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 @@ COPYRIGHT NOTICE
/**
* 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 @@ COPYRIGHT NOTICE
* @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 @@ COPYRIGHT NOTICE
* 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 DeriveForIV(IBlockCipher f8Cipher, byte[] key, byte[] salt)
* 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 uint MaximumBandwidth
///
/// 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 uint MaximumBandwidth
///
/// 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);