//Todo, cleanup and allow existing Rtp and Rtcp socket.
/// <summary>
/// Will create a <see cref="RtpClient"/> based on the given parameters
/// </summary>
/// <param name="sessionDescription"></param>
/// <param name="sharedMemory"></param>
/// <param name="incomingEvents"></param>
/// <param name="rtcpEnabled"></param>
/// <returns></returns>
public static RtpClient FromSessionDescription(Sdp.SessionDescription sessionDescription, Common.MemorySegment sharedMemory = null, bool incomingEvents = true, bool rtcpEnabled = true, System.Net.Sockets.Socket existingSocket = null, int?rtpPort = null, int?rtcpPort = null, int remoteSsrc = 0, int minimumSequentialRtpPackets = 2, bool connect = true, System.Action <System.Net.Sockets.Socket> configure = null)
{
if (Common.IDisposedExtensions.IsNullOrDisposed(sessionDescription))
{
throw new System.ArgumentNullException("sessionDescription");
}
Sdp.Lines.SessionConnectionLine connectionLine = new Sdp.Lines.SessionConnectionLine(sessionDescription.ConnectionLine);
System.Net.IPAddress remoteIp = System.Net.IPAddress.Parse(connectionLine.Host), localIp;
System.Net.NetworkInformation.NetworkInterface localInterface;
//If the socket is NOT null and IS BOUND use the localIp of the same address family
if (object.ReferenceEquals(existingSocket, null).Equals(false) && existingSocket.IsBound)
{
//If the socket is IP based
if (existingSocket.LocalEndPoint is System.Net.IPEndPoint)
{
//Take the localIp from the LocalEndPoint
localIp = (existingSocket.LocalEndPoint as System.Net.IPEndPoint).Address;
}
else
{
throw new System.NotSupportedException("Please create an issue for your use case.");
}
}
else // There is no socket existing.
{
//If the remote address is the broadcast address or the remote address is multicast
if (System.Net.IPAddress.Broadcast.Equals(remoteIp) || IPAddressExtensions.IsMulticast(remoteIp))
{
//This interface should be the interface you plan on using for the Rtp communication
localIp = SocketExtensions.GetFirstMulticastIPAddress(remoteIp.AddressFamily, out localInterface);
}
else
{
//This interface should be the interface you plan on using for the Rtp communication
localIp = SocketExtensions.GetFirstUnicastIPAddress(remoteIp.AddressFamily, out localInterface);
}
}
RtpClient client = new RtpClient(sharedMemory, incomingEvents);
byte lastChannel = 0;
//Todo, check for session level ssrc
//if (remoteSsrc.Equals(0))
//{
// //Sdp.SessionDescriptionLine ssrcLine = sessionDescription.SsrcGroupLine; // SsrcLine @ the session level could imply Group
//}
//For each MediaDescription in the SessionDescription
foreach (Media.Sdp.MediaDescription md in sessionDescription.MediaDescriptions)
{
//Make a RtpClient.TransportContext from the MediaDescription being parsed.
TransportContext tc = TransportContext.FromMediaDescription(sessionDescription, lastChannel++, lastChannel++, md,
rtcpEnabled, remoteSsrc, minimumSequentialRtpPackets,
localIp, remoteIp, //The localIp and remoteIp
rtpPort, rtcpPort, //The remote ports to receive data from
connect, existingSocket, configure);
//Try to add the context
try
{
client.AddContext(tc);
}
catch (System.Exception ex)
{
TaggedExceptionExtensions.RaiseTaggedException(tc, "See Tag, Could not add the created TransportContext.", ex);
}
}
//Return the participant
return(client);
}
internal void ReleaseUnusedResources()
{
//Enumerate each context 'SETUP' in the session
if (m_RtpClient != null)
{
//Iterate each context in the client
foreach (var context in m_RtpClient.GetTransportContexts())
{
//Could be a property of the transport.. (especially the rtpclient.)
//E.g. IsInactive
//If the context does not have any activity
if (false == context.HasAnyRecentActivity)
{
//Session level logger
//Context has no activity
//See if there is still a source for the context
RtpClient.TransportContext sourceContext = GetSourceContext(context.MediaDescription);
//If there was a source context with activity
if (sourceContext != null && sourceContext.HasAnyRecentActivity)
{
//Remove the attachment from the source context to the session context
RemoveSource(Attached[sourceContext]);
//Session level logger
//Removed Attachment for sourceContext.Id
//Remove the reference to the sourceContext
sourceContext = null;
}
}
}
}
//Get rid of any attachment this ClientSession had which no longer have a context.
foreach (IMediaSource source in Attached.Keys.ToList().Where(s=> GetSourceContext(s.MediaDescription) == null))
{
//Remove the attached media
RemoveSource(source);
}
//Remove rtp theads
if (Playing.Count == 0)
{
if (m_RtpClient != null)
{
m_RtpClient.Dispose();
m_RtpClient = null;
}
}
}
/// <summary>
/// Sends the Rtcp Goodbye and detaches all sources
/// </summary>
public override void Dispose()
{
if (IsDisposed) return;
base.Dispose();
RemoveAllAttachmentsAndClearPlaying();
//Mark as disconnected
IsDisconnected = true;
//Disconnect the RtpClient so it's not hanging around wasting resources for nothing
if (m_RtpClient != null)
{
try
{
m_RtpClient.InterleavedData -= m_RtpClient_InterleavedData;
m_RtpClient.Dispose();
m_RtpClient = null;
}
catch { }
}
if (m_Buffer != null)
{
try
{
m_Buffer.Dispose();
m_Buffer = null;
}
catch { }
}
if (m_RtspSocket != null)
{
try
{
if (false == LeaveOpen) m_RtspSocket.Dispose();
m_RtspSocket = null;
}
catch { }
}
if (LastRequest != null)
{
try
{
LastRequest.Dispose();
LastRequest = null;
}
catch { }
}
if (LastResponse != null)
{
try
{
LastResponse.Dispose();
LastResponse = null;
}
catch { }
}
m_Server = m_Contained = null;
}
/// <summary>
///
/// </summary>
/// <param name="request"></param>
/// <param name="sourceContext"></param>
/// <returns></returns>
/// //TODO Should be SourceMedia and SourceContext.
internal RtspMessage ProcessSetup(RtspMessage request, RtpSource sourceStream, RtpClient.TransportContext sourceContext)
{
//Assign the sessionId now if it has not been assigned before.
if (SessionId == null) SessionId = m_Id.GetHashCode().ToString();
//We also have to send one back
string returnTransportHeader = null;
//Create a response
RtspMessage response = CreateRtspResponse(request);
Sdp.MediaDescription mediaDescription = sourceContext.MediaDescription;
bool rtcpDisabled = sourceStream.m_DisableQOS;
//Values in the header we need
int clientRtpPort = -1, clientRtcpPort = -1, serverRtpPort = -1, serverRtcpPort = -1, localSsrc = 0, remoteSsrc = 0;
//Cache this to prevent having to go to get it every time down the line
IPAddress sourceIp = IPAddress.Any;
string mode;
bool unicast, multicast, interleaved, multiplexing;
byte dataChannel = 0, controlChannel = 1;
//Get the transport header
string transportHeader = request[RtspHeaders.Transport];
//If that is not present we cannot determine what transport the client wants
if (string.IsNullOrWhiteSpace(transportHeader) ||
false == (transportHeader.Contains("RTP")) ||
false == RtspHeaders.TryParseTransportHeader(transportHeader,
out localSsrc, out sourceIp, out serverRtpPort, out serverRtcpPort, out clientRtpPort, out clientRtcpPort,
out interleaved, out dataChannel, out controlChannel, out mode, out unicast, out multicast))
{
return CreateRtspResponse(request, RtspStatusCode.BadRequest, "Invalid Transport Header");
}
//RTCP-mux: when RTSP 2.0 is official... (Along with Server Sent Messages)
//Check if the ssrc was 0 which indicates any id
if (localSsrc == 0) localSsrc = RFC3550.Random32((int)sourceContext.MediaDescription.MediaType);
//Could also randomize the setupContext sequenceNumber here.
//We need to make an TransportContext in response to a setup
RtpClient.TransportContext setupContext = null;
//Check for already setup stream and determine if the stream needs to be setup again or just updated
if (Attached.ContainsKey(sourceContext))
{
//The contex may already existm should look first by ssrc.
setupContext = m_RtpClient.GetContextForMediaDescription(sourceContext.MediaDescription);
//If the context exists
if (setupContext != null)
{
//Update the ssrc if it doesn't match.
if (localSsrc != 0 && setupContext.SynchronizationSourceIdentifier != localSsrc)
{
setupContext.SynchronizationSourceIdentifier = localSsrc;
if (remoteSsrc != 0 && setupContext.RemoteSynchronizationSourceIdentifier != remoteSsrc) setupContext.RemoteSynchronizationSourceIdentifier = remoteSsrc;
}
multicast = Media.Common.Extensions.IPAddress.IPAddressExtensions.IsMulticast(((IPEndPoint)setupContext.RemoteRtp).Address);
interleaved = setupContext.RtpSocket.ProtocolType == ProtocolType.Tcp && SharesSocket;
//Then indicate the information for that context in the return transport header.
returnTransportHeader = RtspHeaders.TransportHeader(setupContext.MediaDescription.MediaProtocol, setupContext.SynchronizationSourceIdentifier, ((IPEndPoint)m_RtspSocket.RemoteEndPoint).Address, ((IPEndPoint)setupContext.RemoteRtp).Port, ((IPEndPoint)setupContext.RemoteRtcp).Port, ((IPEndPoint)setupContext.LocalRtp).Port, ((IPEndPoint)setupContext.LocalRtcp).Port, false == multicast, multicast, null, interleaved, setupContext.DataChannel, setupContext.ControlChannel);
setupContext.LeaveOpen = interleaved;
//Attach logger (have option?)
m_RtpClient.Logger = m_Server.Logger;
goto UpdateContext;
}
}
//Should determine intervals here for Rtcp from SessionDescription
//Should determine if aggregate operation is allowed
//Maybe setting up both udp and tcp at the same time? clientRtpPort needs to be nullable.
//Maybe better to just give tokens from the function ..
//Without explicitly checking for !interleaved VLC will recieve what it thinks are RTSP responses unless RTSP Interleaved is Forced.
//Was trying to Quicktime to pickup RTSP Interleaved by default on the first response but it doesn't seem that easy (quick time tries to switch but fails?)
//If the source does not force TCP and interleaved was not given and this is a unicast or multicast connection
if (false == interleaved && (unicast || multicast))
{
//Check requested transport is allowed by server
if (sourceStream.ForceTCP)//The client wanted Udp and Tcp was forced
{
//Return the result
var result = CreateRtspResponse(request, RtspStatusCode.UnsupportedTransport);
//Indicate interleaved is required.
result.SetHeader(RtspHeaders.Transport, RtspHeaders.TransportHeader(RtpClient.RtpAvpProfileIdentifier + "/TCP", localSsrc, ((IPEndPoint)m_RtspSocket.RemoteEndPoint).Address, null, null, null, null, null, false, null, true, dataChannel, controlChannel));
return result;
}
//QuickTime debug
if (clientRtpPort == 0) clientRtpPort = Media.Common.Extensions.Socket.SocketExtensions.FindOpenPort(ProtocolType.Udp, 30000, true);
if (clientRtcpPort == 0) clientRtcpPort = clientRtpPort + 1;
if (serverRtpPort == 0) serverRtpPort = Media.Common.Extensions.Socket.SocketExtensions.FindOpenPort(ProtocolType.Udp, 30000, true);
if (serverRtcpPort == 0) serverRtcpPort = serverRtpPort + 1;
//Ensure the ports are allowed to be used.
if (m_Server.MaximumUdpPort.HasValue &&
(clientRtpPort > m_Server.MaximumUdpPort || clientRtcpPort > m_Server.MaximumUdpPort))
{
//Handle port out of range
return CreateRtspResponse(request, RtspStatusCode.BadRequest, "Requested Udp Ports were out of range. Maximum Port = " + m_Server.MaximumUdpPort);
}
//Create sockets to reserve the ports.
var localAddress = ((IPEndPoint)m_RtspSocket.LocalEndPoint).Address;
Socket tempRtp = Media.Common.Extensions.Socket.SocketExtensions.ReservePort(SocketType.Dgram, ProtocolType.Udp, localAddress, clientRtpPort);
Socket tempRtcp = Media.Common.Extensions.Socket.SocketExtensions.ReservePort(SocketType.Dgram, ProtocolType.Udp, localAddress, clientRtcpPort);
//Check if the client was already created.
if (m_RtpClient == null || m_RtpClient.IsDisposed)
{
//Create a sender using a new segment on the existing buffer.
m_RtpClient = new RtpClient(new Common.MemorySegment(m_Buffer));
//Dont handle frame changed events from the client
m_RtpClient.FrameChangedEventsEnabled = false;
//Dont handle packets from the client
m_RtpClient.HandleIncomingRtpPackets = false;
//Attach the Interleaved data event
m_RtpClient.InterleavedData += m_RtpClient_InterleavedData;
//Attach logger (have option?)
m_RtpClient.Logger = m_Server.Logger;
//Use default data and control channel (Should be option?)
setupContext = new RtpClient.TransportContext(0, 1, localSsrc, mediaDescription, false == rtcpDisabled, remoteSsrc, 0);
}
else //The client was already created.
{
//Have to calculate next data and control channel
RtpClient.TransportContext lastContext = m_RtpClient.GetTransportContexts().LastOrDefault();
if (lastContext != null) setupContext = new RtpClient.TransportContext((byte)(lastContext.DataChannel + 2), (byte)(lastContext.ControlChannel + 2), localSsrc, mediaDescription, false == rtcpDisabled, remoteSsrc, 0);
else setupContext = new RtpClient.TransportContext(dataChannel, controlChannel, localSsrc, mediaDescription, false == rtcpDisabled, remoteSsrc, 0);
}
//Initialize the Udp sockets
setupContext.Initialize(localAddress, ((IPEndPoint)m_RtspSocket.RemoteEndPoint).Address, serverRtpPort, serverRtcpPort, clientRtpPort, clientRtcpPort);
//Ensure the receive buffer size is updated for that context.
Media.Common.ISocketReferenceExtensions.SetReceiveBufferSize(((Media.Common.ISocketReference)setupContext), m_Buffer.Count);
////Check if the punch packets made it out.
//if ((setupContext.IsRtpEnabled && ((IPEndPoint)setupContext.RemoteRtp).Port == 0)
// ||
// (setupContext.IsRtcpEnabled && ((IPEndPoint)setupContext.RemoteRtcp).Port == 0))
//{
// //Response should be a 461 or we should indicate the remote party is not yet listening in the response
// //Could also use StatusCode (100) with a reason phrase or header
//}
//Add the transportChannel
m_RtpClient.TryAddContext(setupContext);
//Create the returnTransportHeader (Should be setupContext.SynchronizationSourceIdentifier)
returnTransportHeader = RtspHeaders.TransportHeader(RtpClient.RtpAvpProfileIdentifier, localSsrc, ((IPEndPoint)m_RtspSocket.RemoteEndPoint).Address, clientRtpPort, clientRtcpPort, serverRtpPort, serverRtcpPort, true, false, null, false, 0, 0);
tempRtp.Dispose();
tempRtp = null;
tempRtcp.Dispose();
tempRtcp = null;
}
//Check for 'interleaved' token or TCP being forced
if (sourceStream.ForceTCP || interleaved)
{
//Check if the client was already created.
if (m_RtpClient == null || m_RtpClient.IsDisposed)
{
//Create a sender using a new segment on the existing buffer.
m_RtpClient = new RtpClient(new Common.MemorySegment(m_Buffer));
m_RtpClient.InterleavedData += m_RtpClient_InterleavedData;
//Attach logger (have option?)
m_RtpClient.Logger = m_Server.Logger;
#region Unused [Helps with debugging]
//m_RtpClient.RtcpPacketReceieved += m_RtpClient_RecievedRtcp;
//m_RtpClient.RtpPacketReceieved += m_RtpClient_RecievedRtp;
//m_RtpClient.RtcpPacketSent += m_RtpClient_SentRtcp;
#endregion
//Dont handle frame changed events from the client
m_RtpClient.FrameChangedEventsEnabled = false;
//Dont handle packets from the client
m_RtpClient.HandleIncomingRtpPackets = false;
//Create a new Interleave (don't use what was given as data or control channels)
setupContext = new RtpClient.TransportContext((byte)(dataChannel = 0), (byte)(controlChannel = 1), localSsrc, mediaDescription, m_RtspSocket, false == rtcpDisabled, remoteSsrc, 0);
//Add the transportChannel the client requested
m_RtpClient.TryAddContext(setupContext);
//Initialize the Interleaved Socket
setupContext.Initialize(m_RtspSocket, m_RtspSocket);
}
else //The client was already created
{
//Have to calculate next data and control channel
RtpClient.TransportContext lastContext = m_RtpClient.GetTransportContexts().LastOrDefault();
//Don't use what was given as data or control channels
if (lastContext != null) setupContext = new RtpClient.TransportContext(dataChannel = (byte)(lastContext.DataChannel + 2), controlChannel = (byte)(lastContext.ControlChannel + 2), localSsrc, mediaDescription, false == rtcpDisabled, remoteSsrc, 0);
else setupContext = new RtpClient.TransportContext(dataChannel, controlChannel, localSsrc, mediaDescription, false == rtcpDisabled, remoteSsrc, 0);
//Add the transportChannel the client requested
m_RtpClient.TryAddContext(setupContext);
//Initialize the current TransportChannel with the interleaved Socket
setupContext.Initialize(m_RtspSocket, m_RtspSocket);
}
//Create the returnTransportHeader
//returnTransportHeader = RtspHeaders.TransportHeader(RtpClient.RtpAvpProfileIdentifier + "/TCP", setupContext.SynchronizationSourceIdentifier, ((IPEndPoint)m_RtspSocket.RemoteEndPoint).Address, LocalEndPoint.Port, LocalEndPoint.Port, ((IPEndPoint)RemoteEndPoint).Port, ((IPEndPoint)RemoteEndPoint).Port, true, false, null, true, dataChannel, controlChannel);
//Leave the socket open when disposing the RtpClient
setupContext.LeaveOpen = true;
returnTransportHeader = RtspHeaders.TransportHeader(RtpClient.RtpAvpProfileIdentifier + "/TCP", localSsrc, ((IPEndPoint)m_RtspSocket.RemoteEndPoint).Address, null, null, null, null, null, false, null, true, dataChannel, controlChannel);
}
//Add the new source
Attached.Add(sourceContext, sourceStream);
UpdateContext:
//Synchronize the context sequence numbers
setupContext.SequenceNumber = sourceContext.SequenceNumber;
//Start and end times are always equal.
setupContext.MediaStartTime = sourceContext.MediaStartTime;
setupContext.MediaEndTime = sourceContext.MediaEndTime;
//Set the returnTransportHeader to the value above
response.SetHeader(RtspHeaders.Transport, returnTransportHeader);
//Give the sessionid for the transport setup
response.SetHeader(RtspHeaders.Session, SessionId);
return response;
}
/// <summary>
/// Will create a <see cref="RtpClient"/> based on the given parameters
/// </summary>
/// <param name="sessionDescription"></param>
/// <param name="sharedMemory"></param>
/// <param name="incomingEvents"></param>
/// <param name="rtcpEnabled"></param>
/// <returns></returns>
public static RtpClient FromSessionDescription(Sdp.SessionDescription sessionDescription, Common.MemorySegment sharedMemory = null, bool incomingEvents = true, bool rtcpEnabled = true, Socket existingSocket = null, int? rtpPort = null, int? rtcpPort = null, int remoteSsrc = 0, int minimumSequentialRtpPackets = 2)
{
if (sessionDescription == null) throw new ArgumentNullException("sessionDescription");
Sdp.Lines.SessionConnectionLine connectionLine = new Sdp.Lines.SessionConnectionLine(sessionDescription.ConnectionLine);
IPAddress remoteIp = IPAddress.Parse(connectionLine.IPAddress), localIp = Media.Common.Extensions.Socket.SocketExtensions.GetFirstIPAddress(remoteIp.AddressFamily);
RtpClient participant = new RtpClient(sharedMemory, incomingEvents);
byte lastChannel = 0;
bool hasSocket = existingSocket != null;
foreach (Media.Sdp.MediaDescription md in sessionDescription.MediaDescriptions)
{
TransportContext tc = TransportContext.FromMediaDescription(sessionDescription, lastChannel++, lastChannel++, md, rtcpEnabled, remoteSsrc, minimumSequentialRtpPackets);
//Find range info in the SDP
var rangeInfo = md.RangeLine;
//If there is a range directive
if (rangeInfo == null)
{
rangeInfo = sessionDescription.RangeLine;
if (rangeInfo != null)
{
string type;
Sdp.SessionDescription.TryParseRange(rangeInfo.Parts[0], out type, out tc.m_StartTime, out tc.m_EndTime);
}
//else if (sessionDescription.TimeDescriptions.Count > 0)
//{
//tc.MediaStartTime = TimeSpan.FromMilliseconds();
//tc.MediaEndTime = TimeSpan.FromMilliseconds();
//}
}
//Check for udp if no existing socket was given
if (false == hasSocket && string.Compare(md.MediaProtocol, Media.Rtp.RtpClient.RtpAvpProfileIdentifier, true) == 0)
{
int localPort = Media.Common.Extensions.Socket.SocketExtensions.FindOpenPort(ProtocolType.Udp);
tc.Initialize(localIp, remoteIp, localPort++, localPort++, rtpPort ?? md.MediaPort, rtcpPort ?? md.MediaPort + 1);
}
else if (hasSocket)//If had a socket use it
{
tc.Initialize(existingSocket);
}
else
{
tc.Initialize(localIp, remoteIp, rtpPort ?? md.MediaPort);
}
//Try to add the context
try
{
participant.AddContext(tc);
}
catch (Exception ex)
{
Media.Common.Extensions.Exception.ExceptionExtensions.TryRaiseTaggedException(tc, "See Tag, Could not add the created TransportContext.", ex);
}
}
//Return the participant
return participant;
}
public static bool EnableFeedbackReports(RtpClient client)
{
if (FeedbackInstances.Add(client))
{
client.RtcpPacketReceieved += SendFeedback;
return true;
}
return false;
}
public static bool DisableFeedbackReports(RtpClient client)
{
if (FeedbackInstances.Remove(client))
{
client.RtcpPacketReceieved -= SendFeedback;
return true;
}
return false;
}
/// <summary>
/// Returns the amount of bytes read to completely read the application layer framed data
/// Where a negitive return value indicates no more data remains.
/// </summary>
/// <param name="received">How much data was received</param>
/// <param name="frameChannel">The output of reading a frameChannel</param>
/// <param name="context">The context assoicated with the frameChannel</param>
/// <param name="offset">The reference to offset to look for framing data</param>
/// <param name="raisedEvent">Indicates if an event was raised</param>
/// <param name="buffer">The optional buffer to use.</param>
/// <returns>The amount of bytes the frame data SHOULD have</returns>
int ReadApplicationLayerFraming(int received, out byte frameChannel, out RtpClient.TransportContext context, ref int offset, out bool raisedEvent, byte[] buffer = null)
{
//There is no relevant TransportContext assoicated yet.
context = null;
//The channel of the frame - The Framing Method
frameChannel = default(byte);
raisedEvent = false;
if (received <= 0) return -1;
buffer = buffer ?? m_Buffer.Array;
int bufferLength = buffer.Length, bufferOffset = offset;
//Look for the frame control octet
int startOfFrame = Array.IndexOf<byte>(buffer, BigEndianFrameControl, bufferOffset, received);
int frameLength = 0;
//If not found everything belongs to the upper layer
if (startOfFrame == -1)
{
//System.Diagnostics.Debug.WriteLine("Interleaving: " + received);
OnInterleavedData(buffer, bufferOffset, received);
raisedEvent = true;
//Indicate the amount of data consumed.
return received;
}
else if (startOfFrame > offset) // If the start of the frame is not at the beginning of the buffer
{
//Determine the amount of data which belongs to the upper layer
int upperLayerData = startOfFrame - bufferOffset;
//System.Diagnostics.Debug.WriteLine("Moved To = " + startOfFrame + " Of = " + received + " - Bytes = " + upperLayerData + " = " + Encoding.ASCII.GetString(m_Buffer, mOffset, startOfFrame - mOffset));
OnInterleavedData(buffer, bufferOffset, upperLayerData);
raisedEvent = true;
//Indicate length from offset until next possible frame. (should always be positive, if somehow -1 is returned this will signal a end of buffer to callers)
//If there is more data related to upperLayerData it will be evented in the next run. (See RtspClient ProcessInterleaveData notes)
return upperLayerData;
}
//If there is not enough data for a frame header return
if (bufferOffset + InterleavedOverhead > bufferLength) return -1;
//Todo Determine from Context to use control channel and length. (Check MediaDescription)
//NEEDS TO HANDLE CASES WHERE RFC4571 Framing are in play and no $ or Channel are used....
//The amount of data needed for the frame comes from TryReadFrameHeader
frameLength = TryReadFrameHeader(buffer, bufferOffset, out frameChannel, true, BigEndianFrameControl, true);
//Assign a context if there is a frame of any size
if (frameLength >= 0)
{
//Assign the context
context = GetContextByChannels(frameChannel);
//Increase the result by the size of the header
frameLength += InterleavedOverhead;
}
//Return the amount of bytes or -1 if any error occured.
return frameLength;
}
