private void ProcessMediaOffer(SIP_Dialog dialog, SIP_ServerTransaction transaction, RTP_MultimediaSession rtpMultimediaSession, SDP_Message offer, SDP_Message localSDP)
{
if (dialog == null)
{
throw new ArgumentNullException("dialog");
}
if (transaction == null)
{
throw new ArgumentNullException("transaction");
}
if (rtpMultimediaSession == null)
{
throw new ArgumentNullException("rtpMultimediaSession");
}
if (offer == null)
{
throw new ArgumentNullException("offer");
}
if (localSDP == null)
{
throw new ArgumentNullException("localSDP");
}
try
{
#region SDP basic validation
// Version field must exist.
if (offer.Version == null)
{
transaction.SendResponse(m_pStack.CreateResponse(SIP_ResponseCodes.x500_Server_Internal_Error + ": Invalid SDP answer: Required 'v'(Protocol Version) field is missing.", transaction.Request));
return;
}
// Origin field must exist.
if (offer.Origin == null)
{
transaction.SendResponse(m_pStack.CreateResponse(SIP_ResponseCodes.x500_Server_Internal_Error + ": Invalid SDP answer: Required 'o'(Origin) field is missing.", transaction.Request));
return;
}
// Session Name field.
// Check That global 'c' connection attribute exists or otherwise each enabled media stream must contain one.
if (offer.Connection == null)
{
for (int i = 0; i < offer.MediaDescriptions.Count; i++)
{
if (offer.MediaDescriptions[i].Connection == null)
{
transaction.SendResponse(m_pStack.CreateResponse(SIP_ResponseCodes.x500_Server_Internal_Error + ": Invalid SDP answer: Global or per media stream no: " + i + " 'c'(Connection) attribute is missing.", transaction.Request));
return;
}
}
}
#endregion
// Re-INVITE media streams count must be >= current SDP streams.
if (localSDP.MediaDescriptions.Count > offer.MediaDescriptions.Count)
{
transaction.SendResponse(m_pStack.CreateResponse(SIP_ResponseCodes.x500_Server_Internal_Error + ": re-INVITE SDP offer media stream count must be >= current session stream count.", transaction.Request));
return;
}
bool audioAccepted = false;
// Process media streams info.
for (int i = 0; i < offer.MediaDescriptions.Count; i++)
{
SDP_MediaDescription offerMedia = offer.MediaDescriptions[i];
SDP_MediaDescription answerMedia = (localSDP.MediaDescriptions.Count > i ? localSDP.MediaDescriptions[i] : null);
// Disabled stream.
if (offerMedia.Port == 0)
{
// Remote-party offered new disabled stream.
if (answerMedia == null)
{
// Just copy offer media stream data to answer and set port to zero.
localSDP.MediaDescriptions.Add(offerMedia);
localSDP.MediaDescriptions[i].Port = 0;
}
// Existing disabled stream or remote party disabled it.
else
{
answerMedia.Port = 0;
#region Cleanup active RTP stream and it's resources, if it exists
// Dispose existing RTP session.
if (answerMedia.Tags.ContainsKey("rtp_session"))
{
((RTP_Session)offerMedia.Tags["rtp_session"]).Dispose();
answerMedia.Tags.Remove("rtp_session");
}
// Release UPnPports if any.
if (answerMedia.Tags.ContainsKey("upnp_rtp_map"))
{
try
{
m_pUPnP.DeletePortMapping((UPnP_NAT_Map)answerMedia.Tags["upnp_rtp_map"]);
}
catch
{
}
answerMedia.Tags.Remove("upnp_rtp_map");
}
if (answerMedia.Tags.ContainsKey("upnp_rtcp_map"))
{
try
{
m_pUPnP.DeletePortMapping((UPnP_NAT_Map)answerMedia.Tags["upnp_rtcp_map"]);
}
catch
{
}
answerMedia.Tags.Remove("upnp_rtcp_map");
}
#endregion
}
}
// Remote-party wants to communicate with this stream.
else
{
// See if we can support this stream.
if (!audioAccepted && CanSupportMedia(offerMedia))
{
// New stream.
if (answerMedia == null)
{
answerMedia = new SDP_MediaDescription(SDP_MediaTypes.audio, 0, 2, "RTP/AVP", null);
localSDP.MediaDescriptions.Add(answerMedia);
}
#region Build audio codec map with codecs which we support
Dictionary<int, AudioCodec> audioCodecs = GetOurSupportedAudioCodecs(offerMedia);
answerMedia.MediaFormats.Clear();
answerMedia.Attributes.Clear();
foreach (KeyValuePair<int, AudioCodec> entry in audioCodecs)
{
answerMedia.Attributes.Add(new SDP_Attribute("rtpmap", entry.Key + " " + entry.Value.Name + "/" + entry.Value.CompressedAudioFormat.SamplesPerSecond));
answerMedia.MediaFormats.Add(entry.Key.ToString());
}
answerMedia.Attributes.Add(new SDP_Attribute("ptime", "20"));
answerMedia.Tags["audio_codecs"] = audioCodecs;
#endregion
#region Create/modify RTP session
// RTP session doesn't exist, create it.
if (!answerMedia.Tags.ContainsKey("rtp_session"))
{
RTP_Session rtpSess = CreateRtpSession(rtpMultimediaSession);
// RTP session creation failed,disable this stream.
if (rtpSess == null)
{
answerMedia.Port = 0;
break;
}
answerMedia.Tags.Add("rtp_session", rtpSess);
rtpSess.NewReceiveStream += delegate(object s, RTP_ReceiveStreamEventArgs e)
{
if (answerMedia.Tags.ContainsKey("rtp_audio_out"))
{
((AudioOut_RTP)answerMedia.Tags["rtp_audio_out"]).Dispose();
}
AudioOut_RTP audioOut = new AudioOut_RTP(m_pAudioOutDevice, e.Stream, audioCodecs);
audioOut.Start();
answerMedia.Tags["rtp_audio_out"] = audioOut;
};
// NAT
if (!HandleNAT(answerMedia, rtpSess))
{
// NAT handling failed,disable this stream.
answerMedia.Port = 0;
break;
}
}
RTP_StreamMode offerStreamMode = GetRtpStreamMode(offer, offerMedia);
if (offerStreamMode == RTP_StreamMode.Inactive)
{
answerMedia.SetStreamMode("inactive");
}
else if (offerStreamMode == RTP_StreamMode.Receive)
{
answerMedia.SetStreamMode("sendonly");
}
else if (offerStreamMode == RTP_StreamMode.Send)
{
answerMedia.SetStreamMode("recvonly");
}
else if (offerStreamMode == RTP_StreamMode.SendReceive)
{
answerMedia.SetStreamMode("sendrecv");
}
RTP_Session rtpSession = (RTP_Session)answerMedia.Tags["rtp_session"];
rtpSession.Payload = Convert.ToInt32(answerMedia.MediaFormats[0]);
rtpSession.StreamMode = GetRtpStreamMode(localSDP, answerMedia);
rtpSession.RemoveTargets();
if (GetSdpHost(offer, offerMedia) != "0.0.0.0")
{
rtpSession.AddTarget(GetRtpTarget(offer, offerMedia));
}
rtpSession.Start();
#endregion
#region Create/modify audio-in source
if (!answerMedia.Tags.ContainsKey("rtp_audio_in"))
{
AudioIn_RTP rtpAudioIn = new AudioIn_RTP(m_pAudioInDevice, 20, audioCodecs, rtpSession.CreateSendStream());
rtpAudioIn.Start();
answerMedia.Tags.Add("rtp_audio_in", rtpAudioIn);
}
else
{
((AudioIn_RTP)answerMedia.Tags["rtp_audio_in"]).AudioCodecs = audioCodecs;
}
#endregion
audioAccepted = true;
}
// We don't accept this stream, so disable it.
else
{
// Just copy offer media stream data to answer and set port to zero.
// Delete exisiting media stream.
if (answerMedia != null)
{
localSDP.MediaDescriptions.RemoveAt(i);
}
localSDP.MediaDescriptions.Add(offerMedia);
localSDP.MediaDescriptions[i].Port = 0;
}
}
}
#region Create and send 2xx response
SIP_Response response = m_pStack.CreateResponse(SIP_ResponseCodes.x200_Ok, transaction.Request, transaction.Flow);
//response.Contact = SIP stack will allocate it as needed;
response.ContentType = "application/sdp";
response.Data = localSDP.ToByte();
transaction.SendResponse(response);
// Start retransmitting 2xx response, while ACK receives.
Handle2xx(dialog, transaction);
// REMOVE ME: 27.11.2010
// Start retransmitting 2xx response, while ACK receives.
//m_pInvite2xxMgr.Add(dialog,transaction);
#endregion
}
catch (Exception x)
{
transaction.SendResponse(m_pStack.CreateResponse(SIP_ResponseCodes.x500_Server_Internal_Error + ": " + x.Message, transaction.Request));
}
}
private void Call(SIP_t_NameAddress from, SIP_t_NameAddress to)
{
if (from == null)
{
throw new ArgumentNullException("from");
}
if (to == null)
{
throw new ArgumentNullException("to");
}
#region Setup RTP session
RTP_MultimediaSession rtpMultimediaSession = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
RTP_Session rtpSession = CreateRtpSession(rtpMultimediaSession);
// Port search failed.
if (rtpSession == null)
{
throw new Exception("Calling not possible, RTP session failed to allocate IP end points.");
}
if (m_IsDebug)
{
wfrm_RTP_Debug rtpDebug = new wfrm_RTP_Debug(rtpMultimediaSession);
rtpDebug.Show();
}
#endregion
#region Create SDP offer
SDP_Message sdpOffer = new SDP_Message();
sdpOffer.Version = "0";
sdpOffer.Origin = new SDP_Origin("-", sdpOffer.GetHashCode(), 1, "IN", "IP4", System.Net.Dns.GetHostAddresses("")[0].ToString());
sdpOffer.SessionName = "SIP Call";
sdpOffer.Times.Add(new SDP_Time(0, 0));
#region Add 1 audio stream
SDP_MediaDescription mediaStream = new SDP_MediaDescription(SDP_MediaTypes.audio, 0, 1, "RTP/AVP", null);
rtpSession.NewReceiveStream += delegate(object s, RTP_ReceiveStreamEventArgs e)
{
AudioOut_RTP audioOut = new AudioOut_RTP(m_pAudioOutDevice, e.Stream, m_pAudioCodecs);
audioOut.Start();
mediaStream.Tags["rtp_audio_out"] = audioOut;
};
if (!HandleNAT(mediaStream, rtpSession))
{
throw new Exception("Calling not possible, because of NAT or firewall restrictions.");
}
foreach (KeyValuePair<int, AudioCodec> entry in m_pAudioCodecs)
{
mediaStream.Attributes.Add(new SDP_Attribute("rtpmap", entry.Key + " " + entry.Value.Name + "/" + entry.Value.CompressedAudioFormat.SamplesPerSecond));
mediaStream.MediaFormats.Add(entry.Key.ToString());
}
mediaStream.Attributes.Add(new SDP_Attribute("ptime", "20"));
mediaStream.Attributes.Add(new SDP_Attribute("sendrecv", ""));
mediaStream.Tags["rtp_session"] = rtpSession;
mediaStream.Tags["audio_codecs"] = m_pAudioCodecs;
sdpOffer.MediaDescriptions.Add(mediaStream);
#endregion
#endregion
// Create INVITE request.
SIP_Request invite = m_pStack.CreateRequest(SIP_Methods.INVITE, to, from);
invite.ContentType = "application/sdp";
invite.Data = sdpOffer.ToByte();
SIP_RequestSender sender = m_pStack.CreateRequestSender(invite);
// Create call.
m_pCall = new SIP_Call(m_pStack, sender, rtpMultimediaSession);
m_pCall.LocalSDP = sdpOffer;
m_pCall.StateChanged += new EventHandler(m_pCall_StateChanged);
bool finalResponseSeen = false;
List<SIP_Dialog_Invite> earlyDialogs = new List<SIP_Dialog_Invite>();
sender.ResponseReceived += delegate(object s, SIP_ResponseReceivedEventArgs e)
{
// Skip 2xx retransmited response.
if (finalResponseSeen)
{
return;
}
if (e.Response.StatusCode >= 200)
{
finalResponseSeen = true;
}
try
{
#region Provisional
if (e.Response.StatusCodeType == SIP_StatusCodeType.Provisional)
{
/* RFC 3261 13.2.2.1.
Zero, one or multiple provisional responses may arrive before one or
more final responses are received. Provisional responses for an
INVITE request can create "early dialogs". If a provisional response
has a tag in the To field, and if the dialog ID of the response does
not match an existing dialog, one is constructed using the procedures
defined in Section 12.1.2.
*/
if (e.Response.StatusCode > 100 && e.Response.To.Tag != null)
{
earlyDialogs.Add((SIP_Dialog_Invite)e.GetOrCreateDialog);
}
// 180_Ringing.
if (e.Response.StatusCode == 180)
{
//m_pPlayer.Play(ResManager.GetStream("ringing.wav"), 10);
// We need BeginInvoke here, otherwise we block client transaction.
m_pStatusBar.BeginInvoke(new MethodInvoker(delegate()
{
m_pStatusBar.Items[0].Text = "Ringing";
}));
}
}
#endregion
#region Success
else if (e.Response.StatusCodeType == SIP_StatusCodeType.Success)
{
SIP_Dialog dialog = e.GetOrCreateDialog;
/* Exit all all other dialogs created by this call (due to forking).
That is not defined in RFC but, since UAC can send BYE to early and confirmed dialogs,
all this is 100% valid.
*/
foreach (SIP_Dialog_Invite d in earlyDialogs.ToArray())
{
if (!d.Equals(dialog))
{
d.Terminate("Another forking leg accepted.", true);
}
}
m_pCall.InitCalling(dialog, sdpOffer);
// Remote-party provided SDP.
if (e.Response.ContentType != null && e.Response.ContentType.ToLower().IndexOf("application/sdp") > -1)
{
try
{
// SDP offer. We sent offerless INVITE, we need to send SDP answer in ACK request.'
if (e.ClientTransaction.Request.ContentType == null || e.ClientTransaction.Request.ContentType.ToLower().IndexOf("application/sdp") == -1)
{
// Currently we never do it, so it never happens. This is place holder, if we ever support it.
}
// SDP answer to our offer.
else
{
// This method takes care of ACK sending and 2xx response retransmission ACK sending.
HandleAck(m_pCall.Dialog, e.ClientTransaction);
ProcessMediaAnswer(m_pCall, m_pCall.LocalSDP, SDP_Message.Parse(Encoding.UTF8.GetString(e.Response.Data)));
}
}
catch
{
m_pCall.Terminate("SDP answer parsing/processing failed.");
}
}
else
{
// If we provided SDP offer, there must be SDP answer.
if (e.ClientTransaction.Request.ContentType != null && e.ClientTransaction.Request.ContentType.ToLower().IndexOf("application/sdp") > -1)
{
m_pCall.Terminate("Invalid 2xx response, required SDP answer is missing.");
}
}
// Stop ringing.
m_pPlayer.Stop();
}
#endregion
#region Failure
else
{
/* RFC 3261 13.2.2.3.
All early dialogs are considered terminated upon reception of the non-2xx final response.
*/
foreach (SIP_Dialog_Invite dialog in earlyDialogs.ToArray())
{
dialog.Terminate("All early dialogs are considered terminated upon reception of the non-2xx final response. (RFC 3261 13.2.2.3)", false);
}
// We need BeginInvoke here, otherwise we block client transaction while message box open.
if (m_pCall.State != SIP_CallState.Terminating)
{
this.BeginInvoke(new MethodInvoker(delegate()
{
m_pConnect.Image = global::PowerSDR.Properties.Resources.call;
connected = false;
MessageBox.Show("Calling failed: " + e.Response.StatusCode_ReasonPhrase, "Error:", MessageBoxButtons.OK, MessageBoxIcon.Error);
}));
}
// We need BeginInvoke here, otherwise we block client transaction.
m_pStatusBar.BeginInvoke(new MethodInvoker(delegate()
{
m_pStatusBar.Items[0].Text = "";
}));
// Stop calling or ringing.
m_pPlayer.Stop();
// Terminate call.
m_pCall.Terminate("Remote party rejected a call.", false);
}
#endregion
}
catch (Exception x)
{
// We need BeginInvoke here, otherwise we block client transaction while message box open.
this.BeginInvoke(new MethodInvoker(delegate()
{
MessageBox.Show("Error: " + x.Message, "Error:", MessageBoxButtons.OK, MessageBoxIcon.Error);
}));
}
};
m_pStatusBar.Items[0].Text = "Calling";
m_pStatusBar.Items[1].Text = "00:00:00";
//m_pPlayer.Play(ResManager.GetStream("calling.wav"), 10);
// Start calling.
sender.Start();
}
/// <summary>
/// This method is called when RTP session gets new receive stream.
/// </summary>
/// <param name="sender">Sender.</param>
/// <param name="e">Event data.</param>
private void m_pRtpSession_NewReceiveStream(object sender, RTP_ReceiveStreamEventArgs e)
{
d_RecieveStream = Dispatcher.BeginInvoke(new Action(delegate()
{
/*wfrm_Receive frm = new wfrm_Receive(e.Stream, m_pAudioCodecs);
frm.Show();*/
var selectedOutDevice = cbAudioOutDevices.SelectedItem as AudioOutDevice;
m_pAudioOut = new AudioOut_RTP(selectedOutDevice, e.Stream, m_pAudioCodecs);
m_pAudioOut.Start();
}));
}