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sipvoiplink.h 5.90 KiB
/**
* Copyright (C) 2004-2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
* Portions Copyright (C) 2002,2003 Aymeric Moizard <jack@atosc.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __SIP_VOIP_LINK_H__
#define __SIP_VOIP_LINK_H__
#include <vector>
#include <eXosip2/eXosip.h>
//#include <osipparser2/sdp_message.h>
#include "voIPLink.h"
#include "audio/audiortp.h"
#include "call.h" // for CALLID
#define EXPIRES_VALUE 180
// To build request
#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
#define FORBIDDEN 403
#define NOT_FOUND 404
#define NOT_ALLOWED 405
#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
#define REQ_TERMINATED 487
#define NOT_ACCEPTABLE_HERE 488 // Not Acceptable Here
// 5XX errors
#define SERVICE_UNAVAILABLE 503
// 6XX errors
#define BUSY_EVERYWHERE 600
#define DECLINE 603
class AudioCodec;
class CodecDescriptor;
class SipCall;
class EventThread;
typedef std::vector< CodecDescriptor* > CodecDescriptorVector;
class SipVoIPLink : public VoIPLink {public:
SipVoIPLink();
virtual ~SipVoIPLink();
virtual bool init (void);
virtual bool checkNetwork (void);
virtual void terminate (void);
virtual int setRegister (void);
virtual int setUnregister (void);
virtual int outgoingInvite (CALLID id, const std::string& to_url);
virtual int answer (CALLID id);
virtual int hangup (CALLID id);
virtual int cancel (CALLID id);
virtual int onhold (CALLID id);
virtual int offhold (CALLID id);
virtual int transfer (CALLID id, const std::string& to);
virtual int refuse (CALLID id);
virtual int getEvent (void);
virtual void carryingDTMFdigits (CALLID id, char code);
/*
* To handle the local port
*/
int getLocalPort (void);
void setLocalPort (int);
bool getSipLocalIp (void);
/*
* Add a new SipCall at the end of the SipCallVector with identifiant 'id'
*/
void newOutgoingCall(CALLID callid);
void newIncomingCall(CALLID callid);
/*
* Erase the SipCall(id) from the SipCallVector
*/
void deleteSipCall(CALLID callid);
/*
* Return a pointer to the SipCall with identifiant 'id'
*/
SipCall* getSipCall(CALLID callid);
/*
* Accessor to the audio codec of SipCall with identifiant 'id'
*/
AudioCodec* getAudioCodec(CALLID callid);
// Handle voice-message
inline void setMsgVoicemail (int nMsg) { _nMsgVoicemail = nMsg; }
inline int getMsgVoicemail (void) { return _nMsgVoicemail; }
private:
/*
* If you are behind a NAT, you have to use STUN server, specified in
* STUN configuration(you can change this one by default) to give you an
* public IP address and assign a port number.
*
* Return 0 if an error occured and 1 if no error.
*/
int behindNat (void);
/*
* Return -1 if an error occured and 0 if no error
*/
int checkUrl(const std::string& url);
/*
* Allow the authentication when you want register
* Return -1 if an error occured and 0 if no error */
int setAuthentication (void);
/*
* Build a sip address from the user configuration
* Example: "Display user name" <sip:user@host.com>
* Return the result in a string
*/
std::string fromHeader (const std::string& user, const std::string& host);
/*
* Build a sip address with the number that you want to call
* Example: sip:124@domain.com
* Return the result in a string
*/
std::string toHeader(const std::string& to);
/*
* Beginning point to make outgoing call.
* Check the 'from' and 'to' url.
* Allocate local audio port.
* Build SDP body.
* Return -1 if an error occured and 0 if no error
*/
int startCall (CALLID id, const std::string& from, const std::string& to,
const std::string& subject, const std::string& route);
/*
* Look for call with same cid/did
* Return the id of the found call
*/
CALLID findCallId (eXosip_event_t *e);
CALLID findCallIdInitial (eXosip_event_t *e);
/*
* Return true if payload is already in the rtpmap and false if not
*/
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 * answerMessage, 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);
/**
* Subscribe to message summary
*/
void subscribeMessageSummary();
/**
* End all sip call not deleted
*/
void endSipCalls();
///////////////////////////
// Private member variables
///////////////////////////
EventThread *_evThread;
std::vector< SipCall * > _sipcallVector;
AudioRtp _audiortp; int _localPort;
int _reg_id;
int _nMsgVoicemail;
bool _registrationSend; // unregistered
bool _started; // eXosip_init and eXosip_start
};
#endif // __SIP_VOIP_LINK_H__