public void TestCustom()
{
byte[] random = new byte[5];
Media.Utility.Random.NextBytes(random);
// Create SDP offer (Step 1).
string originatorAndSession = String.Format("{0} {1} {2} {3} {4} {5}", "-", BitConverter.ToString(random).Replace("-", string.Empty), "0", "IN", "IP4", "10.1.1.2");
using (var sdp = new Media.Sdp.SessionDescription(0, originatorAndSession, "sipsorcery"))
{
sdp.Add(new Media.Sdp.SessionDescriptionLine("c=IN IP4 10.1.1.2"), false);
var audioAnnouncement = new Media.Sdp.MediaDescription(Media.Sdp.MediaType.audio, 0, "SDP_TRANSPORT", 0);
sdp.Add(audioAnnouncement, false);
// Set up the RTP channel (Step 2).
using (var _rtpAudioClient = Media.Rtp.RtpClient.FromSessionDescription(sdp))
{
var _audioRTPTransportContext = _rtpAudioClient.GetTransportContexts().FirstOrDefault(); // The tranpsort context is null at this point.
System.Diagnostics.Debug.Assert(_audioRTPTransportContext != null, "Cannot find the context");
//System.Diagnostics.Debug.Assert(_audioRTPTransportContext.IsActive == false, "Found a active context");
//_rtpAudioClient.Activate();
System.Diagnostics.Debug.Assert(_audioRTPTransportContext.IsActive == true, "Found a active context");
}
}
}
public TestFramework()
{
// Create a receiving socket.
_receiving = new System.Net.Sockets.Socket(_rtspServer.AddressFamily, System.Net.Sockets.SocketType.Stream, System.Net.Sockets.ProtocolType.Tcp);
// Connect to the server.
System.IAsyncResult connectResult = null;
connectResult = _receiving.BeginConnect(_rtspServer, new System.AsyncCallback((iar) =>
{
try { _receiving.EndConnect(iar); }
catch { }
}), null);
// Get the sender socket to be used by the "server".
_sender = _listenSocket.Accept();
// RtspClient default size
byte[] buffer = new byte[8192];
_client = new Media.Rtp.RtpClient(new Media.Common.MemorySegment(buffer, Media.Rtsp.RtspMessage.MaximumLength, buffer.Length - Media.Rtsp.RtspMessage.MaximumLength));
_client.OutOfBandData += ProcessInterleaveData;
_client.RtpPacketReceieved += ProcessRtpPacket;
Media.Sdp.MediaDescription md = new Media.Sdp.MediaDescription(Media.Sdp.MediaType.video, 999, "H.264", 0);
Media.Rtp.RtpClient.TransportContext tc = new Media.Rtp.RtpClient.TransportContext(0, 1,
Media.RFC3550.Random32(9876), md, false, _senderSSRC);
// Create a Duplexed reciever using the RtspClient socket.
tc.Initialize(_receiving);
_client.TryAddContext(tc);
}
public void TestCustom()
{
byte[] random = new byte[5];
Media.Utility.Random.NextBytes(random);
// Create SDP offer (Step 1).
string originatorAndSession = String.Format("{0} {1} {2} {3} {4} {5}", "-", BitConverter.ToString(random).Replace("-", string.Empty), "0", "IN", "IP4", "10.1.1.2");
using (var sdp = new Media.Sdp.SessionDescription(0, originatorAndSession, "sipsorcery"))
{
sdp.Add(new Media.Sdp.SessionDescriptionLine("c=IN IP4 10.1.1.2"), false);
var audioAnnouncement = new Media.Sdp.MediaDescription(Media.Sdp.MediaType.audio, 0, "SDP_TRANSPORT", 0);
sdp.Add(audioAnnouncement, false);
// Set up the RTP channel (Step 2).
using (var _rtpAudioClient = Media.Rtp.RtpClient.FromSessionDescription(sdp))
{
var _audioRTPTransportContext = _rtpAudioClient.GetTransportContexts().FirstOrDefault();
System.Diagnostics.Debug.Assert(_audioRTPTransportContext != null, "Cannot find the context");
System.Diagnostics.Debug.Assert(_audioRTPTransportContext.IsActive == false, "Found an Active context");
//Activate the RtpClient
_rtpAudioClient.Activate();
System.Diagnostics.Debug.Assert(_audioRTPTransportContext.IsActive == false, "Found an Active context");
System.Diagnostics.Debug.Assert(_rtpAudioClient.IsActive == true, "Did not find an Active RtpClient");
System.Diagnostics.Debug.Assert(_rtpAudioClient.SendReports() == false, "SendReports cannot be true, context is not active.");
}
}
}
public void CreateMediaDesciptionTest()
{
//RtpClient has the following property
//Media.Rtp.RtpClient.AvpProfileIdentifier
//I don't think it should be specified in the SDP Classes but I can figure out something else if desired.
string profile = "RTP/AVP";
Media.Sdp.MediaType mediaType = Media.Sdp.MediaType.audio;
int mediaPort = 15000;
//Iterate all possible byte values (should do a seperate test for the list of values?)
for (int mediaFormat = 0; mediaFormat <= 999; ++mediaFormat)
{
//Create a MediaDescription
using (var mediaDescription = new Media.Sdp.MediaDescription(mediaType, mediaPort, profile, mediaFormat))
{
System.Diagnostics.Debug.Assert(mediaDescription.MediaProtocol == profile, "Did not find MediaProtocol '" + profile + "'");
System.Diagnostics.Debug.Assert(mediaDescription.PayloadTypes.Count() == 1, "Found more then 1 payload type in the PayloadTypes List");
System.Diagnostics.Debug.Assert(mediaDescription.PayloadTypes.First() == mediaFormat, "Did not find correct MediaFormat");
System.Diagnostics.Debug.Assert(mediaDescription.ToString() == string.Format("m={0} {1} RTP/AVP {2}\r\n", mediaType, mediaPort, mediaFormat), "Did not output correct result");
}
}
}
public void ParseMediaDescriptionUnitTest()
{
string testVector = @" m=audio 49230 RTP/AVP 96 97 98
a=rtpmap:96 L8/8000
a=rtpmap:97 L16/8000
a=rtpmap:98 L16/11025/2";
using (var md = new Media.Sdp.MediaDescription(testVector))
{
System.Diagnostics.Debug.Assert(md.Lines.Count() == 4, "MediaDescription must have 4 lines");
//CLR not assert correctly with == ....
//md.MediaDescriptionLine.ToString() == "m=audio 49230 RTP/AVP 96 97 98"
System.Diagnostics.Debug.Assert(md.PayloadTypes.Count() == 3, "Could not read the Payload List");
System.Diagnostics.Debug.Assert(md.PayloadTypes.First() == 96, "Could not read the Payload List");
System.Diagnostics.Debug.Assert(md.PayloadTypes.ToArray()[1] == 97, "Could not read the Payload List");
System.Diagnostics.Debug.Assert(md.PayloadTypes.Last() == 98, "Could not read the Payload List");
System.Diagnostics.Debug.Assert(string.Compare(md.MediaDescriptionLine.ToString(), "m=audio 49230 RTP/AVP 96 97 98\r\n") == 0, "Did not handle Payload List Correct");
}
}
public TestFramework()
{
// Create a receiving socket.
_receiving = new System.Net.Sockets.Socket(_rtspServer.AddressFamily, System.Net.Sockets.SocketType.Stream, System.Net.Sockets.ProtocolType.Tcp);
// Connect to the server.
System.IAsyncResult connectResult = null;
connectResult = _receiving.BeginConnect(_rtspServer, new System.AsyncCallback((iar) =>
{
try { _receiving.EndConnect(iar); }
catch { }
}), null);
// Get the sender socket to be used by the "server".
_sender = _listenSocket.Accept();
// RtspClient default size
byte[] buffer = new byte[8192];
_client = new Media.Rtp.RtpClient(new Media.Common.MemorySegment(buffer, Media.Rtsp.RtspMessage.MaximumLength, buffer.Length - Media.Rtsp.RtspMessage.MaximumLength));
_client.InterleavedData += ProcessInterleaveData;
_client.RtpPacketReceieved += ProcessRtpPacket;
Media.Sdp.MediaDescription md = new Media.Sdp.MediaDescription(Media.Sdp.MediaType.video, 999, "H.264", 0);
Media.Rtp.RtpClient.TransportContext tc = new Media.Rtp.RtpClient.TransportContext(0, 1,
Media.RFC3550.Random32(9876), md, false, _senderSSRC);
// Create a Duplexed reciever using the RtspClient socket.
tc.Initialize(_receiving, _receiving);
_client.TryAddContext(tc);
}
