/* * sipcon.h * * Session Initiation Protocol connection. * * Open Phone Abstraction Library (OPAL) * Formally known as the Open H323 project. * * Copyright (c) 2001 Equivalence Pty. Ltd. * * The contents of this file are subject to the Mozilla Public License * Version 1.0 (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.mozilla.org/MPL/ * * Software distributed under the License is distributed on an "AS IS" * basis, WITHOUT WARRANTY OF ANY KIND, either express or implied. See * the License for the specific language governing rights and limitations * under the License. * * The Original Code is Open Phone Abstraction Library. * * The Initial Developer of the Original Code is Equivalence Pty. Ltd. * * Contributor(s): ______________________________________. * * $Revision: 22097 $ * $Author: rjongbloed $ * $Date: 2009-02-21 04:55:10 +0000 (Sat, 21 Feb 2009) $ */ #ifndef OPAL_SIP_SIPCON_H #define OPAL_SIP_SIPCON_H #ifdef P_USE_PRAGMA #pragma interface #endif #include #if OPAL_SIP #include #include #include #include #if OPAL_VIDEO #include // for OpalPCSSConnection #include // for OpalVideoUpdatePicture command #endif class OpalCall; class SIPEndPoint; #define SIP_HEADER_PREFIX "SIP-Header:" #define SIP_HEADER_REPLACES SIP_HEADER_PREFIX"Replaces" ///////////////////////////////////////////////////////////////////////// /**Session Initiation Protocol connection. */ //class SIPConnection : public OpalConnection class SIPConnection : public OpalRTPConnection { PCLASSINFO(SIPConnection, OpalRTPConnection); public: /**@name Construction */ //@{ /**Create a new connection. */ SIPConnection( OpalCall & call, ///< Owner call for connection SIPEndPoint & endpoint, ///< Owner endpoint for connection const PString & token, ///< token to identify the connection const SIPURL & address, ///< Destination address for outgoing call OpalTransport * transport, ///< Transport INVITE came in on unsigned int options = 0, ///< Connection options OpalConnection::StringOptions * stringOptions = NULL ///< complex string options ); /**Destroy connection. */ ~SIPConnection(); //@} /**@name Overrides from OpalConnection */ //@{ /**Get indication of connection being to a "network". This indicates the if the connection may be regarded as a "network" connection. The distinction is about if there is a concept of a "remote" party being connected to and is best described by example: sip, h323, iax and pstn are all "network" connections as they connect to something "remote". While pc, pots and ivr are not as the entity being connected to is intrinsically local. */ virtual bool IsNetworkConnection() const { return true; } /**Get this connections protocol prefix for URLs. */ virtual PString GetPrefixName() const; /**Get the protocol-specific unique identifier for this connection. */ virtual PString GetIdentifier() const; /**Get the remote party address as URL. This will return the "best guess" at an address to use in a to call the user again later. Note that under some circumstances this may be different to the value GetRemotePartyAddress() value returns. In particular if a gatekeeper is involved. */ virtual PString GetRemotePartyURL() const; /**Start an outgoing connection. This function will initiate the connection to the remote entity, for example in H.323 it sends a SETUP, in SIP it sends an INVITE etc. The default behaviour is . */ virtual PBoolean SetUpConnection(); /**Get the destination address of an incoming connection. This will, for example, collect a phone number from a POTS line, or get the fields from the H.225 SETUP pdu in a H.323 connection. The default behaviour for sip returns the request URI in the INVITE. */ virtual PString GetDestinationAddress(); /**Get the fulll URL being indicated by the remote for incoming calls. This may not have any relation to the local name of the endpoint. The default behaviour returns GetDestinationAddress() normalised to a URL. The remote may provide a full URL, if it does not then the prefix for the endpoint is prepended to the destination address. */ virtual PString GetCalledPartyURL(); /**Get alerting type information of an incoming call. The type of "distinctive ringing" for the call. The string is protocol dependent, so the caller would need to be aware of the type of call being made. Some protocols may ignore the field completely. For SIP this corresponds to the string contained in the "Alert-Info" header field of the INVITE. This is typically a URI for the ring file. For H.323 this must be a string representation of an integer from 0 to 7 which will be contained in the Q.931 SIGNAL (0x34) Information Element. Default behaviour returns an empty string. */ virtual PString GetAlertingType() const; /**Set alerting type information for outgoing call. The type of "distinctive ringing" for the call. The string is protocol dependent, so the caller would need to be aware of the type of call being made. Some protocols may ignore the field completely. For SIP this corresponds to the string contained in the "Alert-Info" header field of the INVITE. This is typically a URI for the ring file. For H.323 this must be a string representation of an integer from 0 to 7 which will be contained in the Q.931 SIGNAL (0x34) Information Element. Default behaviour returns false. */ virtual bool SetAlertingType(const PString & info); /**Initiate the transfer of an existing call (connection) to a new remote party. If remoteParty is a valid call token, then the remote party is transferred to that party (consultation transfer) and both calls are cleared. */ virtual bool TransferConnection( const PString & remoteParty ///< Remote party to transfer the existing call to ); /**Put the current connection on hold, suspending all media streams. */ virtual bool HoldConnection(); /**Retrieve the current connection from hold, activating all media * streams. */ virtual bool RetrieveConnection(); /**Return PTrue if the current connection is on hold. */ virtual PBoolean IsConnectionOnHold(); /**Indicate to remote endpoint an alert is in progress. If this is an incoming connection and the AnswerCallResponse is in a AnswerCallDeferred or AnswerCallPending state, then this function is used to indicate to that endpoint that an alert is in progress. This is usually due to another connection which is in the call (the B party) has received an OnAlerting() indicating that its remote endpoint is "ringing". The default behaviour does nothing. */ virtual PBoolean SetAlerting( const PString & calleeName, ///< Name of endpoint being alerted. PBoolean withMedia ); /**Indicate to remote endpoint we are connected. The default behaviour does nothing. */ virtual PBoolean SetConnected(); /**Get the data formats this endpoint is capable of operating in. */ virtual OpalMediaFormatList GetMediaFormats() const; /**Open source or sink media stream for session. */ virtual OpalMediaStreamPtr OpenMediaStream( const OpalMediaFormat & mediaFormat, ///< Media format to open unsigned sessionID, ///< Session to start stream on bool isSource ///< Stream is a source/sink ); /**Request close of a specific media stream. Note that this is usually asymchronous, the OnClosedMediaStream() function is called when the stream is really closed. */ virtual bool CloseMediaStream( OpalMediaStream & stream ///< Stream to close ); /**Clean up the termination of the connection. This function can do any internal cleaning up and waiting on background threads that may be using the connection object. Note that there is not a one to one relationship with the OnEstablishedConnection() function. This function may be called without that function being called. For example if SetUpConnection() was used but the call never completed. Classes that override this function should make sure they call the ancestor version for correct operation. An application will not typically call this function as it is used by the OpalManager during a release of the connection. The default behaviour calls the OpalEndPoint function of the same name. */ virtual void OnReleased(); /**Forward incoming connection to the specified address. This would typically be called from within the OnIncomingConnection() function when an application wishes to redirect an unwanted incoming call. The return value is PTrue if the call is to be forwarded, PFalse otherwise. Note that if the call is forwarded, the current connection is cleared with the ended call code set to EndedByCallForwarded. */ virtual PBoolean ForwardCall( const PString & forwardParty ///< Party to forward call to. ); /**Get the real user input indication transmission mode. This will return the user input mode that will actually be used for transmissions. It will be the value of GetSendUserInputMode() provided the remote endpoint is capable of that mode. */ virtual SendUserInputModes GetRealSendUserInputMode() const; /**Send a user input indication to the remote endpoint. This sends DTMF emulation user input. If something more sophisticated than the simple tones that can be sent using the SendUserInput() function. A duration of zero indicates that no duration is to be indicated. A non-zero logical channel indicates that the tone is to be syncronised with the logical channel at the rtpTimestamp value specified. The tone parameter must be one of "0123456789#*ABCD!" where '!' indicates a hook flash. If tone is a ' ' character then a signalUpdate PDU is sent that updates the last tone indication sent. See the H.245 specifcation for more details on this. The default behaviour sends the tone using RFC2833. */ PBoolean SendUserInputTone(char tone, unsigned duration); /**Callback from the RTP session for statistics monitoring. This is called every so many packets on the transmitter and receiver threads of the RTP session indicating that the statistics have been updated. The default behaviour does nothing. */ virtual void OnRTPStatistics( const RTP_Session & session ///< Session with statistics ) const; //@} /**@name Protocol handling functions */ //@{ /**Handle the fail of a transaction we initiated. */ virtual void OnTransactionFailed( SIPTransaction & transaction ); /**Handle an incoming SIP PDU that has been full decoded */ virtual void OnReceivedPDU(SIP_PDU & pdu); /**Handle an incoming INVITE request */ virtual void OnReceivedINVITE(SIP_PDU & pdu); /**Handle an incoming Re-INVITE request */ virtual void OnReceivedReINVITE(SIP_PDU & pdu); /**Handle an incoming ACK PDU */ virtual void OnReceivedACK(SIP_PDU & pdu); /**Handle an incoming OPTIONS PDU */ virtual void OnReceivedOPTIONS(SIP_PDU & pdu); /**Handle an incoming NOTIFY PDU */ virtual void OnReceivedNOTIFY(SIP_PDU & pdu); /**Handle an incoming REFER PDU */ virtual void OnReceivedREFER(SIP_PDU & pdu); /**Handle an incoming INFO PDU */ virtual void OnReceivedINFO(SIP_PDU & pdu); /**Handle an incoming PING PDU */ virtual void OnReceivedPING(SIP_PDU & pdu); /**Handle an incoming BYE PDU */ virtual void OnReceivedBYE(SIP_PDU & pdu); /**Handle an incoming CANCEL PDU */ virtual void OnReceivedCANCEL(SIP_PDU & pdu); /**Handle an incoming response PDU to our INVITE. Note this is called before th ACK is sent and thus should do as little as possible. All the hard work (SDP processing etc) should be in the usual OnReceivedResponse(). */ virtual void OnReceivedResponseToINVITE( SIPTransaction & transaction, SIP_PDU & response ); /**Handle an incoming response PDU. */ virtual void OnReceivedResponse( SIPTransaction & transaction, SIP_PDU & response ); /**Handle an incoming Trying response PDU */ virtual void OnReceivedTrying(SIP_PDU & pdu); /**Handle an incoming Ringing response PDU */ virtual void OnReceivedRinging(SIP_PDU & pdu); /**Handle an incoming Session Progress response PDU */ virtual void OnReceivedSessionProgress(SIP_PDU & pdu); /**Handle an incoming Proxy Authentication Required response PDU Returns: PTrue if handled, if PFalse is returned connection is released. */ virtual PBoolean OnReceivedAuthenticationRequired( SIPTransaction & transaction, SIP_PDU & response ); /**Handle an incoming redirect response PDU */ virtual void OnReceivedRedirection(SIP_PDU & pdu); /**Handle an incoming OK response PDU. This actually gets any PDU of the class 2xx not just 200. */ virtual void OnReceivedOK( SIPTransaction & transaction, SIP_PDU & response ); /**Handle a sending INVITE request */ virtual void OnCreatingINVITE(SIP_PDU & pdu); /**Send a "200 OK" response for the received INVITE message. */ virtual PBoolean SendInviteOK(const SDPSessionDescription & sdp); /**Send a response for the received INVITE message. */ virtual PBoolean SendInviteResponse( SIP_PDU::StatusCodes code, const char * contact = NULL, const char * extra = NULL, const SDPSessionDescription * sdp = NULL ); //@} OpalTransportAddress GetDefaultSDPConnectAddress(WORD port = 0) const; OpalTransport & GetTransport() const { return *transport; } SIPEndPoint & GetEndPoint() const { return endpoint; } SIPDialogContext & GetDialog() { return m_dialog; } const SIPDialogContext & GetDialog() const { return m_dialog; } SIPAuthentication * GetAuthenticator() const { return authentication; } #if OPAL_VIDEO /**Call when SIP INFO of type application/media_control+xml is received. Return PFalse if default reponse of Failure_UnsupportedMediaType is to be returned */ virtual PBoolean OnMediaControlXML(SIP_PDU & pdu); #endif virtual void OnMediaCommand(OpalMediaCommand & note, INT extra); virtual void OnStartTransaction(SIPTransaction & transaction); virtual void OnReceivedMESSAGE(SIP_PDU & pdu); virtual void OnMessageReceived(const SIPURL & from, const SIP_PDU & pdu); protected: PDECLARE_NOTIFIER(PTimer, SIPConnection, OnInviteResponseRetry); PDECLARE_NOTIFIER(PTimer, SIPConnection, OnAckTimeout); virtual RTP_UDP *OnUseRTPSession( const unsigned rtpSessionId, const OpalMediaType & mediaType, const OpalTransportAddress & mediaAddress, OpalTransportAddress & localAddress ); virtual bool OnSendSDP( bool isAnswerSDP, OpalRTPSessionManager & rtpSessions, SDPSessionDescription & sdpOut ); virtual bool OfferSDPMediaDescription( const OpalMediaType & mediaType, unsigned sessionID, OpalRTPSessionManager & rtpSessions, SDPSessionDescription & sdpOut ); virtual bool AnswerSDPMediaDescription( const SDPSessionDescription & sdpIn, unsigned sessionIndex, SDPSessionDescription & sdpOut ); virtual void OnReceivedSDP( SIP_PDU & pdu ); virtual bool OnReceivedSDPMediaDescription( SDPSessionDescription & sdp, unsigned sessionId ); friend class SIPInvite; static PBoolean WriteINVITE(OpalTransport & transport, void * param); bool WriteINVITE(OpalTransport & transport); OpalTransport * CreateTransport(const OpalTransportAddress & address, PBoolean isLocalAddress = PFalse); void UpdateRemoteAddresses(); void NotifyDialogState( SIPDialogNotification::States state, SIPDialogNotification::Events eventType = SIPDialogNotification::NoEvent, unsigned eventCode = 0 ); // Member variables SIPEndPoint & endpoint; OpalTransport * transport; bool deleteTransport; enum HoldState { eHoldOff, eRetrieveInProgress, // Order is important! eHoldOn, eHoldInProgress }; HoldState m_holdToRemote; bool m_holdFromRemote; PString forwardParty; SIP_PDU * originalInvite; PTime originalInviteTime; bool needReINVITE; SIPDialogContext m_dialog; OpalGloballyUniqueID m_dialogNotifyId; int m_appearanceCode; PString m_alertInfo; SIPAuthentication * authentication; std::map m_lastRxCSeq; PTimer ackTimer; PTimer ackRetry; SIP_PDU ackPacket; bool ackReceived; PSafePtr referTransaction; PSafeList forkedInvitations; // Not for re-INVITE enum { ReleaseWithBYE, ReleaseWithCANCEL, ReleaseWithResponse, ReleaseWithNothing, } releaseMethod; OpalMediaFormatList remoteFormatList; PString GetLocalPartyURL() const; protected: PTimer sessionTimer; public: PDECLARE_NOTIFIER(PTimer, SIPConnection, OnSessionTimeout); }; /**This class is for encpsulating the IETF Real Time Protocol interface. */ class SIP_RTP_Session : public RTP_UserData { PCLASSINFO(SIP_RTP_Session, RTP_UserData); /**@name Construction */ //@{ /**Create a new channel. */ SIP_RTP_Session( const SIPConnection & connection ///< Owner of the RTP session ); //@} /**@name Overrides from RTP_UserData */ //@{ /**Callback from the RTP session for transmit statistics monitoring. This is called every RTP_Session::senderReportInterval packets on the transmitter indicating that the statistics have been updated. The default behaviour calls H323Connection::OnRTPStatistics(). */ virtual void OnTxStatistics( const RTP_Session & session ///< Session with statistics ) const; /**Callback from the RTP session for receive statistics monitoring. This is called every RTP_Session::receiverReportInterval packets on the receiver indicating that the statistics have been updated. The default behaviour calls H323Connection::OnRTPStatistics(). */ virtual void OnRxStatistics( const RTP_Session & session ///< Session with statistics ) const; #if OPAL_VIDEO /**Callback from the RTP session after an IntraFrameRequest is receieved. The default behaviour executes an OpalVideoUpdatePicture command on the connection's source video stream if it exists. */ virtual void OnRxIntraFrameRequest( const RTP_Session & session ///< Session with statistics ) const; /**Callback from the RTP session after an IntraFrameRequest is sent. The default behaviour does nothing. */ virtual void OnTxIntraFrameRequest( const RTP_Session & session ///< Session with statistics ) const; #endif //@} virtual void SessionFailing(RTP_Session & /*session*/); protected: const SIPConnection & connection; /// Owner of the RTP session }; #endif // OPAL_SIP #endif // OPAL_SIP_SIPCON_H // End of File ///////////////////////////////////////////////////////////////