Migrate to eXosip2

ChangeLog

+Laurielle LEA (27 July 2005) version 0.4
+- Migrate from libeXoSIP 0.9.0 to libeXosip2-1.9.1-pre15 
+(http://www.antisip.com/download/)
+
 Laurielle LEA (21 July 2005) version 0.4
 - Change README.
 

DEPS

 PortAudio: http://portaudio.com/archives/pa_snapshot_v19.tar.gz
 Common C++ 2 1.3.6: http://sourceforge.net/projects/cplusplus/
 ccRTP 1.3.0: http://sourceforge.net/projects/cplusplus/
-libosip 2.2.0: http://savannah.gnu.org/projects/osip/
-libeXoSIP 0.9.0: http://savannah.nongnu.org/projects/exosip/
+libosip 2.2.1: http://savannah.gnu.org/projects/osip/
+libeXosip2-1.9.1-pre14: http://www.antisip.com/download/

config.h.in

@@ -15,8 +15,8 @@
 /* Define to 1 if you have the <dlfcn.h> header file. */
 #undef HAVE_DLFCN_H
 
-/* Define to 1 if you have the <eXosip/eXosip.h> header file. */
-#undef HAVE_EXOSIP_EXOSIP_H
+/* Define to 1 if you have the <eXosip2/eXosip.h> header file. */
+#undef HAVE_EXOSIP2_EXOSIP_H
 
 /* Define to 1 if you have the <inttypes.h> header file. */
 #undef HAVE_INTTYPES_H

configure.ac

@@ -54,7 +54,7 @@ dnl Checks for header files.
 AC_HEADER_STDC
 AC_CHECK_HEADERS( \
 ostream \
-eXosip/eXosip.h \
+eXosip2/eXosip.h \
 portaudio.h \
 )
 
@@ -78,11 +78,11 @@ PKG_CHECK_MODULES(libosip2, libosip2 >= ${LIBOSIP2_MIN_VERSION})
 SFLPHONE_CXXFLAGS="$SFLPHONE_CXXFLAGS $libosip2_CFLAGS"
 SFLPHONE_LIBS="$SFLPHONE_LIBS $libosip2_LIBS"
 
-if test $ac_cv_header_eXosip_eXosip_h = no; then
-   AC_MSG_ERROR([*** missing eXipsip/eXosip.h. You need a working eXosip installation. See http://savannah.nongnu.org/projects/exosip/])
+if test $ac_cv_header_eXosip2_eXosip_h = no; then
+   AC_MSG_ERROR([*** missing eXosip2/eXosip.h. You need a working eXosip2 installation. See http://www.antisip.com/download/])
 fi
-libexosip_LIBS="-leXosip "
-SFLPHONE_LIBS="$SFLPHONE_LIBS $libexosip_LIBS"
+libexosip2_LIBS="-leXosip2 "
+SFLPHONE_LIBS="$SFLPHONE_LIBS $libexosip2_LIBS"
 
 AC_SUBST(LIBQT)
 

src/audio/audiortp.cpp

@@ -189,6 +189,7 @@ AudioRtpRTX::initAudioRtpSession (void)
 			setCancel(cancelImmediate);
 		}
 	}
+	_debug("-----------------------\n");
 }
 
 void

src/gui/qt/qtGUImainwindow.cpp

@@ -707,6 +707,7 @@ QtGUIMainWindow::callIsOnHold(int id, int line, int busyLine)
 int
 QtGUIMainWindow::callIsIncoming (int id, int line, int busyLine)
 {
+	_TabIncomingCalls[line] = -1;
 	changeLineStatePixmap(line, BUSY);
 	putOnHoldBusyLine(busyLine);
 	if (qt_answerCall(id) != 1) {
@@ -1218,6 +1219,7 @@ QtGUIMainWindow::hangupLine (void)
 			Manager::instance().displayErrorText("Hangup call failed !\n");
 		}	
 	} else if ((i = isThereIncomingCall()) > 0){
+		_debug("&&&&&&&&&& i = %d\n", i);
 		// To refuse new incoming call 
 		_debug("Refuse call %d\n", id);
 		if (qt_refuseCall(i)) {

src/gui/qt/qtGUImainwindow.h

@@ -286,7 +286,7 @@ private:
 	inline void setPrevLine(int line) { _prevLine = line; }
 	inline int getPrevLine(void) { return _prevLine; }
 
-	// Array of incoming calls
+	// Array of incoming calls, contains call-id
 	int _TabIncomingCalls[NUMBER_OF_LINES];
 
 	// The current phoneline

src/sipcall.h

@@ -22,13 +22,16 @@
 #ifndef __SIP_CALL_H__
 #define __SIP_CALL_H__
 
-#include <eXosip/eXosip.h>
+#include <eXosip2/eXosip.h>
 #include <vector>
 
 class CodecDescriptor;
 class AudioCodec;
 
 #define NOT_USED      0
+#define _SENDRECV 0
+#define _SENDONLY 1
+#define _RECVONLY 2
 using namespace std;
 
 typedef vector<CodecDescriptor*, allocator<CodecDescriptor*> > CodecDescriptorVector;
@@ -43,9 +46,9 @@ public:
 	
 	int  	newIncomingCall 	(eXosip_event_t *);
 	int  	answeredCall 		(eXosip_event_t *);
+	void  	answeredCall_without_hold (eXosip_event_t *);
 	int  	ringingCall			(eXosip_event_t *);
-	int  	onholdCall			(eXosip_event_t *);
-	int  	offholdCall			(eXosip_event_t *);
+	int  	receivedAck			(eXosip_event_t *);
 
 	void	setLocalAudioPort 	(int);
 	int 	getLocalAudioPort 	(void);	
@@ -55,11 +58,15 @@ public:
 	int 	getDid				(void);
 	void 	setCid				(int cid);
 	int 	getCid				(void);
+	void 	setTid				(int tid);
+	int 	getTid				(void);
 	int 	getRemoteSdpAudioPort (void);
 	char* 	getRemoteSdpAudioIp (void);
 	AudioCodec* getAudioCodec	(void);
 	void	setAudioCodec		(AudioCodec* ac);
 
+	inline char* getRemoteUri (void) { return _remote_uri; }
+
 	inline void setStandBy (bool standby) { _standby = standby; }
 	inline bool getStandBy (void) { return _standby; }
 
@@ -67,6 +74,9 @@ private:
 	void	alloc			(void);
 	void	dealloc			(void);
 	void 	noSupportedCodec(void);	
+
+	int sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg);
+	int sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med);
 	
 	CodecDescriptorVector* _cdv;
 	AudioCodec* _audiocodec;
@@ -74,6 +84,7 @@ private:
 	short 	_id;
 	int 	_cid;	// call id
   	int 	_did;	// dialog id
+  	int 	_tid;	// transaction id
 	bool	_standby; // wait for a cid and did when outgoing call is made
 
   	int  	_status_code;
@@ -88,9 +99,12 @@ private:
 	char*	_remote_sdp_audio_ip;
   	char*	_payload_name;
 	char*	_sdp_body;
+	int		_payload;
   	int  	_state;
 	int		_local_audio_port;
   	int  	_remote_sdp_audio_port;
+	int 	_local_sendrecv;           /* _SENDRECV, _SENDONLY, _RECVONLY */
+  	int 	_remote_sendrecv;          /* _SENDRECV, _SENDONLY, _RECVONLY */
 };
 
 #endif // __SIP_CALL_H__

src/sipvoiplink.h

@@ -20,7 +20,8 @@
 #ifndef __SIP_VOIP_LINK_H__
 #define __SIP_VOIP_LINK_H__
 
-#include <eXosip/eXosip.h>
+#include <eXosip2/eXosip.h>
+#include <osipparser2/sdp_message.h>
 
 #include <string>
 #include <vector>
@@ -30,8 +31,12 @@
 using namespace std;
 
 #define EXPIRES_VALUE	180
+#define INVITE_METHOD	"INVITE"
 // 1XX responses
 #define DIALOG_ESTABLISHED 101
+#define RINGING			180
+// 2XX
+#define OK				200
 // 4XX Errors
 #define	BAD_REQ			400
 #define	UNAUTHORIZED	401
@@ -41,6 +46,7 @@ using namespace std;
 #define NOT_ACCEPTABLE	406
 #define AUTH_REQUIRED	407
 #define REQ_TIMEOUT		408
+#define UNSUP_MEDIA_TYPE	415
 #define TEMP_UNAVAILABLE 480
 #define	ADDR_INCOMPLETE	484
 #define	BUSY_HERE		486
@@ -122,6 +128,21 @@ private:
 	short findCallId (eXosip_event_t *e);
 	short findCallIdWhenRinging (void);
 	bool isInRtpmap (int index, int payload, CodecDescriptorVector* cdv);
+
+	/*
+	 * To build sdp with 200 OK when answered call
+	 */
+	int sdp_complete_200ok(int did, osip_message_t * answer, int port);
+	
+	/*
+	 * To build sdp when call is on-hold
+	 */
+	int sdp_hold_call (sdp_message_t * sdp);
+	
+	/*
+	 * To build sdp when call is off-hold
+	 */
+	int sdp_off_hold_call (sdp_message_t * sdp);
 	
 	EventThread* 	_evThread;
 	SipCallVector* 	_sipcallVector;