From 10effcb43fd2272b4c134bcfe4e8c8398f372368 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Standa=20Luke=C5=A1?= Date: Wed, 24 Apr 2024 19:04:57 +0000 Subject: [PATCH] Enable generation of SIPSorcery.xml documentation file (#1106) * Enabled GenerateDocumentationFile in SIPSorcery.csproj * Disabled "missing XML comment" warnings * Fixed most malformed XML comments (warnings about these are not disabled) --- src/SIPSorcery.csproj | 3 + src/app/Media/Codecs/AudioEncoder.cs | 3 +- src/app/Media/Codecs/G729Codec/CodLd8k.cs | 6 +- src/app/Media/Codecs/G729Codec/CorFunc.cs | 4 +- src/app/Media/Codecs/G729Codec/Gainpred.cs | 6 +- src/app/Media/Codecs/G729Codec/Ld8k.cs | 6 +- src/app/Media/Codecs/G729Codec/Pitch.cs | 5 +- src/app/Media/Codecs/G729Codec/Postfil.cs | 4 +- src/app/Media/IMediaSession.cs | 4 +- src/app/Media/Sources/AudioSignalGenerator.cs | 2 +- src/app/SIPUserAgents/SIPCallDescriptor.cs | 2 +- src/app/SIPUserAgents/SIPUserAgent.cs | 4 +- .../DtlsSrtp/Transform/SrtcpTransformer.cs | 150 +++++++++--------- src/net/DtlsSrtp/Transform/SrtpCipherCTR.cs | 78 ++++----- src/net/DtlsSrtp/Transform/SrtpCipherF8.cs | 100 ++++++------ src/net/ICE/RTCIceCandidate.cs | 4 +- src/net/ICE/RtpIceChannel.cs | 3 +- src/net/RTCP/RTCPSession.cs | 2 +- src/net/RTP/MediaStream.cs | 15 +- src/net/RTP/MediaStreamTrack.cs | 6 +- src/net/RTP/Packetisation/H264Depacketiser.cs | 14 +- src/net/RTP/RTPSession.cs | 3 +- src/net/RTP/VideoStream.cs | 9 +- src/net/RTSP/Mjpeg.cs | 2 +- src/net/SCTP/Chunks/SctpDataChunk.cs | 2 +- src/net/SCTP/SctpAssociation.cs | 2 +- src/net/SCTP/SctpDataReceiver.cs | 4 + src/net/SCTP/SctpDataSender.cs | 4 +- src/net/SCTP/SctpPacket.cs | 2 +- src/net/SCTP/SctpTransport.cs | 4 +- src/net/SDP/SDPAudioVideoMediaFormat.cs | 8 +- src/net/WebRTC/IRTCPeerConnection.cs | 1 + src/net/WebRTC/RTCPeerConnection.cs | 12 +- src/net/WebRTC/RTCPeerSctpAssociation.cs | 2 - src/net/WebRTC/RTCSctpTransport.cs | 2 - 35 files changed, 234 insertions(+), 244 deletions(-) diff --git a/src/SIPSorcery.csproj b/src/SIPSorcery.csproj index 6623c6356..ccb8b7241 100755 --- a/src/SIPSorcery.csproj +++ b/src/SIPSorcery.csproj @@ -28,6 +28,9 @@ netstandard2.0;netstandard2.1;netcoreapp3.1;net461;net5.0;net6.0;net8.0 true $(NoWarn);SYSLIB0050 + true + + $(NoWarn);CS1591;CS1573;CS1587 Aaron Clauson, Christophe Irles, Rafael Soares & Contributors Copyright © 2010-2024 Aaron Clauson BSD-3-Clause diff --git a/src/app/Media/Codecs/AudioEncoder.cs b/src/app/Media/Codecs/AudioEncoder.cs index 78e1730f6..363e341d9 100644 --- a/src/app/Media/Codecs/AudioEncoder.cs +++ b/src/app/Media/Codecs/AudioEncoder.cs @@ -123,9 +123,8 @@ public byte[] EncodeAudio(short[] pcm, AudioFormat format) /// /// Event handler for receiving RTP packets from the remote party. /// - /// The remote end point the RTP was received from. + /// Data received from an RTP socket. /// The audio format of the encoded packets. - /// The RTP packet with the media sample. public short[] DecodeAudio(byte[] encodedSample, AudioFormat format) { if (format.Codec == AudioCodecsEnum.G722) diff --git a/src/app/Media/Codecs/G729Codec/CodLd8k.cs b/src/app/Media/Codecs/G729Codec/CodLd8k.cs index d9130cb67..f6be7031e 100644 --- a/src/app/Media/Codecs/G729Codec/CodLd8k.cs +++ b/src/app/Media/Codecs/G729Codec/CodLd8k.cs @@ -155,7 +155,7 @@ Sherbrooke. All rights reserved. /** * Initialization of variables for the encoder. * Initialize pointers to speech vector. - *
+ *
 
+ * ]]>

  */
 
         public void init_coder_ld8k()
@@ -543,4 +543,4 @@ int[] ana
             Util.copy(old_exc, L_FRAME, old_exc, PIT_MAX + L_INTERPOL);
         }
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/Codecs/G729Codec/CorFunc.cs b/src/app/Media/Codecs/G729Codec/CorFunc.cs
index 41da1d466..2d5466af3 100644
--- a/src/app/Media/Codecs/G729Codec/CorFunc.cs
+++ b/src/app/Media/Codecs/G729Codec/CorFunc.cs
@@ -59,7 +59,7 @@ G.729 main body and G.729A
  * @param xn        input : target vector x[0:l_subfr]
  * @param y1        input : filtered adaptive codebook vector
  * @param y2        input : filtered 1st codebook innovation
- * @param g_coeff   output:  , -2 , and 2
+ * @param g_coeff    , -2 , and 2 ]]>
  */
 
         public static void corr_xy2(
@@ -115,4 +115,4 @@ float[] d
             }
         }
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/Codecs/G729Codec/Gainpred.cs b/src/app/Media/Codecs/G729Codec/Gainpred.cs
index 82fc4aa67..871c06252 100644
--- a/src/app/Media/Codecs/G729Codec/Gainpred.cs
+++ b/src/app/Media/Codecs/G729Codec/Gainpred.cs
@@ -118,13 +118,13 @@ float g_code
 
         /**
  * Update table of past quantized energies (frame erasure).
- * 
+ * 
 
+ * ]]>

  *
  * @param past_qua_en   input/output:Past quantized energies
  */
@@ -147,4 +147,4 @@ float[] past_qua_en
             past_qua_en[0] = av_pred_en;
         }
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/Codecs/G729Codec/Ld8k.cs b/src/app/Media/Codecs/G729Codec/Ld8k.cs
index 6a10faadd..34f6aa981 100644
--- a/src/app/Media/Codecs/G729Codec/Ld8k.cs
+++ b/src/app/Media/Codecs/G729Codec/Ld8k.cs
@@ -124,12 +124,12 @@ public class Ld8k
         public static float GAMMA2_PST = 0.55f;
 
         /**
- * tilt weighting factor when k1<0
+ * tilt weighting factor when k1 < 0
  */
         public static float GAMMA3_MINUS = 0.9f;
 
         /**
- * tilt weighting factor when k1>0
+ * tilt weighting factor when k1 > 0
  */
         public static float GAMMA3_PLUS = 0.2f;
 
@@ -479,4 +479,4 @@ Sherbrooke.  All rights reserved.
  * resolution of fractional delays
  */
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/Codecs/G729Codec/Pitch.cs b/src/app/Media/Codecs/G729Codec/Pitch.cs
index 576cc5c00..e077fa84d 100644
--- a/src/app/Media/Codecs/G729Codec/Pitch.cs
+++ b/src/app/Media/Codecs/G729Codec/Pitch.cs
@@ -351,7 +351,7 @@ int corr_norm_offset
             }
         }
 
-        /**
+        /**  , -2
  *
  * @param xn        input : target vector
@@ -359,6 +359,7 @@ int corr_norm_offset
  * @param g_coeff   output:  and -2
  * @param l_subfr   input : vector dimension
  * @return          pitch gain
+ * ]]>
  */
 
         public static float g_pitch(
@@ -528,4 +529,4 @@ float x
             return 1.0f / (float)Math.Sqrt(x);
         }
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/Codecs/G729Codec/Postfil.cs b/src/app/Media/Codecs/G729Codec/Postfil.cs
index 9a441997c..bd730c858 100644
--- a/src/app/Media/Codecs/G729Codec/Postfil.cs
+++ b/src/app/Media/Codecs/G729Codec/Postfil.cs
@@ -635,7 +635,7 @@ float gain_plt
 
         /**
  * Compute delayed signal,
- * num & den of gain for fractional delay
+ * num & den of gain for fractional delay
  * with long interpolation filter
  *
  * @param s_in           input signal with past
@@ -906,4 +906,4 @@ float gain_prec
             return gain_prec;
         }
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/IMediaSession.cs b/src/app/Media/IMediaSession.cs
index ba668646a..31012b358 100644
--- a/src/app/Media/IMediaSession.cs
+++ b/src/app/Media/IMediaSession.cs
@@ -102,7 +102,7 @@ public interface IMediaSession
         /// if they are.
         /// 
         /// Whether the SDP being set is an offer or answer.
-        /// The SDP description from the remote party.
+        /// The SDP description from the remote party.
         /// If successful an OK enum result. If not an enum result indicating the 
         /// failure cause.
         SetDescriptionResultEnum SetRemoteDescription(SdpType sdpType, SDP sessionDescription);
@@ -150,4 +150,4 @@ public interface IMediaSession
         /// Optional. A descriptive reason for closing the session.
         void Close(string reason);
     }
-}
\ No newline at end of file
+}
diff --git a/src/app/Media/Sources/AudioSignalGenerator.cs b/src/app/Media/Sources/AudioSignalGenerator.cs
index 90eaf8c15..a74614c14 100644
--- a/src/app/Media/Sources/AudioSignalGenerator.cs
+++ b/src/app/Media/Sources/AudioSignalGenerator.cs
@@ -67,7 +67,7 @@ public SignalGenerator()
         /// Initializes a new instance for the Generator (UserDef SampleRate & Channels)
         /// 
         /// Desired sample rate
-        /// Number of channels
+        /// Number of channels
         public SignalGenerator(int sampleRate, int channels)
         {
             phi = 0;
diff --git a/src/app/SIPUserAgents/SIPCallDescriptor.cs b/src/app/SIPUserAgents/SIPCallDescriptor.cs
index e8387f82e..1288ca7b5 100644
--- a/src/app/SIPUserAgents/SIPCallDescriptor.cs
+++ b/src/app/SIPUserAgents/SIPCallDescriptor.cs
@@ -265,7 +265,7 @@ public SIPFromHeader GetFromHeader()
         /// 
         /// 
         /// 
-        /// 
+        /// 
         public void SetGeneralFromHeaderFields(string fromDisplayName, string fromUsername, string fromHost)
         {
             if (!fromDisplayName.IsNullOrBlank() && FromDisplayName == null)
diff --git a/src/app/SIPUserAgents/SIPUserAgent.cs b/src/app/SIPUserAgents/SIPUserAgent.cs
index a2b6a987a..e47bbab1c 100644
--- a/src/app/SIPUserAgents/SIPUserAgent.cs
+++ b/src/app/SIPUserAgents/SIPUserAgent.cs
@@ -416,7 +416,7 @@ public SIPUserAgent(SIPTransport transport, SIPEndPoint outboundProxy, bool isTr
 
         /// 
         /// Attempts to place a new outgoing call AND waits for the call to be answered or fail.
-        /// Use  to start a call without
+        /// Use  to start a call without
         /// waiting for it to complete and monitor  and
         ///  to detect an answer or failure.
         /// 
@@ -464,7 +464,7 @@ public Task Call(string dst, string username, string password, IMediaSessi
 
         /// 
         /// Attempts to place a new outgoing call AND waits for the call to be answered or fail.
-        /// Use  to start a call without
+        /// Use  to start a call without
         /// waiting for it to complete and monitor  and
         ///  to detect an answer or failure.
         /// 
diff --git a/src/net/DtlsSrtp/Transform/SrtcpTransformer.cs b/src/net/DtlsSrtp/Transform/SrtcpTransformer.cs
index 8efb73adf..734f9b120 100644
--- a/src/net/DtlsSrtp/Transform/SrtcpTransformer.cs
+++ b/src/net/DtlsSrtp/Transform/SrtcpTransformer.cs
@@ -15,58 +15,58 @@
 // License:
 // BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
 // Original Source: AGPL-3.0 License
-//-----------------------------------------------------------------------------
-
+//-----------------------------------------------------------------------------
+
 using System.Collections.Concurrent;
-using System.Collections.Generic;
+using System.Collections.Generic;
 using System.Threading;
 
-namespace SIPSorcery.Net
+namespace SIPSorcery.Net
 {
     /// 
-    /// SRTCPTransformer implements PacketTransformer.
-    /// It encapsulate the encryption / decryption logic for SRTCP packets
-    ///
-    /// @author Bing SU (nova.su @gmail.com)
-    /// @author Werner Dittmann
-    /// 
-    public class SrtcpTransformer : IPacketTransformer
-    {
-        private int _isLocked = 0;
-        private RawPacket packet;
-
-        private SrtpTransformEngine forwardEngine;
-        private SrtpTransformEngine reverseEngine;
-
-        /** All the known SSRC's corresponding SRTCPCryptoContexts */
-        private ConcurrentDictionary contexts;
-
-        public SrtcpTransformer(SrtpTransformEngine engine) : this(engine, engine)
-        {
-
-        }
-
-        public SrtcpTransformer(SrtpTransformEngine forwardEngine, SrtpTransformEngine reverseEngine)
-        {
-            this.packet = new RawPacket();
-            this.forwardEngine = forwardEngine;
-            this.reverseEngine = reverseEngine;
-            this.contexts = new ConcurrentDictionary();
-        }
-
-        /// 
-        /// Encrypts a SRTCP packet
+    /// SRTCPTransformer implements PacketTransformer.
+    /// It encapsulate the encryption / decryption logic for SRTCP packets
+    ///
+    /// @author Bing SU (nova.su @gmail.com)
+    /// @author Werner Dittmann (Werner.Dittmann@t-online.de)
+    /// 
+    public class SrtcpTransformer : IPacketTransformer
+    {
+        private int _isLocked = 0;
+        private RawPacket packet;
+
+        private SrtpTransformEngine forwardEngine;
+        private SrtpTransformEngine reverseEngine;
+
+        /** All the known SSRC's corresponding SRTCPCryptoContexts */
+        private ConcurrentDictionary contexts;
+
+        public SrtcpTransformer(SrtpTransformEngine engine) : this(engine, engine)
+        {
+
+        }
+
+        public SrtcpTransformer(SrtpTransformEngine forwardEngine, SrtpTransformEngine reverseEngine)
+        {
+            this.packet = new RawPacket();
+            this.forwardEngine = forwardEngine;
+            this.reverseEngine = reverseEngine;
+            this.contexts = new ConcurrentDictionary();
+        }
+
+        /// 
+        /// Encrypts a SRTCP packet
         /// 
         /// plain SRTCP packet to be encrypted.
         /// encrypted SRTCP packet.
-        public byte[] Transform(byte[] pkt)
-        {
-            return Transform(pkt, 0, pkt.Length);
-        }
-
-        public byte[] Transform(byte[] pkt, int offset, int length)
-        {
-            var isLocked = Interlocked.CompareExchange(ref _isLocked, 1, 0) != 0;
+        public byte[] Transform(byte[] pkt)
+        {
+            return Transform(pkt, 0, pkt.Length);
+        }
+
+        public byte[] Transform(byte[] pkt, int offset, int length)
+        {
+            var isLocked = Interlocked.CompareExchange(ref _isLocked, 1, 0) != 0;
             try
             {
                 // Wrap the data into raw packet for readable format
@@ -90,23 +90,23 @@ public byte[] Transform(byte[] pkt, int offset, int length)
                 byte[] result = packet.GetData();
 
                 return result;
-            }
-            finally
+            }
+            finally
             {
                 //Unlock
                 if (!isLocked)
                     Interlocked.CompareExchange(ref _isLocked, 0, 1);
-            }
-        }
-
-        public byte[] ReverseTransform(byte[] pkt)
-        {
-            return ReverseTransform(pkt, 0, pkt.Length);
-        }
-
-        public byte[] ReverseTransform(byte[] pkt, int offset, int length)
-        {
-            var isLocked = Interlocked.CompareExchange(ref _isLocked, 1, 0) != 0;
+            }
+        }
+
+        public byte[] ReverseTransform(byte[] pkt)
+        {
+            return ReverseTransform(pkt, 0, pkt.Length);
+        }
+
+        public byte[] ReverseTransform(byte[] pkt, int offset, int length)
+        {
+            var isLocked = Interlocked.CompareExchange(ref _isLocked, 1, 0) != 0;
             try
             {
                 // wrap data into raw packet for readable format
@@ -133,36 +133,36 @@ public byte[] ReverseTransform(byte[] pkt, int offset, int length)
                     result = packet.GetData();
                 }
                 return result;
-            }
-            finally
+            }
+            finally
             {
                 //Unlock
                 if (!isLocked)
                     Interlocked.CompareExchange(ref _isLocked, 0, 1);
-            }
-        }
-
+            }
+        }
+
         /// 
         /// Close the transformer and underlying transform engine.
         /// The close functions closes all stored crypto contexts. This deletes key data
         /// and forces a cleanup of the crypto contexts.
-        /// 
-        public void Close()
-        {
-            forwardEngine.Close();
+        /// 
+        public void Close()
+        {
+            forwardEngine.Close();
             if (forwardEngine != reverseEngine)
             {
                 reverseEngine.Close();
-            }
-
-            var keys = new List(contexts.Keys);
-            foreach (var ssrc in keys)
+            }
+
+            var keys = new List(contexts.Keys);
+            foreach (var ssrc in keys)
             {
                 if (contexts.TryRemove(ssrc, out var context))
                 {
                     context.Close();
-                }
-            }
-        }
-    }
-}
+                }
+            }
+        }
+    }
+}
diff --git a/src/net/DtlsSrtp/Transform/SrtpCipherCTR.cs b/src/net/DtlsSrtp/Transform/SrtpCipherCTR.cs
index 1629eb098..ba0566a3b 100644
--- a/src/net/DtlsSrtp/Transform/SrtpCipherCTR.cs
+++ b/src/net/DtlsSrtp/Transform/SrtpCipherCTR.cs
@@ -1,39 +1,39 @@
-//-----------------------------------------------------------------------------
-// Filename: SrtpCipherCTR.cs
-//
-// Description: Implements SRTP Counter Mode Encryption.
-//
-// Derived From:
-// https://github.com/jitsi/jitsi-srtp/blob/master/src/main/java/org/jitsi/srtp/crypto/SrtpCipherCtr.java
-//
-// Author(s):
-// Rafael Soares (raf.csoares@kyubinteractive.com)
-//
-// History:
-// 01 Jul 2020	Rafael Soares   Created.
-//
-// License:
-// Customisations: BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
-// Original Source: Apache License: see below
+//-----------------------------------------------------------------------------
+// Filename: SrtpCipherCTR.cs
+//
+// Description: Implements SRTP Counter Mode Encryption.
+//
+// Derived From:
+// https://github.com/jitsi/jitsi-srtp/blob/master/src/main/java/org/jitsi/srtp/crypto/SrtpCipherCtr.java
+//
+// Author(s):
+// Rafael Soares (raf.csoares@kyubinteractive.com)
+//
+// History:
+// 01 Jul 2020	Rafael Soares   Created.
+//
+// License:
+// Customisations: BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
+// Original Source: Apache License: see below
 //-----------------------------------------------------------------------------
-
-/*
- * Copyright @ 2016 - present 8x8, Inc
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**
+
+/*
+ * Copyright @ 2016 - present 8x8, Inc
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/*
  * SRTPCipherF8 implements SRTP F8 Mode AES Encryption (AES-f8).
  * F8 Mode AES Encryption algorithm is defined in RFC3711, section 4.1.2.
  * 
@@ -59,10 +59,10 @@
  * 
  * @author Bing SU (nova.su@gmail.com)
  * @author Werner Dittmann 
- */
-
-using System.IO;
-using Org.BouncyCastle.Crypto;
+ */
+
+using System.IO;
+using Org.BouncyCastle.Crypto;
 
 namespace SIPSorcery.Net
 {
diff --git a/src/net/DtlsSrtp/Transform/SrtpCipherF8.cs b/src/net/DtlsSrtp/Transform/SrtpCipherF8.cs
index 06f1c0438..5952712e2 100644
--- a/src/net/DtlsSrtp/Transform/SrtpCipherF8.cs
+++ b/src/net/DtlsSrtp/Transform/SrtpCipherF8.cs
@@ -1,37 +1,37 @@
-//-----------------------------------------------------------------------------
-// Filename: SrtpCipherF8.cs
-//
-// Description: Implements SRTP F8 Mode Encryption for 128 bits block cipher.
-//
-// Derived From:
-// https://github.com/jitsi/jitsi-srtp/blob/master/src/main/java/org/jitsi/srtp/crypto/SrtpCipherF8.java
-//
-// Author(s):
-// Rafael Soares (raf.csoares@kyubinteractive.com)
-//
-// History:
-// 01 Jul 2020	Rafael Soares   Created.
-//
-// License:
-// Customisations: BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
-// Original Source: Apache License: see below
-//-----------------------------------------------------------------------------
-
-/*
- * Copyright @ 2015 - present 8x8, Inc
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *     http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
+//-----------------------------------------------------------------------------
+// Filename: SrtpCipherF8.cs
+//
+// Description: Implements SRTP F8 Mode Encryption for 128 bits block cipher.
+//
+// Derived From:
+// https://github.com/jitsi/jitsi-srtp/blob/master/src/main/java/org/jitsi/srtp/crypto/SrtpCipherF8.java
+//
+// Author(s):
+// Rafael Soares (raf.csoares@kyubinteractive.com)
+//
+// History:
+// 01 Jul 2020	Rafael Soares   Created.
+//
+// License:
+// Customisations: BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
+// Original Source: Apache License: see below
+//-----------------------------------------------------------------------------
+
+/*
+ * Copyright @ 2015 - present 8x8, Inc
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
 
 /**
  * SRTPCipherF8 implements SRTP F8 Mode AES Encryption (AES-f8).
@@ -58,10 +58,10 @@
  * We use AESCipher to handle basic AES encryption / decryption.
  * 
  * @author Bing SU (nova.su@gmail.com)
- * @author Werner Dittmann 
- */
-
-using System.IO;
+ * @author Werner Dittmann (werner.dittmann@t-online.de)
+ */
+
+using System.IO;
 using Org.BouncyCastle.Crypto;
 using Org.BouncyCastle.Crypto.Parameters;
 using Org.BouncyCastle.Utilities;
@@ -100,19 +100,19 @@ public static void DeriveForIV(IBlockCipher f8Cipher, byte[] key, byte[] salt)
              * full key.
              */
             System.Array.Copy(salt, 0, saltMask, 0, salt.Length);
-            for (int i = salt.Length; i < saltMask.Length; ++i)
-            {
-                saltMask[i] = 0x55;
-            }
+            for (int i = salt.Length; i < saltMask.Length; ++i)
+            {
+                saltMask[i] = 0x55;
+            }
 
             /*
              * XOR the original key with the above created mask to get the special
              * key.
              */
-            for (int i = 0; i < key.Length; i++)
-            {
-                maskedKey[i] = (byte)(key[i] ^ saltMask[i]);
-            }
+            for (int i = 0; i < key.Length; i++)
+            {
+                maskedKey[i] = (byte)(key[i] ^ saltMask[i]);
+            }
 
             /*
              * Prepare the f8Cipher with the special key to compute IV'
@@ -182,10 +182,10 @@ public static void ProcessBlock(IBlockCipher cipher, F8Context f8ctx,
              * XOR the previous key stream with IV'
              * ( S(-1) xor IV' )
              */
-            for (int i = 0; i < BLKLEN; i++)
-            {
-                f8ctx.S[i] ^= f8ctx.ivAccent[i];
-            }
+            for (int i = 0; i < BLKLEN; i++)
+            {
+                f8ctx.S[i] ^= f8ctx.ivAccent[i];
+            }
 
             /*
              * Now XOR (S(n-1) xor IV') with the current counter, then increment 
diff --git a/src/net/ICE/RTCIceCandidate.cs b/src/net/ICE/RTCIceCandidate.cs
index d404bb7f3..7e2264f76 100644
--- a/src/net/ICE/RTCIceCandidate.cs
+++ b/src/net/ICE/RTCIceCandidate.cs
@@ -396,7 +396,7 @@ public string toJSON()
         /// protocol and IP end point. Primary use case is to check whether a candidate
         /// is a match for a remote end point that a message has been received from.
         /// 
-        /// The protocol to check equivalence for.
+        /// The protocol to check equivalence for.
         /// The IP end point to check equivalence for.
         /// True if the candidate is deemed equivalent or false if not.
         public bool IsEquivalentEndPoint(RTCIceProtocol epPotocol, IPEndPoint ep)
@@ -466,4 +466,4 @@ private uint UpdateCrc32(uint start, byte[] buf)
         }
 
     }
-}
\ No newline at end of file
+}
diff --git a/src/net/ICE/RtpIceChannel.cs b/src/net/ICE/RtpIceChannel.cs
index 011e88500..9ce685ffc 100755
--- a/src/net/ICE/RtpIceChannel.cs
+++ b/src/net/ICE/RtpIceChannel.cs
@@ -2601,7 +2601,6 @@ protected override void OnRTPPacketReceived(UdpReceiver receiver, int localPort,
         /// 
         /// Sends a packet via a TURN relay server.
         /// 
-        /// The local socket to send the packet from.
         /// The peer destination end point.
         /// The data to send to the peer.
         /// The TURN server end point to send the relayed request to.
@@ -2701,4 +2700,4 @@ public override SocketError Send(RTPChannelSocketsEnum sendOn, IPEndPoint dstEnd
             }
         }
     }
-}
\ No newline at end of file
+}
diff --git a/src/net/RTCP/RTCPSession.cs b/src/net/RTCP/RTCPSession.cs
index 1bf465ce6..18af4171b 100644
--- a/src/net/RTCP/RTCPSession.cs
+++ b/src/net/RTCP/RTCPSession.cs
@@ -261,7 +261,7 @@ public void RecordRtpPacketSend(RTPPacket rtpPacket)
         /// Event handler for an RTCP packet being received from the remote party.
         /// 
         /// The end point the packet was received from.
-        /// The data received.
+        /// The data received.
         public void ReportReceived(IPEndPoint remoteEndPoint, RTCPCompoundPacket rtcpCompoundPacket)
         {
             try
diff --git a/src/net/RTP/MediaStream.cs b/src/net/RTP/MediaStream.cs
index 9d9016eaf..2d1d97ec9 100644
--- a/src/net/RTP/MediaStream.cs
+++ b/src/net/RTP/MediaStream.cs
@@ -402,11 +402,10 @@ protected void SendRtpRaw(byte[] data, uint timestamp, int markerBit, int payloa
         /// 
         /// Allows additional control for sending raw RTP payloads. No framing or other processing is carried out.
         /// 
-        /// The media type of the RTP packet being sent. Must be audio or video.
-        /// The RTP packet payload.
+        /// The RTP packet payload.
         /// The timestamp to set on the RTP header.
         /// The value to set on the RTP header marker bit, should be 0 or 1.
-        /// The payload ID to set in the RTP header.
+        /// The payload ID to set in the RTP header.
         ///  The RTP sequence number 
         public void SendRtpRaw(byte[] data, uint timestamp, int markerBit, int payloadType, ushort seqNum)
         {
@@ -416,11 +415,10 @@ public void SendRtpRaw(byte[] data, uint timestamp, int markerBit, int payloadTy
         /// 
         /// Allows additional control for sending raw RTP payloads. No framing or other processing is carried out.
         /// 
-        /// The media type of the RTP packet being sent. Must be audio or video.
-        /// The RTP packet payload.
+        /// The RTP packet payload.
         /// The timestamp to set on the RTP header.
         /// The value to set on the RTP header marker bit, should be 0 or 1.
-        /// The payload ID to set in the RTP header.
+        /// The payload ID to set in the RTP header.
         public void SendRtpRaw(byte[] data, uint timestamp, int markerBit, int payloadType)
         {
             SendRtpRaw(data, timestamp, markerBit, payloadType, false);
@@ -519,7 +517,6 @@ public void SendRtcpReport(RTCPCompoundPacket report)
         /// 
         /// Allows sending of RTCP feedback reports.
         /// 
-        /// The media type of the RTCP report  being sent. Must be audio or video.
         /// The feedback report to send.
         public void SendRtcpFeedback(RTCPFeedback feedback)
         {
@@ -718,7 +715,6 @@ protected void LogIfWrongSeqNumber(string trackType, RTPHeader header, MediaStre
         /// 
         /// Adjusts the expected remote end point for a particular media type.
         /// 
-        /// The media type of the RTP packet received.
         /// The SSRC from the RTP packet header.
         /// The actual remote end point that the RTP packet came from.
         /// True if remote end point for this media type was the expected one or it was adjusted. False if
@@ -773,8 +769,6 @@ protected bool AdjustRemoteEndPoint(uint ssrc, IPEndPoint receivedOnEndPoint)
         /// 
         /// Creates a new RTCP session for a media track belonging to this RTP session.
         /// 
-        /// The media type to create the RTP session for. Must be
-        /// audio or video.
         /// A new RTCPSession object. The RTCPSession must have its Start method called
         /// in order to commence sending RTCP reports.
         public Boolean CreateRtcpSession()
@@ -791,7 +785,6 @@ public Boolean CreateRtcpSession()
         /// 
         /// Sets the remote end points for a media type supported by this RTP session.
         /// 
-        /// The media type, must be audio or video, to set the remote end point for.
         /// The remote end point for RTP packets corresponding to the media type.
         /// The remote end point for RTCP packets corresponding to the media type.
         public void SetDestination(IPEndPoint rtpEndPoint, IPEndPoint rtcpEndPoint)
diff --git a/src/net/RTP/MediaStreamTrack.cs b/src/net/RTP/MediaStreamTrack.cs
index e14a45870..dad47da56 100644
--- a/src/net/RTP/MediaStreamTrack.cs
+++ b/src/net/RTP/MediaStreamTrack.cs
@@ -80,7 +80,7 @@ public class MediaStreamTrack
         /// 
         public List Capabilities { get; internal set; }
 
-        // 
+        /// 
         ///  a=extmap - Mapping for RTP header extensions
         /// 
         public Dictionary HeaderExtensions { get; }
@@ -205,7 +205,7 @@ public MediaStreamTrack(
         /// 
         /// Add a local audio track.
         /// 
-        /// The audio formats that the local application supports.
+        /// The audio formats that the local application supports.
         /// Optional. The stream status for the audio track, e.g. whether
         /// send and receive or only one of.
         public MediaStreamTrack(
@@ -229,7 +229,7 @@ public MediaStreamTrack(
         /// 
         /// Add a local video track.
         /// 
-        /// The video formats that the local application supports.
+        /// The video formats that the local application supports.
         /// Optional. The stream status for the video track, e.g. whether
         /// send and receive or only one of.
         public MediaStreamTrack(
diff --git a/src/net/RTP/Packetisation/H264Depacketiser.cs b/src/net/RTP/Packetisation/H264Depacketiser.cs
index 86d2a5e73..b30e7f4e6 100644
--- a/src/net/RTP/Packetisation/H264Depacketiser.cs
+++ b/src/net/RTP/Packetisation/H264Depacketiser.cs
@@ -1,16 +1,16 @@
-/// 
-/// Based in https://github.com/BogdanovKirill/RtspClientSharp/blob/master/RtspClientSharp/MediaParsers/H264VideoPayloadParser.cs 
-/// Distributed under MIT License
-/// 
-/// @author raf.csoares@kyubinteractive.com
-/// 
-
+
 using System;
 using System.Collections.Generic;
 using System.IO;
 
 namespace SIPSorcery.Net
 {
+    /// 
+    /// Based in https://github.com/BogdanovKirill/RtspClientSharp/blob/master/RtspClientSharp/MediaParsers/H264VideoPayloadParser.cs 
+    /// Distributed under MIT License
+    /// 
+    /// @author raf.csoares@kyubinteractive.com
+    /// 
     public class H264Depacketiser
     {
         const int SPS = 7;
diff --git a/src/net/RTP/RTPSession.cs b/src/net/RTP/RTPSession.cs
index 741035eb5..0362957ca 100644
--- a/src/net/RTP/RTPSession.cs
+++ b/src/net/RTP/RTPSession.cs
@@ -1585,7 +1585,7 @@ private void SetLocalTrackStreamStatus(MediaStreamTrack localTrack, MediaStreamS
         /// 
         /// Generates a session description from the provided list of MediaStream.
         /// 
-        /// The list of tracks to generate the session description for.
+        /// The list of tracks to generate the session description for.
         /// Optional. If set this address will be used as 
         /// the SDP Connection address. If not specified the Internet facing address will
         /// be used. IPAddress.Any and IPAddress. Any and IPv6Any are special cases. If they are set the respective
@@ -1799,7 +1799,6 @@ private SDP GetSessionDescription(List mediaStreamList, IPAddress c
         /// Creates a new RTP channel (which manages the UDP socket sending and receiving RTP
         /// packets) for use with this session.
         /// 
-        /// The type of media the RTP channel is for. Must be audio or video.
         /// A new RTPChannel instance.
         protected virtual RTPChannel CreateRtpChannel()
         {
diff --git a/src/net/RTP/VideoStream.cs b/src/net/RTP/VideoStream.cs
index e64c23a58..54dd24161 100644
--- a/src/net/RTP/VideoStream.cs
+++ b/src/net/RTP/VideoStream.cs
@@ -83,11 +83,11 @@ public bool HasVideo
         /// It's intended as a quick convenient way to send something like a test pattern image over an RTSP connection. More than likely it won't be suitable when a high
         /// quality image is required since the header used in this method does not support quantization tables.
         /// 
+        /// The duration in timestamp units of the payload (e.g. 3000 for 30fps).
         /// The raw encoded bytes of the JPEG image to transmit.
         /// The encoder quality of the JPEG image.
         /// The width of the JPEG image.
         /// The height of the JPEG image.
-        /// The rate at which the JPEG frames are being transmitted at. used to calculate the timestamp.
         public void SendJpegFrame(uint duration, int payloadTypeID, byte[] jpegBytes, int jpegQuality, int jpegWidth, int jpegHeight)
         {
             if (CheckIfCanSendRtpRaw())
@@ -129,8 +129,9 @@ public void SendJpegFrame(uint duration, int payloadTypeID, byte[] jpegBytes, in
         /// An Access Unit can contain one or more NAL's. The NAL's have to be parsed in order to be able to package 
         /// in RTP packets.
         /// 
-        /// See https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.264-201602-S!!PDF-E&type=items Annex B for byte stream specification.
+        /// See  Annex B for byte stream specification.
         /// 
+        // The same URL without XML escape sequences: https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.264-201602-S!!PDF-E&type=items
         public void SendH264Frame(uint duration, int payloadTypeID, byte[] accessUnit)
         {
             if (CheckIfCanSendRtpRaw())
@@ -150,8 +151,6 @@ public void SendH264Frame(uint duration, int payloadTypeID, byte[] accessUnit)
         /// The buffer containing the NAL to send.
         /// Should be set for the last NAL in the H264 access unit. Determines when the markbit gets set 
         /// and the timestamp incremented.
-        /// The destination end point to send to.
-        /// The video track to send on.
         private void SendH264Nal(uint duration, int payloadTypeID, byte[] nal, bool isLastNal)
         {
             //logger.LogDebug($"Send NAL {nal.Length}, is last {isLastNal}, timestamp {videoTrack.Timestamp}.");
@@ -204,7 +203,7 @@ private void SendH264Nal(uint duration, int payloadTypeID, byte[] nal, bool isLa
         /// 
         /// Sends a VP8 frame as one or more RTP packets.
         /// 
-        /// The timestamp to place in the RTP header. Needs
+        ///  The duration in timestamp units of the payload. Needs
         /// to be based on a 90Khz clock.
         /// The payload ID to place in the RTP header.
         /// The VP8 encoded payload.
diff --git a/src/net/RTSP/Mjpeg.cs b/src/net/RTSP/Mjpeg.cs
index 4c5de60e4..86c3992d7 100644
--- a/src/net/RTSP/Mjpeg.cs
+++ b/src/net/RTSP/Mjpeg.cs
@@ -321,7 +321,7 @@ static byte[] CreateQuantizationTables(uint type, uint Q, byte precision)
         /// Creates a Jpeg QuantizationTableMarker for each table given in the tables
         /// 
         /// The tables verbatim, either 1 or 2 (Luminance and Chrominance)
-        /// The table with marker and prefix/returns>
+        /// The table with marker and prefix
         static byte[] CreateQuantizationTablesMarker(ArraySegment tables, byte precision)
         {
             //List result = new List();
diff --git a/src/net/SCTP/Chunks/SctpDataChunk.cs b/src/net/SCTP/Chunks/SctpDataChunk.cs
index 2d32a07b5..2b2a15a48 100644
--- a/src/net/SCTP/Chunks/SctpDataChunk.cs
+++ b/src/net/SCTP/Chunks/SctpDataChunk.cs
@@ -98,7 +98,7 @@ private SctpDataChunk()
         /// without requiring it to be delivered to the remote part in order.
         /// Must be set to true for the first chunk in a user data payload.
         /// Must be set to true for the last chunk in a user data payload. Note that
-        ///  and  must both be set to true when the full payload
+        ///  and  must both be set to true when the full payload
         /// is being sent in a single data chunk.
         /// The Transmission Sequence Number for this chunk.
         /// Optional. The stream ID for this data chunk.
diff --git a/src/net/SCTP/SctpAssociation.cs b/src/net/SCTP/SctpAssociation.cs
index 4f43216fb..e823cc776 100644
--- a/src/net/SCTP/SctpAssociation.cs
+++ b/src/net/SCTP/SctpAssociation.cs
@@ -588,7 +588,7 @@ public void SendData(ushort streamID, uint ppid, string message)
         /// 
         /// The stream ID to sent the data on.
         /// The payload protocol ID for the data.
-        /// The byte data to send.
+        /// The byte data to send.
         public void SendData(ushort streamID, uint ppid, byte[] data)
         {
             if (_wasAborted)
diff --git a/src/net/SCTP/SctpDataReceiver.cs b/src/net/SCTP/SctpDataReceiver.cs
index 6842ee6fb..8106e42b0 100644
--- a/src/net/SCTP/SctpDataReceiver.cs
+++ b/src/net/SCTP/SctpDataReceiver.cs
@@ -33,6 +33,10 @@ public struct SctpDataFrame
         public uint PPID;
         public byte[] UserData;
 
+        /// The stream ID of the chunk.
+        /// The stream sequence number of the chunk. Will be 0 for unordered streams.
+        /// The payload protocol ID for the chunk.
+        /// The chunk data.
         public SctpDataFrame(bool unordered, ushort streamID, ushort streamSeqNum, uint ppid, byte[] userData)
         {
             Unordered = unordered;
diff --git a/src/net/SCTP/SctpDataSender.cs b/src/net/SCTP/SctpDataSender.cs
index 37ce5e823..dd814bf84 100644
--- a/src/net/SCTP/SctpDataSender.cs
+++ b/src/net/SCTP/SctpDataSender.cs
@@ -305,7 +305,7 @@ public void GotSack(SctpSackChunk sack)
         /// 
         /// The stream ID to sent the data on.
         /// The payload protocol ID for the data.
-        /// The byte data to send.
+        /// The byte data to send.
         public void SendData(ushort streamID, uint ppid, byte[] data)
         {
             lock (_sendQueue)
@@ -665,7 +665,7 @@ private int GetSendWaitMilliseconds()
         /// Updates the round trip time. 
         /// See https://datatracker.ietf.org/doc/html/rfc4960#section-6.3.1
         /// 
-        /// The last round trip time
+        /// The last sent and acknowledged chunk used to calculate the round trip time
         private void UpdateRoundTripTime(SctpDataChunk acknowledgedChunk)
         {
             // rfc 4960 6.3.1 C5: RTT measurements MUST NOT be made using packets that were retransmitted
diff --git a/src/net/SCTP/SctpPacket.cs b/src/net/SCTP/SctpPacket.cs
index 9f68bb5cf..de6e675cf 100644
--- a/src/net/SCTP/SctpPacket.cs
+++ b/src/net/SCTP/SctpPacket.cs
@@ -82,7 +82,7 @@ public class SctpPacket
         public SctpHeader Header;
 
         /// 
-        /// The list of one or recognised chunks after parsing with  
+        /// The list of one or recognised chunks after parsing with 
         /// or chunks that have been manually added for an outgoing SCTP packet.
         /// 
         public List Chunks;
diff --git a/src/net/SCTP/SctpTransport.cs b/src/net/SCTP/SctpTransport.cs
index 06f227761..dd166c11f 100644
--- a/src/net/SCTP/SctpTransport.cs
+++ b/src/net/SCTP/SctpTransport.cs
@@ -297,8 +297,8 @@ protected string GetCookieHMAC(byte[] buffer)
         /// 
         /// Send an SCTP packet with one of the error type chunks (ABORT or ERROR) to the remote peer.
         /// 
-        /// Set to true to use an ABORT chunk otherwise an ERROR chunk will be used.
-        /// The SCTP destination port.
+        /// Set to true to use an ABORT chunk otherwise an ERROR chunk will be used.
+        /// The SCTP destination port.
         /// The SCTP source port.
         /// If available the initial tag for the remote peer.
         /// The error to send.
diff --git a/src/net/SDP/SDPAudioVideoMediaFormat.cs b/src/net/SDP/SDPAudioVideoMediaFormat.cs
index f8b7dcdd8..c183c433f 100644
--- a/src/net/SDP/SDPAudioVideoMediaFormat.cs
+++ b/src/net/SDP/SDPAudioVideoMediaFormat.cs
@@ -54,7 +54,7 @@ public struct SDPAudioVideoMediaFormat
         /// Note (rj2): FormatID MUST be string (not int), in case ID is 't38' and type is 'image'
         /// Note to above: The FormatID is always numeric for profile "RTP/AVP" and "RTP/SAVP", see 
         /// https://tools.ietf.org/html/rfc4566#section-5.14 and section on "fmt":
-        /// "If the  sub-field is "RTP/AVP" or "RTP/SAVP" the 
+        /// "If the [proto] sub-field is "RTP/AVP" or "RTP/SAVP" the [fmt]
         /// sub-fields contain RTP payload type numbers"
         /// In the case of T38 the format name is "t38" but the formatID must be set as a dynamic ID.
         /// 
@@ -79,7 +79,7 @@ public struct SDPAudioVideoMediaFormat
         /// 
         /// // Example
         /// a=rtpmap:0 PCMU/8000
-        /// a=rtpmap:101 telephone-event/8000 <-- "101 telephone-event/8000" is the rtpmap properties.
+        /// a=rtpmap:101 telephone-event/8000 ← "101 telephone-event/8000" is the rtpmap properties.
         /// a=fmtp:101 0-16
         /// 
         /// 
@@ -91,7 +91,7 @@ public struct SDPAudioVideoMediaFormat
         /// // Example
         /// a=rtpmap:0 PCMU/8000
         /// a=rtpmap:101 telephone-event/8000 
-        /// a=fmtp:101 0-16                     <-- "101 0-16" is the fmtp attribute.
+        /// a=fmtp:101 0-16                     ← "101 0-16" is the fmtp attribute.
         /// 
         /// 
         public string Fmtp { get; }
@@ -108,7 +108,7 @@ public IEnumerable SupportedRtcpFeedbackMessages
         /// The standard name of the media format.
         /// 
         /// // Example
-        /// a=rtpmap:0 PCMU/8000                <-- "PCMU" is the media format name.
+        /// a=rtpmap:0 PCMU/8000                ← "PCMU" is the media format name.
         /// a=rtpmap:101 telephone-event/8000 
         /// a=fmtp:101 0-16
         /// 
diff --git a/src/net/WebRTC/IRTCPeerConnection.cs b/src/net/WebRTC/IRTCPeerConnection.cs
index 1919f67f9..e65c0a52c 100644
--- a/src/net/WebRTC/IRTCPeerConnection.cs
+++ b/src/net/WebRTC/IRTCPeerConnection.cs
@@ -54,6 +54,7 @@ public class RTCOfferOptions
     /// 
     public class RTCAnswerOptions
     {
+        /// 
         /// If set it indicates that any available ICE candidates should NOT be added
         /// to the offer SDP. By default "host" candidates should always be available
         /// and will be added to the offer SDP.
diff --git a/src/net/WebRTC/RTCPeerConnection.cs b/src/net/WebRTC/RTCPeerConnection.cs
index 3ed676cad..22181fb44 100644
--- a/src/net/WebRTC/RTCPeerConnection.cs
+++ b/src/net/WebRTC/RTCPeerConnection.cs
@@ -518,7 +518,7 @@ private bool InitializeCertificates2(RTCConfiguration configuration)
         /// 
         /// Event handler for ICE connection state changes.
         /// 
-        /// The new ICE connection state.
+        /// The new ICE connection state.
         private async void IceConnectionStateChange(RTCIceConnectionState iceState)
         {
             oniceconnectionstatechange?.Invoke(iceConnectionState);
@@ -624,7 +624,6 @@ private async void IceConnectionStateChange(RTCIceConnectionState iceState)
         /// Creates a new RTP ICE channel (which manages the UDP socket sending and receiving RTP
         /// packets) for use with this session.
         /// 
-        /// The type of media the RTP channel is for. Must be audio or video.
         /// A new RTPChannel instance.
         protected override RTPChannel CreateRtpChannel()
         {
@@ -1258,7 +1257,7 @@ void AddIceCandidates(SDPMediaAnnouncement announcement)
         }
 
         /// 
-        /// From RFC5764:
+        /// From RFC5764:  forward to RTP
         ///             |                |
@@ -1266,6 +1265,7 @@ void AddIceCandidates(SDPMediaAnnouncement announcement)
         ///             |                |
         ///             |       B< 2    -+--> forward to STUN
         ///             +----------------+
+        /// ]]>
         /// 
         /// The local port on the RTP socket that received the packet.
         /// The remote end point the packet was received from.
@@ -1314,7 +1314,7 @@ private void OnRTPDataReceived(int localPort, IPEndPoint remoteEP, byte[] buffer
         /// example is when a machine is behind a 1:1 NAT and the application wants a host 
         /// candidate with the public IP address to be included.
         /// 
-        /// The ICE candidate to add.
+        /// The ICE candidate to add.
         /// 
         /// var natCandidate = new RTCIceCandidate(RTCIceProtocol.udp, natAddress, natPort, RTCIceCandidateType.host);
         /// pc.addLocalIceCandidate(natCandidate);
@@ -1538,10 +1538,6 @@ private void OnSctpAssociationDataChannelOpened(ushort streamID)
         /// 
         /// Event handler for an SCTP DATA chunk being received on the SCTP association.
         /// 
-        /// The stream ID of the chunk.
-        /// The stream sequence number of the chunk. Will be 0 for unordered streams.
-        /// The payload protocol ID for the chunk.
-        /// The chunk data.
         private void OnSctpAssociationDataChunk(SctpDataFrame frame)
         {
             if (dataChannels.TryGetChannel(frame.StreamID, out var dc))
diff --git a/src/net/WebRTC/RTCPeerSctpAssociation.cs b/src/net/WebRTC/RTCPeerSctpAssociation.cs
index 599cd2da1..d7f9c6480 100644
--- a/src/net/WebRTC/RTCPeerSctpAssociation.cs
+++ b/src/net/WebRTC/RTCPeerSctpAssociation.cs
@@ -69,8 +69,6 @@ public class RTCPeerSctpAssociation : SctpAssociation
         /// 
         /// The DTLS transport that will be used to encapsulate the
         /// SCTP packets.
-        /// True if this peer will be the client within the association. This
-        /// dictates whether streams created use odd or even ID's.
         /// The source port to use when forming the association.
         /// The destination port to use when forming the association.
         /// Optional. The local UDP port being used for the DTLS connection. This
diff --git a/src/net/WebRTC/RTCSctpTransport.cs b/src/net/WebRTC/RTCSctpTransport.cs
index cc5f113c4..c918e3e00 100644
--- a/src/net/WebRTC/RTCSctpTransport.cs
+++ b/src/net/WebRTC/RTCSctpTransport.cs
@@ -179,8 +179,6 @@ public void Start(DatagramTransport dtlsTransport, bool isDtlsClient)
         /// 
         /// Attempts to create and initialise a new SCTP association with the remote party.
         /// 
-        /// The source port to use for the SCTP association.
-        /// The destination port to use for the SCTP association.
         public void Associate()
         {
             SetState(RTCSctpTransportState.Connecting);