public ChildMedia(SourceMedia source)
: base(source.Name, source.Source)
{
if (!source.IsParent) throw new ArgumentException("Cannot make a Child of a Child.");
m_Parent = source;
//m_Child = true;
}
public ChildMedia(SourceMedia source)
: base(source.Name, source.Source)
{
if (!source.IsParent)
{
throw new ArgumentException("Cannot make a Child of a Child.");
}
m_Parent = source;
//m_Child = true;
}
/// <summary>
/// Process a Rtsp DESCRIBE.
/// Re-writes the Sdp.SessionDescription in a manner which contains the values of the server and not of the origional source.
/// </summary>
/// <param name="describeRequest">The request received from the server</param>
/// <param name="source">Tje source stream to describe</param>
/// <returns>A RtspMessage with a Sdp.SessionDescription in the Body and ContentType set to application/sdp</returns>
internal RtspMessage ProcessDescribe(RtspMessage describeRequest, SourceMedia source)
{
RtspMessage describeResponse = CreateRtspResponse(describeRequest);
describeResponse.SetHeader(RtspHeaders.ContentType, Sdp.SessionDescription.MimeType);
//Don't cache this SDP
describeResponse.SetHeader(RtspHeaders.CacheControl, "no-cache");
//describeResponse.SetHeader(RtspHeaders.Pragma, "no-cache");
//If desired you will need a way to determine when the last time the sdp was modified, could use the sessionId from the SDP or its' created property etc.)
//describeResponse.SetHeader(RtspHeaders.LastModified,
if (describeRequest.Location.ToString().ToLowerInvariant().Contains("live"))
{
describeResponse.SetHeader(RtspHeaders.ContentBase, "rtsp://" + ((IPEndPoint)m_RtspSocket.LocalEndPoint).Address.ToString() + "/live/" + source.Id + '/');
}
else
{
describeResponse.SetHeader(RtspHeaders.ContentBase, describeRequest.Location.ToString());
}
//Create a Session Description to describe the media requested
using (var sessionDescription = CreateSessionDescription(source))
{
//Set the body
describeResponse.Body = sessionDescription.ToString();
//Clients sessionId is created from the Sdp's SessionId Line
//if (string.IsNullOrWhiteSpace(SessionId)) SessionId = sessionDescription.SessionId;
}
//Return the resulting message
return describeResponse;
}
/// <summary>
/// Dynamically creates a Sdp.SessionDescription for the given SourceStream using the information already present and only re-writing the necessary values.
/// </summary>
/// <param name="stream">The source stream to create a SessionDescription for</param>
/// <returns>The created SessionDescription</returns>
internal Sdp.SessionDescription CreateSessionDescription(SourceMedia stream)
{
if (stream == null) throw new ArgumentNullException("stream");
//else if (SessionDescription != null) throw new NotImplementedException("There is already a m_SessionDescription for this session, updating is not implemented at this time");
string sessionId = Media.Ntp.NetworkTimeProtocol.DateTimeToNptTimestamp(DateTime.UtcNow).ToString(), sessionVersion = Media.Ntp.NetworkTimeProtocol.DateTimeToNptTimestamp(DateTime.UtcNow).ToString();
string originatorString = "ASTI-Media-Server " + sessionId + " " + sessionVersion + " IN " + (m_RtspSocket.AddressFamily == AddressFamily.InterNetworkV6 ? "IP6 " : "IP4 " ) + ((IPEndPoint)m_RtspSocket.LocalEndPoint).Address.ToString();
string sessionName = "ASTI-Streaming-Session-" + stream.Name;
Sdp.SessionDescription sdp;
RtpClient sourceClient;
if (stream is RtpSource)
{
RtpSource rtpSource = stream as RtpSource;
//Make the new SessionDescription
sdp = new Sdp.SessionDescription(rtpSource.SessionDescription.ToString());
//Remove the old connection lines if they exist
while(sdp.ConnectionLine != null) sdp.Remove(sdp.ConnectionLine, false);
sourceClient = rtpSource.RtpClient;
}
else
{
sdp = new Sdp.SessionDescription(0);
}
//Change the DocumentVersion and update the name
sdp.SessionName = sessionName;
//Change the DocumentVersion and update the originator
sdp.OriginatorAndSessionIdentifier = originatorString;
//Type = broadcast
//charset
string protcol = RtspMessage.MessageIdentifier.ToLowerInvariant(), controlLineBase = "a=control:" + protcol + "://" + ((IPEndPoint)(m_RtspSocket.LocalEndPoint)).Address.ToString() + "/live/" + stream.Id;
//check for rtspu later...
//Find an existing control line
Media.Sdp.SessionDescriptionLine controlLine = sdp.ControlLine;
//If there was one remove it
while (controlLine != null)
{
sdp.Remove(controlLine);
controlLine = sdp.ControlLine;
}
//Determine if session level control line should be present
//Rewrite a new connection line
string addressString = LocalEndPoint.Address.ToString();// +"/127/2";
//int lastPort = Utility.FindOpenPort( stream.m_ForceTCP ? ProtocolType.Tcp : ProtocolType.Udp);
//Indicate a port in the sdp, setup should also use this port, this should essentially reserve the port for the setup process...
//if (!stream.m_ForceTCP)
//addressString += "/127" +'/' + lastPort + 1;
//else
//addressString += + ((IPEndPoint)RemoteEndPoint).Port;
//Check for the existing connectionLine
Sdp.Lines.SessionConnectionLine connectionLine = sdp.ConnectionLine as Sdp.Lines.SessionConnectionLine;
//Add the new line if needed
if (connectionLine == null) sdp.ConnectionLine = connectionLine = new Sdp.Lines.SessionConnectionLine()
{
ConnectionAddress = addressString,
ConnectionAddressType = m_RtspSocket.AddressFamily == AddressFamily.InterNetworkV6 ? Media.Sdp.Lines.SessionConnectionLine.IP6 : Media.Sdp.Lines.SessionConnectionLine.IP4,
ConnectionNetworkType = Media.Sdp.Lines.SessionConnectionLine.InConnectionToken
};
IEnumerable<Sdp.SessionDescriptionLine> bandwithLines;
//Indicate that the server will not accept media as input for this session
//Put the attribute in the Session Description,
//Should check that its not already set?
//sdp.Add(new Sdp.SessionDescriptionLine("a=recvonly"));
//Remove any existing session range lines, don't upate the version
while (sdp.RangeLine != null) sdp.Remove(sdp.RangeLine, false);
//Todo add a Range line which shows the length of this media.
//Iterate the source MediaDescriptions, could just create a new one with the fmt which contains the profile level information
foreach (Sdp.MediaDescription md in sdp.MediaDescriptions)
{
//Find a control line
controlLine = md.ControlLine;
//Rewrite it if present to reflect the appropriate MediaDescription
while (controlLine != null)
{
md.Remove(controlLine);
controlLine = md.ControlLine;
}
//Remove old bandwith lines
bandwithLines = md.BandwidthLines;
//Remove existing bandwidth information, should check for AS
if(stream.m_DisableQOS) foreach (Sdp.SessionDescriptionLine line in bandwithLines) md.Remove(line);
//Remove all other alternate information
//Should probably only remove certain ones.
foreach (Sdp.SessionDescriptionLine line in md.Lines.Where(l => l.Parts.Any(p => p.Contains("alt"))).ToArray()) md.Remove(line);
//Add a control line for the MedaiDescription (which is `rtsp://./Id/audio` (video etc)
//Should be a TrackId and not the media type to allow more then one media type to be controlled.
//e.g. Two audio streams or text streams is valid.
md.Add(new Sdp.SessionDescriptionLine("a=control:" + "/live/" + stream.Id + '/' + md.MediaType));
//Add the connection line for the media
//md.Add(connectionLine);
//Should check for Timing Info and update for playing streams
if (stream.m_DisableQOS)
{
md.Add(Sdp.Lines.SessionBandwidthLine.DisabledSendLine);
md.Add(Sdp.Lines.SessionBandwidthLine.DisabledReceiveLine);
md.Add(Sdp.Lines.SessionBandwidthLine.DisabledApplicationSpecificLine);
}
//else
//{
// //ToDo use whatever values are defined for the session's bandwidth atp
// md.Add(new Sdp.SessionDescriptionLine("b=RS:140"));
// md.Add(new Sdp.SessionDescriptionLine("b=RR:140"));
// md.Add(new Sdp.SessionDescriptionLine("b=AS:0")); //Determine if AS needs to be forwarded
//}
//Should actually reflect outgoing port for this session
md.MediaPort = 0;
//Determine if attached and set the MediaPort.
if (m_RtpClient != null)
{
var context = m_RtpClient.GetContextForMediaDescription(md);
if (context != null) md.MediaPort = ((IPEndPoint)context.RemoteRtp).Port;
//Set any other variables which may be been changed in the session
}
#region Independent TCP
//if (!stream.m_ForceTCP)
//{
// //md.MediaPort = lastPort;
// //lastPort += 2;
//}
//else
//{
// //VLC `Blows up` when this happens
// //bad SDP "m=" line: m=audio 40563 TCP/RTP/AVP 96
// //md.MediaProtocol = "TCP/RTP/AVP";
// //fmt should be the same
// //This mainly implies that stand-alone RTP over TCP is occuring anyway.
// //Since this code supports the RtspServer this is fine for now.
// //The RtpClient also deals with the framing from RTSP when used in conjunction with so..
// //This needs to be addressed in the RtpClient which allows currently allows Rtp and Rtcp to be duplexed in TCP and UDP
// //but does not handle the case of TCP when a sender wants to connect with 2 seperate TCP sockets as per RFC4571.
// //a=setup:passive
// //a=connection:new
// //The RtpClient would then need to have a RtcpSocket ready on the 'standby' just in case the remote end point connected and began sending the data.
// //The other way to hanle this would be use only a single socket in both cases and change the remote endpoint = 0.... and decypher the data based on the end point... e.g. the port.
//}
#endregion
//Verify Timing lines.
//Lines should have a startTime equal the Uptime of the stream
//Lines should have a stopTime equal to the EndTime of the stream.
}
//Top level stream control line (Should only be added if Aggregate Control of the stream is allowed.
//sdp.Add(new Sdp.SessionDescriptionLine(controlLineBase));
return sdp;
}
