userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
-            userAgent.OnReinviteRequest += (uasInviteTx) =>
-            {
-                Log.LogDebug("Reinvite request received.");
-
-                // Re-INVITEs can also be changing the RTP end point. We can update this each time.
-                IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(uasInviteTx.TransactionRequest.Body);
-                if(rtpSession.DestinationEndPoint != dstRtpEndPoint)
-                {
-                    Log.LogDebug($"Remote call party RTP end point changed from {rtpSession.DestinationEndPoint} to {dstRtpEndPoint}.");
-                    rtpSession.DestinationEndPoint = dstRtpEndPoint;
-                }
-                uasInviteTx.SendFinalResponse(uasInviteTx.GetOkResponse(SDP.SDP_MIME_CONTENTTYPE, userAgent.Dialogue.SDP));
-            };
 
             sipTransport.SIPTransportRequestReceived += (locelEndPoint, remoteEndPoint, sipRequest) =>
             {
@@ -137,12 +120,13 @@ static void Main()
                     {
                         Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
                         var incomingCall = userAgent.AcceptCall(sipRequest);
+                        userAgent.Answer(incomingCall);
 
-                        SDP remoteSDP = SDP.ParseSDPDescription(sipRequest.Body);
-                        IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(IPAddress.Parse(remoteSDP.Connection.ConnectionAddress));
-                        var sdpOffer = rtpSession.GetSDP(localIPAddress);
+                        PlayRemoteMedia(userAgent.RtpSession, audioOutProvider);
 
-                        userAgent.Answer(incomingCall, sdpOffer);
+                        string caller = sipRequest.Header.From.FromURI.ToString();
+                        caller = (string.IsNullOrEmpty(sipRequest.Header.From.FromName)) ? caller : sipRequest.Header.From.FromName + " " + caller;
+                        Log.LogInformation($"Answered incoming call from {caller} at {remoteEndPoint}.");
                     }
                 }
                 else
@@ -153,18 +137,6 @@ static void Main()
                 }
             };
 
-            // Wire up the RTP receive session to the default speaker.
-            var (audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
-            rtpSession.OnReceivedSampleReady += (sample) =>
-            {
-                for (int index = 0; index < sample.Length; index++)
-                {
-                    short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]);
-                    byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
-                    audioOutProvider.AddSamples(pcmSample, 0, 2);
-                }
-            };
-
             // Wire up the RTP send session to the audio output device.
             WaveInEvent waveInEvent = GetAudioInputDevice();
             uint rtpSendTimestamp = 0;
@@ -179,9 +151,9 @@ static void Main()
                     sample[sampleIndex++] = ulawByte;
                 }
 
-                if (rtpSession.DestinationEndPoint != null)
+                if (userAgent?.RtpSession?.DestinationEndPoint != null)
                 {
-                    rtpSession.SendAudioFrame(rtpSendTimestamp, sample);
+                    userAgent.RtpSession.SendAudioFrame(rtpSendTimestamp, sample);
                     rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds);
                 }
 
@@ -201,8 +173,7 @@ static void Main()
                         {
                             if (!userAgent.IsCallActive)
                             {
-                                SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
-                                var callDescriptor = GetCallDescriptor(callUri, rtpSession);
+                                var callDescriptor = GetCallDescriptor(DEFAULT_DESTINATION_SIP_URI);
                                 userAgent.Call(callDescriptor);
                             }
                             else
@@ -281,7 +252,7 @@ static void Main()
 
             Log.LogInformation("Exiting...");
 
-            rtpSession?.Close();
+            userAgent?.RtpSession?.Close();
             waveInEvent?.StopRecording();
             audioOutEvent?.Stop();
 
@@ -320,32 +291,42 @@ static void Main()
         /// <param name="callUri">The URI to place the call to.</param>
         /// <param name="rtpSession">The RTP session that will be handling the RTP/RTCP packets for the call.</param>
         /// <returns>A call descriptor.</returns>
-        private static SIPCallDescriptor GetCallDescriptor(SIPURI callUri, RTPSession rtpSession)
+        private static SIPCallDescriptor GetCallDescriptor(string callUri)
         {
-            var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
-            Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
-            var dstAddress = lookupResult.GetSIPEndPoint().Address;
-
-            IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
-
-            var sdpOffer = rtpSession.GetSDP(localIPAddress);
-
-            // Start the thread that places the call.
+            // Create a call descriptor to place an outgoing call.
             SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
                 SIP_USERNAME,
                 SIP_PASSWORD,
-                callUri.ToString(),
+                callUri,
                 $"sip:{SIP_USERNAME}@localhost",
-                callUri.CanonicalAddress,
+                callUri,
                 null, null, null,
                 SIPCallDirection.Out,
                 SDP.SDP_MIME_CONTENTTYPE,
-                sdpOffer.ToString(),
+                null,
                 null);
 
             return callDescriptor;
         }
 
+        /// <summary>
+        /// Wires up the active RTP session to the speaker.
+        /// </summary>
+        /// <param name="rtpSession">The active RTP session receiving the remote party's RTP packets.</param>
+        /// <param name="audioOutProvider">The audio buffer for the default system audio output device.</param>
+        private static void PlayRemoteMedia(RTPSession rtpSession, BufferedWaveProvider audioOutProvider)
+        {
+            rtpSession.OnReceivedSampleReady += (sample) =>
+            {
+                for (int index = 0; index < sample.Length; index++)
+                {
+                    short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]);
+                    byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
+                    audioOutProvider.AddSamples(pcmSample, 0, 2);
+                }
+            };
+        }
+
         /// <summary>
         /// Get the audio output device, e.g. speaker.
         /// Note that NAudio.Wave.WaveOut is not available for .Net Standard so no easy way to check if 

examples/Softphone/SIPSorcery.SoftPhone/Media/AudioChannel.cs

@@ -2,7 +2,7 @@
 // Filename: AudioChannel.cs
 //
 // Description: This class manages the coding and decoding of audio from physical
-// devices into and for RTP packets. 
+// devices and playback from samples recevied from RTP packets. 
 //
 // Author(s):
 // Aaron Clauson ([email protected])
@@ -26,6 +26,8 @@ namespace SIPSorcery.SoftPhone
 {
     public class AudioChannel
     {
+        public const int AUDIO_INPUT_BUFFER_MILLISECONDS = 80;
+
         private ILog logger = AppState.logger;
 
         private BufferedWaveProvider m_waveProvider;
@@ -57,7 +59,7 @@ public AudioChannel()
             else
             {
                 m_waveInEvent = new WaveInEvent();
-                m_waveInEvent.BufferMilliseconds = 20;
+                m_waveInEvent.BufferMilliseconds = AUDIO_INPUT_BUFFER_MILLISECONDS;
                 m_waveInEvent.NumberOfBuffers = 1;
                 m_waveInEvent.DeviceNumber = 0;
                 m_waveInEvent.DataAvailable += AudioSampleAvailable;