diff --git a/src/Makefile.am b/src/Makefile.am
index 2d5e1af3133debef313b4a3046209c7bce46da23..311be41e02e478b31e549bd5a1a2e9c29cc6b65f 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -39,8 +39,7 @@ sflphoned_SOURCES = \
call.cpp \
account.cpp \
sipcall.cpp \
- $(IAXSOURCES) \
- useragent.cpp
+ $(IAXSOURCES)
sflphoned_CXXFLAGS = \
-DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone\" \
@@ -80,8 +79,7 @@ noinst_HEADERS = \
accountcreator.h \
sipvoiplink.h \
call.h \
- sipcall.h \
- useragent.h
+ sipcall.h
libsflphone_la_LIBADD = \
$(src)/libs/stund/libstun.la \
diff --git a/src/eventthread.cpp b/src/eventthread.cpp
index 66e6d441090be9d70a785ecc7fa0ff44a5256b44..e7103908a6b2e8df05d0528c37c1cb1936b1a88e 100644
--- a/src/eventthread.cpp
+++ b/src/eventthread.cpp
@@ -1,8 +1,7 @@
/*
- * Copyright (C) 2004, 2005, 2006 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
* 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 3 of the License, or
@@ -21,43 +20,31 @@
#include "eventthread.h"
#include "voiplink.h"
-EventThread::EventThread (VoIPLink* link) : Thread (), _linkthread(link), stopIt(false)
-{
- setCancel(cancelDeferred);
-}
-
-EventThread::~EventThread (void)
-{
- terminate();
-}
-
+/********************************** IAX Voiplink thread *************************************/
/**
* Reimplementation of run()
*/
-void
-EventThread::run (void)
+void IAXEventThread::run (void)
{
- //stopIt = false;
while(!testCancel()) {
_linkthread->getEvent();
}
-}
+}
-void
-EventThread::stop( void )
-{
- stopIt = true;
-}
+/********************************************************************************************/
-void
-EventThread::startLoop( void )
-{
- stopIt = false;
- //start();
-}
-bool
-EventThread::isStopped( void )
+
+/********************************** SIP Voiplink thread *************************************/
+/**
+ * Reimplementation of run()
+ */
+void SIPEventThread::run (void)
+
{
- return stopIt;
+ while(!testCancel()) {
+ _linkthread->getEvent();
+ }
}
+
+/********************************************************************************************/
diff --git a/src/eventthread.h b/src/eventthread.h
index 8772c72043fec8d719b7c50aecf706dbdb241396..f09afb016fc20c2de26ba9098de7987f1a7b2d67 100644
--- a/src/eventthread.h
+++ b/src/eventthread.h
@@ -1,7 +1,6 @@
/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
+ * Copyright (C) 2009 Savoir-Faire Linux inc.
+ * Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* 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
@@ -31,24 +30,52 @@ class VoIPLink;
*/
class EventThread : public ost::Thread {
-public:
- /**
- * Build a thread that call getEvents
- */
- EventThread (VoIPLink*);
- ~EventThread (void);
- virtual void run ();
- virtual void stop();
- virtual void startLoop();
- bool isStopped();
-
-private:
+
+ friend class SIPEventThread;
+ friend class IAXEventThread;
+
+ public:
+ /**
+ * Thread constructor
+ */
+ EventThread (VoIPLink* link) : Thread(), _linkthread(link){
+ setCancel( cancelDeferred );
+ }
+
+
+ virtual ~EventThread (void){
+ terminate();
+ }
+
+ virtual void run () = 0;
+
+ private:
EventThread(const EventThread& rh); // copy constructor
EventThread& operator=(const EventThread& rh); // assignment operator
/** VoIPLink is the object being called by getEvents() method */
- VoIPLink* _linkthread;
- bool stopIt;
+ VoIPLink* _linkthread;
+};
+
+class IAXEventThread : public EventThread {
+
+ public:
+ IAXEventThread( VoIPLink* voiplink )
+ : EventThread( voiplink ){
+ }
+
+ virtual void run();
};
+class SIPEventThread : public EventThread {
+
+ public:
+ SIPEventThread( VoIPLink* voiplink )
+ : EventThread( voiplink ){
+ }
+
+ virtual void run();
+};
+
+
#endif // __EVENT_THREAD_H__
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 6d04dffcaeae225d5d75207136262f3a36705b1c..464bf2b9cc7f477f60697b652845062ac819e0fc 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -45,7 +45,7 @@
IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID)
{
- _evThread = new EventThread(this);
+ _evThread = new IAXEventThread(this);
_regSession = NULL;
_nextRefreshStamp = 0;
@@ -917,6 +917,8 @@ IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
void IAXVoIPLink::updateAudiolayer( void )
{
+ _mutexIAX.enterMutex();
audiolayer = NULL;
audiolayer = Manager::instance().getAudioDriver();
+ _mutexIAX.leaveMutex();
}
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 626aca506c6114558d6a1db42fae6259e55a019b..e7af691255f2101c6a2f637c2384271e2a088523 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -46,15 +46,8 @@
#include "accountcreator.h" // create new account
#include "sipvoiplink.h"
-#include "useragent.h"
-
#include "user_cfg.h"
-#ifdef USE_ZEROCONF
-#include "zeroconf/DNSService.h"
-#include "zeroconf/DNSServiceTXTRecord.h"
-#endif
-
#define fill_config_str(name, value) \
(_config.addConfigTreeItem(section, Conf::ConfigTreeItem(std::string(name), std::string(value), type_str)))
#define fill_config_int(name, value) \
@@ -88,26 +81,10 @@ ManagerImpl::ManagerImpl (void)
, _callAccountMap()
//, _callAccountMapMutex()
, _accountMap()
- , _userAgent(NULL)
- , _userAgentInitlized(false)
- , _sipThreadStop()
-
{
- /* Init private variables
- setup: _path, _exist, _setupLoaded , _dbus
- sound: _audiodriver, _dtmfKey,
- _spkr_volume,_mic_volume = 0; // Initialize after by init() -> initVolume()
- Call: _nbIncomingWaitingCall, _hasTriedToRegister
- SIP Link: _userAgent, _userAgentInitlized
- */
-
-#ifdef USE_ZEROCONF
- _hasZeroconf = true;
- _DNSService = new DNSService();
-#endif
-
- // initialize random generator for call id
- srand (time(NULL));
+
+ // initialize random generator for call id
+ srand (time(NULL));
#ifdef TEST
testAccountMap();
@@ -123,82 +100,60 @@ ManagerImpl::ManagerImpl (void)
// never call if we use only the singleton...
ManagerImpl::~ManagerImpl (void)
{
- terminate();
-
-#ifdef USE_ZEROCONF
- delete _DNSService; _DNSService = 0;
-#endif
-
- _debug("%s stop correctly.\n", PROGNAME);
+ terminate();
+ _debug("%s stop correctly.\n", PROGNAME);
}
void
ManagerImpl::init()
{
- // Load accounts, init map
- loadAccountMap();
+ // Load accounts, init map
+ loadAccountMap();
- //Initialize sip manager
- if(_userAgentInitlized) {
- _userAgent->sipCreate();
- _userAgent->sipInit();
- }
+ initVolume();
- initVolume();
-
- if (_exist == 0) {
- _debug("Cannot create config file in your home directory\n");
- }
-
- initAudioDriver();
- selectAudioDriver();
+ if (_exist == 0) {
+ _debug("Cannot create config file in your home directory\n");
+ }
- // Initialize the list of supported audio codecs
- initAudioCodec();
+ initAudioDriver();
+ selectAudioDriver();
- getAudioInputDeviceList();
+ // Initialize the list of supported audio codecs
+ initAudioCodec();
- AudioLayer *audiolayer = getAudioDriver();
- if (audiolayer!=0) {
- unsigned int sampleRate = audiolayer->getSampleRate();
+ getAudioInputDeviceList();
- _debugInit("Load Telephone Tone");
- std::string country = getConfigString(PREFERENCES, ZONE_TONE);
- _telephoneTone = new TelephoneTone(country, sampleRate);
+ AudioLayer *audiolayer = getAudioDriver();
+ if (audiolayer!=0) {
+ unsigned int sampleRate = audiolayer->getSampleRate();
- _debugInit("Loading DTMF key");
- _dtmfKey = new DTMF(sampleRate);
- }
+ _debugInit("Load Telephone Tone");
+ std::string country = getConfigString(PREFERENCES, ZONE_TONE);
+ _telephoneTone = new TelephoneTone(country, sampleRate);
- // initRegisterAccounts was here, but we doing it after the gui loaded...
- // the stun detection is long, so it's a better idea to do it after getEvents
- initZeroconf();
-
+ _debugInit("Loading DTMF key");
+ _dtmfKey = new DTMF(sampleRate);
+ }
}
void ManagerImpl::terminate()
{
- saveConfig();
+ saveConfig();
- unloadAccountMap();
+ unloadAccountMap();
- if(_userAgentInitlized) {
- delete _userAgent;
- _userAgent = NULL;
- _userAgentInitlized = false;
- }
+ _debug("Unload DTMF Key\n");
+ delete _dtmfKey;
- _debug("Unload DTMF Key\n");
- delete _dtmfKey;
+ _debug("Unload Audio Driver\n");
+ delete _audiodriver; _audiodriver = NULL;
- _debug("Unload Audio Driver\n");
- delete _audiodriver; _audiodriver = NULL;
+ _debug("Unload Telephone Tone\n");
+ delete _telephoneTone; _telephoneTone = NULL;
- _debug("Unload Telephone Tone\n");
- delete _telephoneTone; _telephoneTone = NULL;
-
- _debug("Unload Audio Codecs\n");
- _codecDescriptorMap.deleteHandlePointer();
+ _debug("Unload Audio Codecs\n");
+ _codecDescriptorMap.deleteHandlePointer();
}
bool
@@ -443,25 +398,29 @@ ManagerImpl::saveConfig (void)
int
ManagerImpl::initRegisterAccounts()
{
-
int status;
+ //TODO What the flag is for ??
bool flag = true;
AccountMap::iterator iter;
_debugInit("Initiate VoIP Links Registration");
iter = _accountMap.begin();
+ /* Loop on the account map previously loaded */
while( iter != _accountMap.end() ) {
- if ( iter->second ) {
- iter->second->loadConfig();
- if ( iter->second->isEnabled() ) {
- status = iter->second->registerVoIPLink();
- if (status != SUCCESS)
- flag = false;
+ if ( iter->second ) {
+ iter->second->loadConfig();
+ /* If the account is set as enabled, try to register */
+ if ( iter->second->isEnabled() ) {
+ status = iter->second->registerVoIPLink();
+ if (status != SUCCESS){
+ flag = false;
+ }
+ }
}
- }
- iter++;
+ iter++;
}
+
// calls the client notification here in case of errors at startup...
if( _audiodriver -> getErrorMessage() != -1 )
notifyErrClient( _audiodriver -> getErrorMessage() );
@@ -1694,24 +1653,6 @@ ManagerImpl::switchAudioManager( void )
notifyErrClient( _audiodriver -> getErrorMessage());
}
-/**
- * Initialize the Zeroconf scanning services loop
- * Informations will be store inside a map DNSService->_services
- * Initialization: Main Thread
- */
- void
-ManagerImpl::initZeroconf(void)
-{
-#ifdef USE_ZEROCONF
- _debugInit("Zeroconf Initialization");
- int useZeroconf = getConfigInt(PREFERENCES, CONFIG_ZEROCONF);
-
- if (useZeroconf) {
- _DNSService->startScanServices();
- }
-#endif
-}
-
/**
* Init the volume for speakers/micro from 0 to 100 value
* Initialization: Main Thread
@@ -1750,6 +1691,16 @@ void ManagerImpl::setMicVolume(unsigned short mic_vol)
//}
}
+void ManagerImpl::setSipPort( int port )
+{
+}
+
+int ManagerImpl::getSipPort( void )
+{
+ return 5060;
+}
+
+
/**
* configuration function requests
* Main Thread
@@ -2021,26 +1972,17 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
if (accountType == "SIP") {
setConfig(accountID, SIP_STUN_SERVER,(*details.find(SIP_STUN_SERVER)).second);
setConfig(accountID, SIP_USE_STUN, (*details.find(SIP_USE_STUN)).second == "TRUE" ? "1" : "0");
-
- if(!_userAgentInitlized) {
- _userAgentInitlized = true;
-
- if((*details.find(SIP_USE_STUN)).second == "TRUE")
- _userAgent->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
- else
- _userAgent->setStunServer(NULL);
- _userAgent->sipCreate();
- _userAgent->sipInit();
- } else {
- if((*details.find(SIP_USE_STUN)).second == "TRUE")
- _userAgent->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
- else
- _userAgent->setStunServer(NULL);
-
- restartPjsip();
- }
- }
+ if((*details.find(SIP_USE_STUN)).second == "TRUE")
+ {
+ //TODO Replace: _userAgent->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
+ }
+ else
+ {
+ //TODO: replace: _userAgent->setStunServer(NULL);
+ }
+ //restartPjsip();
+ }
saveConfig();
@@ -2082,38 +2024,30 @@ ManagerImpl::sendRegister( const std::string& accountID , const int32_t& expire
ManagerImpl::addAccount(const std::map< std::string, std::string >& details)
{
- /** @todo Deal with both the _accountMap and the Configuration */
- std::string accountType = (*details.find(CONFIG_ACCOUNT_TYPE)).second;
- Account* newAccount;
- std::stringstream accountID;
- accountID << "Account:" << time(NULL);
- AccountID newAccountID = accountID.str();
- /** @todo Verify the uniqueness, in case a program adds accounts, two in a row. */
+ /** @todo Deal with both the _accountMap and the Configuration */
+ std::string accountType = (*details.find(CONFIG_ACCOUNT_TYPE)).second;
+ Account* newAccount;
+ std::stringstream accountID;
+ accountID << "Account:" << time(NULL);
+ AccountID newAccountID = accountID.str();
+ /** @todo Verify the uniqueness, in case a program adds accounts, two in a row. */
- if (accountType == "SIP") {
- if(!_userAgentInitlized) {
- // Initialize the SIP Manager
- _userAgent = new UserAgent();
- _userAgent->setSipPort(Manager::instance().getConfigInt(PREFERENCES , CONFIG_SIP_PORT));
- }
-
- newAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, newAccountID);
- }
- else if (accountType == "IAX") {
- newAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, newAccountID);
- }
- else {
- _debug("Unknown %s param when calling addAccount(): %s\n", CONFIG_ACCOUNT_TYPE, accountType.c_str());
- return;
- }
- _accountMap[newAccountID] = newAccount;
- setAccountDetails(accountID.str(), details);
-
- saveConfig();
+ if (accountType == "SIP") {
+ newAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, newAccountID);
+ }
+ else if (accountType == "IAX") {
+ newAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, newAccountID);
+ }
+ else {
+ _debug("Unknown %s param when calling addAccount(): %s\n", CONFIG_ACCOUNT_TYPE, accountType.c_str());
+ return;
+ }
+ _accountMap[newAccountID] = newAccount;
+ setAccountDetails(accountID.str(), details);
- if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
+ saveConfig();
- //restartPjsip();
+ if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
}
void
@@ -2192,20 +2126,6 @@ ManagerImpl::getNewCallID()
return random_id.str();
}
- void
-ManagerImpl::restartPjsip()
-{
- if ( _userAgentInitlized ){
- unregisterCurSIPAccounts();
- _userAgent->sipDestory();
- //_userAgent->setSipPort(Manager::instance().getConfigInt(PREFERENCES , CONFIG_SIP_PORT));
-
- _userAgent->sipCreate();
- _userAgent->sipInit();
- registerCurSIPAccounts();
- }
-}
-
short
ManagerImpl::loadAccountMap()
{
@@ -2214,7 +2134,6 @@ ManagerImpl::loadAccountMap()
TokenList sections = _config.getSections();
std::string accountType;
Account* tmpAccount;
- std::string port;
TokenList::iterator iter = sections.begin();
while(iter != sections.end()) {
@@ -2226,28 +2145,7 @@ ManagerImpl::loadAccountMap()
accountType = getConfigString(*iter, CONFIG_ACCOUNT_TYPE);
if (accountType == "SIP") {
- if(!_userAgentInitlized) {
- // Initialize the SIP Manager
- _userAgent = new UserAgent();
- _userAgentInitlized = true;
- _userAgent->setSipPort(Manager::instance().getConfigInt(PREFERENCES , CONFIG_SIP_PORT));
- }
-
tmpAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, *iter);
-
- // Determine whether to use stun for the current account or not
- int useStun = Manager::instance().getConfigInt(tmpAccount->getAccountID(),SIP_USE_STUN);
-
- if(useStun == 1) {
- _userAgent->setStunServer(Manager::instance().getConfigString(tmpAccount->getAccountID(), SIP_STUN_SERVER).data());
- }
-
- /*// Set registration port for all accounts, The last non-5060 port will be recorded in _userAgent.
- port = Manager::instance().getConfigString(tmpAccount->getAccountID(), SIP_PORT);
- std::istringstream is(port);
- is >> iPort;
- if (iPort != DEFAULT_SIP_PORT)
- _userAgent->setRegPort(iPort); */
}
else if (accountType == "IAX") {
tmpAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, *iter);
@@ -2262,8 +2160,6 @@ ManagerImpl::loadAccountMap()
nbAccount++;
}
- _debug("\n");
-
iter++;
}
@@ -2351,37 +2247,6 @@ pjsip_regc
return NULL;
}
-
-/**
- * Return the instance of sip manager
- */
-UserAgent *ManagerImpl::getUserAgent()
-{
- return _userAgent;
-}
-
-int
-ManagerImpl::getSipPort()
-{
- if( _userAgent )
- return _userAgent->getSipPort();
- else
- {
- // It means that no SIP accounts are configured, so return a default value
- return 0;
- }
-}
-
-void
-ManagerImpl::setSipPort(int portNum)
-{
- if(portNum != _userAgent->getSipPort()) {
- _userAgent->setSipPort(portNum);
- restartPjsip();
- setConfig( PREFERENCES , CONFIG_SIP_PORT , portNum );
- }
-}
-
void ManagerImpl::unregisterCurSIPAccounts()
{
AccountMap::iterator iter = _accountMap.begin();
diff --git a/src/managerimpl.h b/src/managerimpl.h
index b13fc911a88d79580e739ac8ea8b5439d5ad7424..68a40075d5b0eda4dd3451cb47796e44757040f8 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -47,7 +47,6 @@ class CodecDescriptor;
class GuiFramework;
class TelephoneTone;
class VoIPLink;
-class UserAgent;
#ifdef USE_ZEROCONF
class DNSService;
@@ -795,10 +794,6 @@ class ManagerImpl {
*/
void restartPjsip();
- int getSipPort();
-
- void setSipPort(int port);
-
void unregisterCurSIPAccounts();
void registerCurSIPAccounts();
@@ -1019,6 +1014,10 @@ public:
AccountID getAccountIdFromNameAndServer(const std::string& userName, const std::string& server);
+ int getSipPort();
+
+ void setSipPort( int port );
+
private:
// Copy Constructor
@@ -1027,32 +1026,12 @@ private:
// Assignment Operator
ManagerImpl& operator=( const ManagerImpl& rh);
- /**
- * The UserAgent provides sip operation facilities for all sip accounts
- */
- UserAgent *_userAgent;
-
- /** Whether the _UserAgent has been initialized */
- bool _userAgentInitlized;
-
- bool _sipThreadStop;
-
#ifdef TEST
bool testCallAccountMap();
bool testAccountMap();
#endif
friend class ConfigurationTest;
-
-public:
- /**
- * Retuun the instance of sip manager
- */
- UserAgent *getUserAgent();
-
- void setSipThreadStatus(bool status) {_sipThreadStop = status;}
-
- bool getSipThreadStatus() {return _sipThreadStop;}
};
#endif // __MANAGER_H__
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index acce1bb66ed5fff56fad4d6a3d20b77c6ba146a0..27f1acacee36034d29c2af543c77d0da7cced26a 100644
--- a/src/sipaccount.cpp
+++ b/src/sipaccount.cpp
@@ -23,7 +23,6 @@
#include "sipaccount.h"
#include "manager.h"
#include "user_cfg.h"
-#include "useragent.h"
SIPAccount::SIPAccount(const AccountID& accountID)
: Account(accountID)
@@ -33,6 +32,7 @@ SIPAccount::SIPAccount(const AccountID& accountID)
, _contact("")
{
_link = new SIPVoIPLink(accountID);
+ //_link = SIPVoIPLink::instance( accountID );
}
@@ -44,16 +44,15 @@ SIPAccount::~SIPAccount()
_cred = NULL;
}
-int
-SIPAccount::registerVoIPLink()
+int SIPAccount::registerVoIPLink()
{
int status, useStun;
SIPVoIPLink *thislink;
+ /* Retrieve the account information */
_link->setHostname(Manager::instance().getConfigString(_accountID,HOSTNAME));
useStun = Manager::instance().getConfigInt(_accountID,SIP_USE_STUN);
-
thislink = dynamic_cast<SIPVoIPLink*> (_link);
thislink->setStunServer(Manager::instance().getConfigString(_accountID,SIP_STUN_SERVER));
thislink->setUseStun( useStun!=0 ? true : false);
@@ -65,6 +64,7 @@ SIPAccount::registerVoIPLink()
thislink->setPassword(Manager::instance().getConfigString(_accountID, PASSWORD));
thislink->setHostname(Manager::instance().getConfigString(_accountID, HOSTNAME));
+ // Start registration
status = _link->sendRegister();
ASSERT( status , SUCCESS );
@@ -76,7 +76,6 @@ SIPAccount::unregisterVoIPLink()
{
_debug("SIPAccount: unregister account %s\n" , getAccountID().c_str());
_link->sendUnregister();
- _link->terminate();
return SUCCESS;
}
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 0e53bdd013d066744d80db93204ffbf14a46549f..bcc596ccf7f1f3a1b33a9befc4fccc2bc03167ce 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -51,13 +51,13 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
{
pj_status_t status;
+ // We retrieve the remote sdp offer in the rdata struct to begin the negociation
pjmedia_sdp_session* remote_sdp = getRemoteSDPFromRequest(rdata);
if (remote_sdp == 0) {
return false;
}
// Have to do some stuff here with the SDP
- // We retrieve the remote sdp offer in the rdata struct to begin the negociation
_localSDP = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_session);
_localSDP->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
@@ -68,7 +68,11 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
_localSDP->time.start = _localSDP->time.stop = 0;
sdpAddMediaDescription(pool);
+ _debug("SDP: addr: %s\nuser: %s\nid: %i\nversion: %i\naddr: %s\nattr count: %i\n",
+ _ipAddr.c_str(), _localSDP->origin.user, _localSDP->origin.id, _localSDP->origin.version, _localSDP->origin.addr.ptr, _localSDP->attr_count );
+
_debug("Before validate SDP!\n");
+
status = pjmedia_sdp_validate( _localSDP );
if (status != PJ_SUCCESS) {
_debug("Can not generate valid local sdp\n");
diff --git a/src/sipcall.h b/src/sipcall.h
index fe23553d5dcdd5edda45b80c6466ea3d6b124725..2ef3d2793451e85bd1fd48151428959cfd7db854 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -22,8 +22,8 @@
#define SIPCALL_H
#include "call.h"
+#include "sipvoiplink.h"
#include "audio/codecDescriptor.h"
-#include "useragent.h"
class AudioCodec;
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index c71fd1da9eb1392f4074c80f29d88a3c6ea40327..d05ebd56f83ba2db81708bbee7b8d62d840ee778 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -27,256 +27,500 @@
#include "sipcall.h"
#include <sstream> // for istringstream
#include "sipaccount.h"
-#include "useragent.h"
#include "audio/audiortp.h"
-
+
#include "manager.h"
#include "user_cfg.h" // SIGNALISATION / PULSE #define
-#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
-
-// 1XX responses
-#define DIALOG_ESTABLISHED 101
-// see: osip_const.h
-
-// need for hold/unhold
-#define INVITE_METHOD "INVITE"
-
-SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
- : VoIPLink(accountID)
- , _initDone(false)
- , _nbTryListenAddr(2) // number of times to try to start SIP listener
- , _useStun(false)
- , _stunServer("")
- , _localExternAddress("")
- , _localExternPort(0)
- , _audiortp(new AudioRtp())
- , _regc()
- , _bRegister(false)
+/** PJSIP related variables */
+pj_caching_pool _cp;
+pj_pool_t *_pool;
+pjsip_endpoint *_endpt;
+pjsip_module _mod_ua; /** PJSIP module. */
+pj_thread_t *thread;
+pj_thread_desc desc;
+
+/*****************************/
+
+void set_voicemail_info( AccountID account, pjsip_msg_body *body );
+
+/**
+ * Set audio (SDP) configuration for a call
+ * localport, localip, localexternalport
+ * @param call a SIPCall valid pointer
+ * @return bool True
+ */
+bool setCallAudioLocal(SIPCall* call);
+
+
+/** Do we use stun? */
+bool _useStun;
+
+/** The current STUN server address */
+std::string _stunServer;
+
+std::string _localIPAddress;
+
+/** PJSIP callbacks */
+void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e);
+void call_on_media_update( pjsip_inv_session *inv UNUSED, pj_status_t status UNUSED);
+void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
+void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
+void regc_cb(struct pjsip_regc_cbparam *param);
+pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata);
+pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata UNUSED) ;
+void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event);
+void xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event);
+
+void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata);
+/*******************************/
+
+
+
+ SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
+ : VoIPLink(accountID)
+ , _initDone(false)
+ , _nbTryListenAddr(2) // number of times to try to start SIP listener
+ , _stunServer("")
+ , _localExternAddress("")
+ , _localExternPort(0)
+ , _audiortp(new AudioRtp())
+ , _regc()
+ , _bRegister(false)
+ ,_regPort(DEFAULT_SIP_PORT)
{
- // to get random number for RANDOM_PORT
- srand (time(NULL));
+ // to get random number for RANDOM_PORT
+ srand (time(NULL));
+ _useStun = false;
+ _localIPAddress = "127.0.0.1";
+
+ _evThread = new SIPEventThread(this);
}
SIPVoIPLink::~SIPVoIPLink()
{
- terminate();
+ delete _evThread; _evThread = NULL;
+ terminate();
+}
+
+SIPVoIPLink* SIPVoIPLink::instance( const AccountID& id){
+ /*if(!_instance ){
+ _instance = new SIPVoIPLink( id );
+ }
+ return _instance;*/
}
-bool
-SIPVoIPLink::init()
+
+bool SIPVoIPLink::init()
{
- _regc = NULL;
- _initDone = true;
- return true;
+ /* Initialize the pjsip library */
+ if( !isInitDone() ){
+ _regc = NULL;
+ pjsip_init();
+ }
+
+ setInitState(true);
+ return true;
}
-void
+ void
SIPVoIPLink::terminate()
{
- _initDone = false;
+ /* Clean shutdown of pjsip library */
+ pjsip_shutdown();
+ _initDone = false;
}
-void
+ void
SIPVoIPLink::terminateSIPCall()
{
-
- //ost::MutexLock m(_callMapMutex);
- CallMap::iterator iter = _callMap.begin();
- SIPCall *call;
- while( iter != _callMap.end() ) {
- call = dynamic_cast<SIPCall*>(iter->second);
- if (call) {
- //TODO terminate the sip call
- delete call; call = 0;
+
+ //ost::MutexLock m(_callMapMutex);
+ CallMap::iterator iter = _callMap.begin();
+ SIPCall *call;
+ while( iter != _callMap.end() ) {
+ call = dynamic_cast<SIPCall*>(iter->second);
+ if (call) {
+ //TODO terminate the sip call
+ delete call; call = 0;
+ }
+ iter++;
}
- iter++;
- }
- _callMap.clear();
+ _callMap.clear();
}
-void
+ void
SIPVoIPLink::getEvent()
{
- // Nothing anymore. PJSIP is based on asynchronous events
+ // We have to register the external thread so it could access the pjsip framework
+ if(!pj_thread_is_registered())
+ {
+ pj_thread_register( NULL, desc, &thread );
+ }
+
+ _mutexSIP.enterMutex();
+
+ // PJSIP polling
+ pj_time_val timeout = {0, 10};
+ pjsip_endpt_handle_events( _endpt, &timeout);
+
+ _mutexSIP.leaveMutex();
}
-int
-SIPVoIPLink::sendRegister()
+int SIPVoIPLink::sendRegister()
{
- AccountID id;
- pj_status_t status;
-
- id = getAccountID();
- if(_regc) {
- status = pjsip_regc_destroy(_regc);
- _regc = NULL;
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
- }
+ AccountID id;
+ pj_status_t status;
+ int expire_value;
+ char contactTmp[256];
+ pj_str_t svr, aor, contact;
+ pjsip_tx_data *tdata;
+ std::string tmp, hostname, username, password;
+ SIPAccount *account;
+
+ /* Get the current account ID */
+ id = getAccountID();
+ hostname = getHostname();
+ username = getUsername();
+ password = getPassword();
+
+ _mutexSIP.enterMutex();
+
+ /* If the registration already exists, delete it */
+ if(_regc) {
+ status = pjsip_regc_destroy(_regc);
+ _regc = NULL;
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+ }
+
+ setRegister(true);
+
+ /* Set the expire value of the message from the config file */
+ expire_value = Manager::instance().getRegistrationExpireValue();
+
+ /* Update the state of the voip link */
+ setRegistrationState(Trying);
+
+ if (!validStunServer) {
+ setRegistrationState(VoIPLink::ErrorExistStun);
+ setRegister(false);
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ status = pjsip_regc_create(_endpt, (void*)new AccountID(id), ®c_cb, &_regc);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create regc.\n");
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ tmp = "sip:" + hostname;
+ pj_strdup2(_pool, &svr, tmp.data());
+
+ tmp = "<sip:" + username + "@" + hostname + ">";
+ pj_strdup2(_pool, &aor, tmp.data());
+
+ sprintf(contactTmp, "<sip:%s@%s:%d>", username.data(), _localExternAddress.data(), _localExternPort);
+ pj_strdup2(_pool, &contact, contactTmp);
+
+ status = pjsip_regc_init(_regc, &svr, &aor, &aor, 1, &contact, 600); //timeout);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to initialize regc. %d\n", status); //, regc->str_srv_url.ptr);
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount(id));
+ pjsip_cred_info *cred = account->getCredInfo();
+
+ if(!cred)
+ cred = new pjsip_cred_info();
+
+ pj_bzero(cred, sizeof (pjsip_cred_info));
+ pj_strdup2(_pool, &cred->username, username.data());
+ cred->data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
+ pj_strdup2(_pool, &cred->data, password.data());
+ pj_strdup2(_pool, &cred->realm, "*");
+ pj_strdup2(_pool, &cred->scheme, "digest");
+ pjsip_regc_set_credentials(_regc, 1, cred);
- _bRegister = true;
-
- int expire_value = Manager::instance().getRegistrationExpireValue();
- _debug("SIP Registration Expire Value = %i\n" , expire_value);
+ account->setCredInfo(cred);
- setRegistrationState(Trying);
+ status = pjsip_regc_register(_regc, PJ_TRUE, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to register regc.\n");
+ _mutexSIP.leaveMutex();
+ return false;
+ }
+
+ status = pjsip_regc_send(_regc, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send regc request.\n");
+ _mutexSIP.leaveMutex();
+ return false;
+ }
- return Manager::instance().getUserAgent()->addAccount(id, &_regc, getHostname(), getUsername(), getPassword(), expire_value);
+ account->setUserName(username);
+ account->setServer(hostname);
+ account->setContact(contactTmp);
+
+ _mutexSIP.leaveMutex();
+
+ return true;
}
-std::string
+
+ std::string
SIPVoIPLink::SIPFromHeader(const std::string& userpart, const std::string& hostpart)
{
- return ("\"" + getFullName() + "\"" + " <sip:" + userpart + "@" + hostpart + ">");
+ return ("\"" + getFullName() + "\"" + " <sip:" + userpart + "@" + hostpart + ">");
}
-bool
+ bool
SIPVoIPLink::sendSIPAuthentification()
{
- if (getUsername().empty()) {
- /** @todo Ajouter ici un call à setRegistrationState(Error, "Fill balh") ? */
- return false;
- }
- if (getPassword().empty()) {
- /** @todo Même chose ici ? */
- return false;
- }
+ if (getUsername().empty()) {
+ /** @todo Ajouter ici un call à setRegistrationState(Error, "Fill balh") ? */
+ return false;
+ }
+ if (getPassword().empty()) {
+ /** @todo Même chose ici ? */
+ return false;
+ }
- return true;
+ return true;
}
-int
+ int
SIPVoIPLink::sendUnregister()
{
- _debug("SEND UNREGISTER for account %s\n" , getAccountID().c_str());
+ pj_status_t status = 0;
+ pjsip_tx_data *tdata = NULL;
+
+ _debug("SEND UNREGISTER for account %s\n" , getAccountID().c_str());
- if(!_bRegister){
- setRegistrationState(VoIPLink::Unregistered);
- return true;
- }
-
- _bRegister = false;
-
- Manager::instance().getUserAgent()->removeAccount(_regc);
-
- return true;
+ if(!isRegister()){
+ setRegistrationState(VoIPLink::Unregistered);
+ return true;
+ }
+
+ if(_regc) {
+ status = pjsip_regc_unregister(_regc, &tdata);
+ if(status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to unregister regc.\n");
+ return false;
+ }
+
+ status = pjsip_regc_send( _regc, tdata );
+ if(status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send regc request.\n");
+ return false;
+ }
+ } else {
+ _debug("UserAgent: regc is null!\n");
+ return false;
+ }
+
+ setRegister(false);
+
+ return true;
}
-Call*
+ Call*
SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
- SIPCall* call = new SIPCall(id, Call::Outgoing);
- if (call) {
- //call->setPeerNumber(toUrl);
- call->setPeerNumber(getSipTo(toUrl));
- _debug("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
- // we have to add the codec before using it in SIPOutgoingInvite...
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- if ( SIPOutgoingInvite(call) ) {
- call->setConnectionState(Call::Progressing);
- call->setState(Call::Active);
- addCall(call);
- } else {
- delete call; call = 0;
+ SIPCall* call = new SIPCall(id, Call::Outgoing);
+ if (call) {
+ //call->setPeerNumber(toUrl);
+ call->setPeerNumber(getSipTo(toUrl));
+ _debug("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
+ // we have to add the codec before using it in SIPOutgoingInvite...
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ if ( SIPOutgoingInvite(call) ) {
+ call->setConnectionState(Call::Progressing);
+ call->setState(Call::Active);
+ addCall(call);
+ } else {
+ delete call; call = 0;
+ }
}
- }
- return call;
+ return call;
}
-bool
+ bool
SIPVoIPLink::answer(const CallID& id)
{
- _debug("- SIP Action: start answering\n");
- SIPCall* call = getSIPCall(id);
- if (call==0) {
- _debug("! SIP Failure: SIPCall doesn't exists\n");
- return false;
- }
-
- int i = Manager::instance().getUserAgent()->answer(call);
-
- if (i != 0) {
- _debug("< SIP Building Error: send 400 Bad Request\n");
- } else {
- // use exosip, bug locked
- i = 0;
- _debug("* SIP Info: Starting AudioRTP when answering\n");
- if (_audiortp->createNewSession(call) >= 0) {
- call->setAudioStart(true);
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
- return true;
- } else {
- _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ int i;
+ SIPCall *call;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
+ _debug("- SIP Action: start answering\n");
+
+ call = getSIPCall(id);
+
+ if (call==0) {
+ _debug("! SIP Failure: SIPCall doesn't exists\n");
+ return false;
+ }
+
+ // User answered the incoming call, tell peer this news
+ if (call->startNegociation(_pool)) {
+ // Create and send a 200(OK) response
+ _debug("UserAgent: Negociation success!\n");
+ status = pjsip_inv_answer(call->getInvSession(), PJSIP_SC_OK, NULL, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
+ if (_audiortp->createNewSession(call) >= 0) {
+ call->setAudioStart(true);
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+ return true;
+ } else {
+ _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ }
}
- }
- removeCall(call->getCallId());
- return false;
+ removeCall(call->getCallId());
+ return false;
}
-bool
+ bool
SIPVoIPLink::hangup(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
+ pj_status_t status;
+ pjsip_tx_data *tdata = NULL;
+ SIPCall* call;
+
+ call = getSIPCall(id);
+
if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- if(!Manager::instance().getUserAgent()->hangup(call))
+ // User hangup current call. Notify peer
+ status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
+ if(status != PJ_SUCCESS)
return false;
-
+
+ if(tdata == NULL)
+ return true;
+
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ if(status != PJ_SUCCESS)
+ return false;
+
+ call->getInvSession()->mod_data[getModId()] = NULL;
+ return true;
+
// Release RTP thread
if (Manager::instance().isCurrentCall(id)) {
_debug("* SIP Info: Stopping AudioRTP for hangup\n");
_audiortp->closeRtpSession();
}
-
+
removeCall(id);
-
+
return true;
}
-bool
+ bool
SIPVoIPLink::cancel(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- _debug("- SIP Action: Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
+ _debug("- SIP Action: Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
- removeCall(id);
+ removeCall(id);
- return true;
+ return true;
}
-bool
+ bool
SIPVoIPLink::onhold(const CallID& id)
{
- SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_attr *attr;
+ pjmedia_sdp_session* local_sdp;
+ SIPCall* call;
- // Stop sound
- call->setAudioStart(false);
- call->setState(Call::Hold);
- _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
- _audiortp->closeRtpSession();
+ call = getSIPCall(id);
+
+ if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
- return Manager::instance().getUserAgent()->onhold(call);
+ // Stop sound
+ call->setAudioStart(false);
+ call->setState(Call::Hold);
+ _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
+ _audiortp->closeRtpSession();
+ local_sdp = call->getLocalSDPSession();
+ if( local_sdp == NULL ){
+ _debug("! SIP Failure: unable to find local_sdp\n");
+ return false;
+ }
+
+ /* Create re-INVITE with new offer */
+ // Remove all the attributes with the specified name
+ pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendrecv");
+ attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
+ pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
+
+ status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp, &tdata);
+ if( status != PJ_SUCCESS )
+ {
+ _debug("On hold: creation of the Re-invite request failed\n");
+ return false;
+ }
+ /* Send the request */
+ status = pjsip_inv_send_msg( call->getInvSession(), tdata);
+
+ return (status == PJ_SUCCESS);
}
-bool
+ bool
SIPVoIPLink::offhold(const CallID& id)
{
SIPCall *call;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_attr *attr;
+ pjmedia_sdp_session* local_sdp;
call = getSIPCall(id);
+
if (call==0) {
_debug("! SIP Error: Call doesn't exist\n");
return false;
}
- if(!Manager::instance().getUserAgent()->offhold(call))
+ local_sdp = call->getLocalSDPSession();
+ if( local_sdp == NULL ){
+ _debug("! SIP Failure: unable to find local_sdp\n");
+ return false;
+ }
+
+ /* Create re-INVITE with new offer */
+ // Remove all the attributes with the specified name
+ pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendonly");
+ attr = pjmedia_sdp_attr_create(_pool, "sendrecv", NULL);
+ pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
+
+ status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp , &tdata);
+ if( status != PJ_SUCCESS )
+ {
+ _debug("Off hold: creation of the Re-invite request failed\n");
+ return false;
+ }
+
+ /* Send the request */
+ status = pjsip_inv_send_msg( call->getInvSession(), tdata);
+ if( status != PJ_SUCCESS )
return false;
// Enable audio
@@ -287,17 +531,23 @@ SIPVoIPLink::offhold(const CallID& id)
_debug("! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
return false;
}
-
+
return true;
}
-bool
+ bool
SIPVoIPLink::transfer(const CallID& id, const std::string& to)
{
SIPCall *call;
std::string tmp_to;
+ pjsip_evsub *sub;
+ pjsip_tx_data *tdata;
+ struct pjsip_evsub_user xfer_cb;
+ pj_status_t status;
+ pj_str_t dest;
call = getSIPCall(id);
+
if (call==0) {
_debug("! SIP Failure: Call doesn't exist\n");
return false;
@@ -310,11 +560,43 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
_debug("In transfer, tmp_to is %s\n", tmp_to.data());
- return Manager::instance().getUserAgent()->transfer(call, tmp_to);
+ pj_strdup2(_pool, &dest, to.data());
+
+ /* Create xfer client subscription. */
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_func_cb;
+
+ status = pjsip_xfer_create_uac(call->getInvSession()->dlg, &xfer_cb, &sub);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create xfer -- %d\n", status);
+ return false;
+ }
- //_audiortp->closeRtpSession();
- // shall we delete the call?
- //removeCall(id);
+ /* Associate this voiplink of call with the client subscription
+ * We can not just associate call with the client subscription
+ * because after this function, we can not find the cooresponding
+ * voiplink from the call any more. But the voiplink is useful!
+ */
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ pjsip_evsub_set_mod_data(sub, getModId(), this);
+
+ /*
+ * Create REFER request.
+ */
+ status = pjsip_xfer_initiate(sub, &dest, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create REFER request -- %d\n", status);
+ return false;
+ }
+
+ /* Send. */
+ status = pjsip_xfer_send_request(sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send REFER request -- %d\n", status);
+ return false;
+ }
+
+ return true;
}
bool SIPVoIPLink::transferStep2()
@@ -323,10 +605,13 @@ bool SIPVoIPLink::transferStep2()
return true;
}
-bool
+ bool
SIPVoIPLink::refuse (const CallID& id)
{
SIPCall *call;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
call = getSIPCall(id);
@@ -341,10 +626,20 @@ SIPVoIPLink::refuse (const CallID& id)
return false;
}
- return Manager::instance().getUserAgent()->refuse(call);
+ // User refuse current call. Notify peer
+ status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata); //603
+ if(status != PJ_SUCCESS)
+ return false;
+
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ if(status != PJ_SUCCESS)
+ return false;
+
+ call->getInvSession()->mod_data[getModId()] = NULL;
+ return true;
}
-bool
+ bool
SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
@@ -352,8 +647,14 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
int duration;
const int body_len = 1000;
char *dtmf_body;
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pj_str_t methodName, content;
+ pjsip_method method;
+ pjsip_media_type ctype;
call = getSIPCall(id);
+
if (call==0) {
_debug("Call doesn't exist\n");
return false;
@@ -361,210 +662,1443 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
dtmf_body = new char[body_len];
-
+
snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
-
- return Manager::instance().getUserAgent()->carryingDTMFdigits(call, dtmf_body);
+
+ pj_strdup2(_pool, &methodName, "INFO");
+ pjsip_method_init_np(&method, &methodName);
+
+ /* Create request message. */
+ status = pjsip_dlg_create_request( call->getInvSession()->dlg, &method, -1, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create INFO request -- %d\n", status);
+ return false;
+ }
+
+ /* Get MIME type */
+ pj_strdup2(_pool, &ctype.type, "application");
+ pj_strdup2(_pool, &ctype.subtype, "dtmf-relay");
+
+ /* Create "application/dtmf-relay" message body. */
+ pj_strdup2(_pool, &content, dtmf_body);
+ tdata->msg->body = pjsip_msg_body_create( tdata->pool, &ctype.type, &ctype.subtype, &content);
+ if (tdata->msg->body == NULL) {
+ _debug("UserAgent: Unable to create msg body!\n");
+ pjsip_tx_data_dec_ref(tdata);
+ return false;
+ }
+
+ /* Send the request. */
+ status = pjsip_dlg_send_request( call->getInvSession()->dlg, tdata, getModId(), NULL);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send MESSAGE request -- %d\n", status);
+ return false;
+ }
+
+ return true;
}
-bool
+ bool
SIPVoIPLink::SIPOutgoingInvite(SIPCall* call)
{
- // If no SIP proxy setting for direct call with only IP address
- if (!SIPStartCall(call, "")) {
- _debug("! SIP Failure: call not started\n");
- return false;
- }
- return true;
+ // If no SIP proxy setting for direct call with only IP address
+ if (!SIPStartCall(call, "")) {
+ _debug("! SIP Failure: call not started\n");
+ return false;
+ }
+ return true;
}
-bool
+ bool
SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject UNUSED)
{
- std::string to;
+ std::string strTo, strFrom;
+ pj_status_t status;
+ pjsip_dialog *dialog;
+ pjsip_tx_data *tdata;
+ pj_str_t from, to, contact;
+ AccountID id;
+ SIPAccount *account;
if (!call)
return false;
- to = getSipTo(call->getPeerNumber());
- _debug(" To: %s\n", to.data());
+ strTo = getSipTo(call->getPeerNumber());
+ _debug(" To: %s\n", strTo.data());
+
+ id = getAccountID();
+ // Get the basic information about the callee account
+ account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
+
+ // Generate the from URI
+ strFrom = "sip:" + account->getUserName() + "@" + account->getServer();
+
+ // pjsip need the from and to information in pj_str_t format
+ pj_strdup2(_pool, &from, strFrom.data());
+ pj_strdup2(_pool, &to, strTo.data());
+ pj_strdup2(_pool, &contact, account->getContact().data());
+
+ // create the dialog (UAC)
+ status = pjsip_dlg_create_uac(pjsip_ua_instance(), &from,
+ &contact,
+ &to,
+ NULL,
+ &dialog);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ setCallAudioLocal(call);
+ call->setIp(getLocalIP());
+
+ // Building the local SDP offer
+ call->createInitialOffer(_pool);
+
+ // Create the invite session for this call
+ pjsip_inv_session *inv;
+ status = pjsip_inv_create_uac(dialog, call->getLocalSDPSession(), 0, &inv);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
- return Manager::instance().getUserAgent()->makeOutgoingCall(to, call, getAccountID());
+ // Set auth information
+ pjsip_auth_clt_set_credentials(&dialog->auth_sess, 1, account->getCredInfo());
+
+ // Associate current call in the invite session
+ inv->mod_data[getModId()] = call;
+
+ status = pjsip_inv_invite(inv, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ // Associate current invite session in the call
+ call->setInvSession(inv);
+
+ status = pjsip_inv_send_msg(inv, tdata);
+ if(status != PJ_SUCCESS) {
+ return false;
+ }
+
+ return true;
}
std::string
SIPVoIPLink::getSipFrom() {
- // Form the From header field basis on configuration panel
- std::string hostname;
-
- hostname = getHostname();
+ // Form the From header field basis on configuration panel
+ std::string hostname;
- if ( hostname.empty() ) {
- hostname = _localIPAddress;
- }
- return SIPFromHeader(getUsername(), hostname);
+ hostname = getHostname();
+
+ if ( hostname.empty() ) {
+ hostname = _localIPAddress;
+ }
+ return SIPFromHeader(getUsername(), hostname);
}
-std::string
-SIPVoIPLink::getSipTo(const std::string& to_url) {
- // Form the From header field basis on configuration panel
- //bool isRegistered = (_eXosipRegID == EXOSIP_ERROR_STD) ? false : true;
+std::string SIPVoIPLink::getSipTo(const std::string& to_url) {
+ // Form the From header field basis on configuration panel
+ //bool isRegistered = (_eXosipRegID == EXOSIP_ERROR_STD) ? false : true;
- // add a @host if we are registered and there is no one inside the url
+ // add a @host if we are registered and there is no one inside the url
if (to_url.find("@") == std::string::npos) {// && isRegistered) {
- std::string host = getHostname();
- if(!host.empty()) {
- return SIPToHeader(to_url + "@" + host);
+ std::string host = getHostname();
+ if(!host.empty()) {
+ return SIPToHeader(to_url + "@" + host);
+ }
+ }
+ return SIPToHeader(to_url);
}
- }
- return SIPToHeader(to_url);
-}
-std::string
-SIPVoIPLink::SIPToHeader(const std::string& to)
+ std::string SIPVoIPLink::SIPToHeader(const std::string& to)
+ {
+ if (to.find("sip:") == std::string::npos) {
+ return ("sip:" + to );
+ } else {
+ return to;
+ }
+ }
+
+ bool
+ SIPVoIPLink::SIPCheckUrl(const std::string& url UNUSED)
+ {
+ return true;
+ }
+
+bool setCallAudioLocal(SIPCall* call)
{
- if (to.find("sip:") == std::string::npos) {
- return ("sip:" + to );
- } else {
- return to;
- }
+ // Setting Audio
+ unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
+ unsigned int callLocalExternAudioPort = callLocalAudioPort;
+ if (_useStun) {
+ // If use Stun server
+ if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
+ callLocalExternAudioPort = Manager::instance().getFirewallPort();
+ }
+ }
+ _debug(" Setting local audio port to: %d\n", callLocalAudioPort);
+ _debug(" Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
+
+ // Set local audio port for SIPCall(id)
+ call->setLocalIp(_localIPAddress);
+ call->setLocalAudioPort(callLocalAudioPort);
+ call->setLocalExternAudioPort(callLocalExternAudioPort);
+
+ return true;
+ }
+
+ void
+ SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
+ {
+ //if (!event->response) { return; }
+ //switch(event->response->status_code) {
+ //case SIP_SERVICE_UNAVAILABLE: // 500
+ //case SIP_BUSY_EVRYWHERE: // 600
+ //case SIP_DECLINE: // 603
+ //SIPCall* call = findSIPCallWithCid(event->cid);
+ if (call != 0) {
+ _debug("Server error!\n");
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ }
+ //break;
+ //}
+ }
+
+ void
+ SIPVoIPLink::SIPCallClosed(SIPCall *call)
+ {
+ // it was without did before
+ //SIPCall* call = findSIPCallWithCid(event->cid);
+ if (!call) { return; }
+
+ CallID id = call->getCallId();
+ //call->setDid(event->did);
+ if (Manager::instance().isCurrentCall(id)) {
+ call->setAudioStart(false);
+ _debug("* SIP Info: Stopping AudioRTP when closing\n");
+ _audiortp->closeRtpSession();
+ }
+ _debug("After close RTP\n");
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ _debug("After remove call ID\n");
+ }
+
+ void
+ SIPVoIPLink::SIPCallReleased(SIPCall *call)
+ {
+ // do cleanup if exists
+ // only cid because did is always 0 in these case..
+ //SIPCall* call = findSIPCallWithCid(event->cid);
+ if (!call) { return; }
+
+ // if we are here.. something when wrong before...
+ _debug("SIP call release\n");
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ }
+
+ void
+ SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
+ {
+ //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
+ if (!call) {
+ _debug("! SIP Failure: unknown call\n");
+ return;
+ }
+ //call->setDid(event->did);
+
+ if (call->getConnectionState() != Call::Connected) {
+ //call->SIPCallAnswered(event);
+ call->SIPCallAnsweredWithoutHold(rdata);
+
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ Manager::instance().peerAnsweredCall(call->getCallId());
+ if (Manager::instance().isCurrentCall(call->getCallId())) {
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
+ if ( _audiortp->createNewSession(call) < 0) {
+ _debug("RTP Failure: unable to create new session\n");
+ } else {
+ call->setAudioStart(true);
+ }
+ }
+ } else {
+ _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
+ //call->SIPCallAnswered(event);
+ }
+ }
+
+
+ SIPCall*
+ SIPVoIPLink::getSIPCall(const CallID& id)
+ {
+ Call* call = getCall(id);
+ if (call) {
+ return dynamic_cast<SIPCall*>(call);
+ }
+ return NULL;
+ }
+
+ ///////////////////////////////////////////////////////////////////////////////
+ // Private functions
+ ///////////////////////////////////////////////////////////////////////////////
+
+ pj_str_t SIPVoIPLink::string2PJStr(const std::string &value)
+ {
+ char tmp[256];
+
+ strcpy(tmp, value.data());
+ return pj_str(tmp);
+ }
+
+
+ bool SIPVoIPLink::pjsip_init()
+ {
+ pj_status_t status;
+ int errPjsip = 0;
+ int port;
+ pjsip_inv_callback inv_cb;
+ pj_str_t accepted;
+ std::string name_mod;
+
+ validStunServer = true;
+
+ name_mod = "sflphone";
+
+ // Init PJLIB: must be called before any call to the pjsip library
+ status = pj_init();
+ // Use pjsip macros for sanity check
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init PJLIB-UTIL library
+ status = pjlib_util_init();
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Set the pjsip log level
+ pj_log_set_level( PJ_LOG_LEVEL );
+
+ // Init PJNATH
+ status = pjnath_init();
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Create a pool factory to allocate memory
+ pj_caching_pool_init(&_cp, &pj_pool_factory_default_policy, 0);
+
+ // Create memory pool for application.
+ _pool = pj_pool_create(&_cp.factory, "sflphone", 4000, 4000, NULL);
+
+ if (!_pool) {
+ _debug("UserAgent: Could not initialize memory pool\n");
+ return PJ_ENOMEM;
+ }
+
+ // Create the SIP endpoint
+ status = pjsip_endpt_create(&_cp.factory, pj_gethostname()->ptr, &_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ /* Start resolving STUN server */
+ // if we useStun and we failed to receive something on port 5060, we try a random port
+ // If use STUN server, firewall address setup
+ if (!loadSIPLocalIP()) {
+ _debug("UserAgent: Unable to determine network capabilities\n");
+ return false;
+ }
+
+ port = _regPort;
+
+ if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
+ port = RANDOM_SIP_PORT;
+ if (!Manager::instance().behindNat(_stunServer, port)) {
+ _debug("UserAgent: Unable to check NAT setting\n");
+ validStunServer = false;
+ return false; // hoho we can't use the random sip port too...
+ }
+ }
+
+ _localPort = port;
+ if (_useStun) {
+ // set by last behindNat() call (ish)...
+ stunServerResolve();
+ _localExternAddress = Manager::instance().getFirewallAddress();
+ _localExternPort = Manager::instance().getFirewallPort();
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("UserAgent: Could not initialize SIP listener on port %d\n", port);
+ return errPjsip;
+ }
+ } else {
+ _localExternAddress = _localIPAddress;
+ _localExternPort = _localPort;
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("UserAgent: Could not initialize SIP listener on port %d\n", _localExternPort);
+ _localExternPort = _localPort = RANDOM_SIP_PORT;
+ _debug("UserAgent: Try to initialize SIP listener on port %d\n", _localExternPort);
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("UserAgent: Fail to initialize SIP listener on port %d\n", _localExternPort);
+ return errPjsip;
+ }
+ }
+ }
+
+ _debug("UserAgent: SIP Init -- listening on port %d\n", _localExternPort);
+
+ // Initialize transaction layer
+ status = pjsip_tsx_layer_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize UA layer module
+ status = pjsip_ua_init_module(_endpt, NULL);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize Replaces support. See the Replaces specification in RFC 3891
+ status = pjsip_replaces_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize 100rel support
+ status = pjsip_100rel_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Initialize and register sflphone module
+ _mod_ua.name = pj_str((char*)name_mod.c_str());
+ _mod_ua.id = -1;
+ _mod_ua.priority = PJSIP_MOD_PRIORITY_APPLICATION;
+ _mod_ua.on_rx_request = &mod_on_rx_request;
+ _mod_ua.on_rx_response = &mod_on_rx_response;
+
+ status = pjsip_endpt_register_module(_endpt, &_mod_ua);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init the event subscription module.
+ // It extends PJSIP by supporting SUBSCRIBE and NOTIFY methods
+ status = pjsip_evsub_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init presence module.
+ // TODO We probably do not need that extension
+ status = pjsip_pres_init_module(_endpt, pjsip_evsub_instance());
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init PUBLISH module
+ // Provide an implementation of SIP Extension for Event State Publication (RFC 3903)
+ // TODO Check if it is necessary
+ status = pjsip_publishc_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init xfer/REFER module
+ status = pjsip_xfer_init_module(_endpt);
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
+
+ // Init the callback for INVITE session:
+ // TODO The invite session as global ?
+ pj_bzero(&inv_cb, sizeof (inv_cb));
+
+ inv_cb.on_state_changed = &call_on_state_changed;
+ inv_cb.on_new_session = &call_on_forked;
+ inv_cb.on_media_update = &call_on_media_update;
+ inv_cb.on_tsx_state_changed = &call_on_tsx_changed;
+
+ // Initialize session invite module
+ status = pjsip_inv_usage_init(_endpt, &inv_cb);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ _debug("UserAgent: VOIP callbacks initialized\n");
+
+ // Add endpoint capabilities (INFO, OPTIONS, etc) for this UA
+ pj_str_t allowed[] = { {(char*)"INFO", 4}, {(char*)"REGISTER", 8} }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}, {"OPTIONS", 7},
+ accepted = pj_str((char*)"application/sdp");
+
+ // Register supported methods
+ pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
+
+ // Register "application/sdp" in ACCEPT header
+ pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ACCEPT, NULL, 1, &accepted);
+
+ _debug("UserAgent: pjsip version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
+
+ //TODO Create the secondary thread to poll sip events
+ _evThread->start();
+
+ /* Done! */
+ return PJ_SUCCESS;
+ }
+
+ pj_status_t SIPVoIPLink::stunServerResolve( void )
+ {
+ pj_str_t stun_adr;
+ pj_hostent he;
+ pj_stun_config stunCfg;
+ pj_status_t stun_status;
+ pj_sockaddr stun_srv;
+ size_t pos;
+ std::string serverName, serverPort;
+ int nPort;
+
+ // Initialize STUN configuration
+ pj_stun_config_init(&stunCfg, &_cp.factory, 0, pjsip_endpt_get_ioqueue(_endpt), pjsip_endpt_get_timer_heap(_endpt));
+
+ stun_status = PJ_EPENDING;
+
+ // Init STUN socket
+ pos = _stunServer.find(':');
+ if(pos == std::string::npos) {
+ pj_strdup2(_pool, &stun_adr, _stunServer.data());
+ stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) 3478);
+ } else {
+ serverName = _stunServer.substr(0, pos);
+ serverPort = _stunServer.substr(pos + 1);
+ nPort = atoi(serverPort.data());
+ pj_strdup2(_pool, &stun_adr, serverName.data());
+ stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) nPort);
+ }
+
+ if (stun_status != PJ_SUCCESS) {
+ _debug("UserAgent: Unresolved stun server!\n");
+ stun_status = pj_gethostbyname(&stun_adr, &he);
+
+ if (stun_status == PJ_SUCCESS) {
+ pj_sockaddr_in_init(&stun_srv.ipv4, NULL, 0);
+ stun_srv.ipv4.sin_addr = *(pj_in_addr*) he.h_addr;
+ stun_srv.ipv4.sin_port = pj_htons((pj_uint16_t) 3478);
+ }
+ }
+
+ return stun_status;
+ }
+
+ int SIPVoIPLink::createUDPServer( void )
+ {
+
+ pj_status_t status;
+ pj_sockaddr_in bound_addr;
+ pjsip_host_port a_name;
+ char tmpIP[32];
+ pj_sock_t sock;
+
+ // Init bound address to ANY
+ pj_memset(&bound_addr, 0, sizeof (bound_addr));
+ bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
+
+ // Create UDP server socket
+ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &sock);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: (%d) UDP socket() error\n", status);
+ return status;
+ }
+
+ status = pj_sock_bind_in(sock, pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t) _localPort);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: (%d) UDP bind() error\n", status);
+ pj_sock_close(sock);
+ return status;
+ }
+
+ _debug("UserAgent: Use IP: %s\n", _localExternAddress.data());
+
+ // Create UDP-Server (default port: 5060)
+ strcpy(tmpIP, _localExternAddress.data());
+ pj_strdup2(_pool, &a_name.host, tmpIP);
+ a_name.port = (pj_uint16_t) _localExternPort;
+
+ _debug("a_name: host: %s - port : %i\n", a_name.host.ptr, a_name.port);
+
+ status = pjsip_udp_transport_attach(_endpt, sock, &a_name, 1, NULL);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: (%d) Unable to start UDP transport!\n", status);
+ return -1;
+ } else {
+ _debug("UserAgent: UDP server listening on port %d\n", _localExternPort);
+ }
+
+ return PJ_SUCCESS;
+ }
+
+ bool SIPVoIPLink::loadSIPLocalIP() {
+
+ bool returnValue = true;
+
+ if (_localIPAddress == "127.0.0.1") {
+ pj_sockaddr ip_addr;
+ if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
+ // Update the registration state if no network capabilities found
+ _debug("UserAgent: Get host ip failed!\n");
+ returnValue = false;
+ } else {
+ _localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
+ _debug("UserAgent: Checking network, setting local IP address to: %s\n", _localIPAddress.data());
+ }
+ }
+ return returnValue;
+ }
+
+ bool SIPVoIPLink::pjsip_shutdown( void )
+ {
+ /* Destroy endpoint. */
+ if (_endpt) {
+ pjsip_endpt_destroy(_endpt);
+ _endpt = NULL;
+ }
+
+ /* Destroy pool and pool factory. */
+ if (_pool) {
+ pj_pool_release(_pool);
+ _pool = NULL;
+ pj_caching_pool_destroy(&_cp);
+ }
+
+ /* Shutdown PJLIB */
+ pj_shutdown();
+
+ /* Done. */
+ }
+
+ int SIPVoIPLink::getModId(){
+ return _mod_ua.id;
+ }
+
+void set_voicemail_info( AccountID account, pjsip_msg_body *body ){
+
+ int voicemail, pos_begin, pos_end;
+ std::string voice_str = "Voice-Message: ";
+ std::string delimiter = "/";
+ std::string msg_body, voicemail_str;
+
+ _debug("UserAgent: checking the voice message!\n");
+ // The voicemail message is formated like that:
+ // Voice-Message: 1/0 . 1 is the number we want to retrieve in this case
+
+ // We get the notification body
+ msg_body = (char*)body->data;
+
+ // We need the position of the first character of the string voice_str
+ pos_begin = msg_body.find(voice_str);
+ // We need the position of the delimiter
+ pos_end = msg_body.find(delimiter);
+
+ // So our voicemail number between the both index
+ try {
+
+ voicemail_str = msg_body.substr(pos_begin + voice_str.length(), pos_end - ( pos_begin + voice_str.length()));
+ std::cout << "voicemail number : " << voicemail_str << std::endl;
+ voicemail = atoi( voicemail_str.c_str() );
+ }
+ catch( std::out_of_range& e ){
+ std::cerr << e.what() << std::endl;
+ }
+
+ // We need now to notify the manager
+ if( voicemail != 0 )
+ Manager::instance().startVoiceMessageNotification(account, voicemail);
}
-bool
-SIPVoIPLink::SIPCheckUrl(const std::string& url UNUSED)
-{
- return true;
+bool SIPVoIPLink::useStun( void ){
+ return _useStun;
}
-bool
-SIPVoIPLink::setCallAudioLocal(SIPCall* call)
+void SIPVoIPLink::setUseStun( bool use )
{
- // Setting Audio
- unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
- unsigned int callLocalExternAudioPort = callLocalAudioPort;
- if (_useStun) {
- // If use Stun server
- if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
- callLocalExternAudioPort = Manager::instance().getFirewallPort();
- }
- }
- _debug(" Setting local audio port to: %d\n", callLocalAudioPort);
- _debug(" Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
-
- // Set local audio port for SIPCall(id)
- call->setLocalIp(_localIPAddress);
- call->setLocalAudioPort(callLocalAudioPort);
- call->setLocalExternAudioPort(callLocalExternAudioPort);
-
- return true;
-}
-
-void
-SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
-{
- //if (!event->response) { return; }
- //switch(event->response->status_code) {
- //case SIP_SERVICE_UNAVAILABLE: // 500
- //case SIP_BUSY_EVRYWHERE: // 600
- //case SIP_DECLINE: // 603
- //SIPCall* call = findSIPCallWithCid(event->cid);
- if (call != 0) {
- _debug("Server error!\n");
- CallID id = call->getCallId();
- Manager::instance().callFailure(id);
- removeCall(id);
- }
- //break;
- //}
-}
-
-void
-SIPVoIPLink::SIPCallClosed(SIPCall *call)
-{
- // it was without did before
- //SIPCall* call = findSIPCallWithCid(event->cid);
- if (!call) { return; }
+ _useStun = use;
+}
- CallID id = call->getCallId();
- //call->setDid(event->did);
- if (Manager::instance().isCurrentCall(id)) {
- call->setAudioStart(false);
- _debug("* SIP Info: Stopping AudioRTP when closing\n");
- _audiortp->closeRtpSession();
- }
- _debug("After close RTP\n");
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- _debug("After remove call ID\n");
+
+ /*******************************/
+ /* CALLBACKS IMPLEMENTATION */
+ /*******************************/
+
+void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e){
+
+ PJ_UNUSED_ARG(inv);
+
+ SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
+ if(!call)
+ return;
+
+ /* If this is an outgoing INVITE that was created because of
+ * REFER/transfer, send NOTIFY to transferer.
+ */
+ if (call->getXferSub() && e->type==PJSIP_EVENT_TSX_STATE) {
+ int st_code = -1;
+ pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
+
+ switch (call->getInvSession()->state) {
+ case PJSIP_INV_STATE_NULL:
+ case PJSIP_INV_STATE_CALLING:
+ /* Do nothing */
+ break;
+
+ case PJSIP_INV_STATE_EARLY:
+ case PJSIP_INV_STATE_CONNECTING:
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_ACTIVE;
+ break;
+
+ case PJSIP_INV_STATE_CONFIRMED:
+ /* When state is confirmed, send the final 200/OK and terminate
+ * subscription.
+ */
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ break;
+
+ case PJSIP_INV_STATE_DISCONNECTED:
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ break;
+
+ case PJSIP_INV_STATE_INCOMING:
+ /* Nothing to do. Just to keep gcc from complaining about
+ * unused enums.
+ */
+ break;
+ }
+
+ if (st_code != -1) {
+ pjsip_tx_data *tdata;
+ pj_status_t status;
+
+ status = pjsip_xfer_notify( call->getXferSub(),
+ ev_state, st_code,
+ NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create NOTIFY -- %d\n", status);
+ } else {
+ status = pjsip_xfer_send_request(call->getXferSub(), tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send NOTIFY -- %d\n", status);
+ }
+ }
+ }
+ }
}
-void
-SIPVoIPLink::SIPCallReleased(SIPCall *call)
-{
- // do cleanup if exists
- // only cid because did is always 0 in these case..
- //SIPCall* call = findSIPCallWithCid(event->cid);
- if (!call) { return; }
+ void call_on_media_update( pjsip_inv_session *inv UNUSED, pj_status_t status UNUSED) {
+ _debug("call_on_media_updated\n");
+ }
+
+ void call_on_forked(pjsip_inv_session *inv, pjsip_event *e){
+ _debug("call_on_forked\n");
+ }
+
+void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e){
- // if we are here.. something when wrong before...
- _debug("SIP call release\n");
- CallID id = call->getCallId();
- Manager::instance().callFailure(id);
- removeCall(id);
+ pjsip_rx_data *rdata;
+ AccountID accId;
+ SIPCall *call;
+ SIPVoIPLink *link;
+ pjsip_msg *msg;
+
+ _debug("UserAgent: TSX Changed! The tsx->state is %d; tsx->role is %d; code is %d; method id is %.*s.\n",
+ tsx->state, tsx->role, tsx->status_code, (int)tsx->method.name.slen, tsx->method.name.ptr);
+
+ if(pj_strcmp2(&tsx->method.name, "INFO") == 0) {
+ // Receive a INFO message, ingore it!
+ return;
+ }
+
+ //Retrieve the body message
+ rdata = e->body.tsx_state.src.rdata;
+
+ if (tsx->role == PJSIP_ROLE_UAC) {
+ switch (tsx->state) {
+ case PJSIP_TSX_STATE_TERMINATED:
+ if (tsx->status_code == 200 &&
+ pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) != 0) {
+ // Peer answered the outgoing call
+ _debug("UserAgent: Peer answered the outgoing call!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL)
+ return;
+
+ //_debug("UserAgent: The call id is %s\n", call->getCallId().data());
+
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link)
+ link->SIPCallAnswered(call, rdata);
+ } else if (tsx->status_code / 100 == 5) {
+ _debug("UserAgent: 5xx error message received\n");
+ }
+ break;
+ case PJSIP_TSX_STATE_PROCEEDING:
+ // Peer is ringing for the outgoing call
+ msg = rdata->msg_info.msg;
+
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL)
+ return;
+
+ if (msg->line.status.code == 180) {
+ _debug("UserAgent: Peer is ringing!\n");
+
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ }
+ break;
+ case PJSIP_TSX_STATE_COMPLETED:
+ if (tsx->status_code == 407 || tsx->status_code == 401) //FIXME
+ break;
+ if (tsx->status_code / 100 == 6 || tsx->status_code / 100 == 4) {
+ // We get error message of outgoing call from server
+ _debug("UserAgent: Server error message is received!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL) {
+ _debug("UserAgent: Call has been removed!\n");
+ return;
+ }
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallServerFailure(call);
+ }
+ }
+ break;
+ default:
+ break;
+ } // end of switch
+
+ } else {
+ switch (tsx->state) {
+ case PJSIP_TSX_STATE_TRYING:
+ if (pjsip_method_cmp(&tsx->method, pjsip_get_refer_method()) == 0) {
+ // Peer ask me to transfer call to another number.
+ _debug("UserAgent: Incoming REFER request!\n");
+ //onCallTransfered(inv, e->body.tsx_state.src.rdata);
+ }
+ break;
+ case PJSIP_TSX_STATE_COMPLETED:
+ if (tsx->status_code == 200 && tsx->method.id == PJSIP_BYE_METHOD) {
+ // Peer hangup the call
+ _debug("UserAgent: Peer hangup(bye) message is received!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL) {
+ _debug("UserAgent: Call has been removed!\n");
+ return;
+ }
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallClosed(call);
+ }
+ } else if (tsx->status_code == 200 && tsx->method.id == PJSIP_CANCEL_METHOD) {
+ // Peer refuse the call
+ _debug("UserAgent: Cancel message is received!\n");
+ call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
+ if (call == NULL) {
+ _debug("UserAgent: Call has been removed!\n");
+ return;
+ }
+
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if (link) {
+ link->SIPCallClosed(call);
+ }
+ }
+ break;
+ default:
+ break;
+ } // end of switch
+ }
}
-void
-SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
+ void regc_cb(struct pjsip_regc_cbparam *param){
+
+ AccountID *id = static_cast<AccountID *> (param->token);
+ SIPVoIPLink *voipLink;
+
+ //_debug("UserAgent: Account ID is %s, Register result: %d, Status: %d\n", id->data(), param->status, param->code);
+ voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(*id));
+ if(!voipLink)
+ return;
+
+ if (param->status == PJ_SUCCESS) {
+ if (param->code < 0 || param->code >= 300) {
+ /* Sometimes, the status is OK, but we still failed.
+ * So checking the code for real result
+ */
+ _debug("UserAgent: The error is: %d\n", param->code);
+ switch(param->code) {
+ case 408:
+ case 606:
+ voipLink->setRegistrationState(VoIPLink::ErrorConfStun);
+ break;
+ case 503:
+ voipLink->setRegistrationState(VoIPLink::ErrorHost);
+ break;
+ case 401:
+ case 403:
+ case 404:
+ voipLink->setRegistrationState(VoIPLink::ErrorAuth);
+ break;
+ default:
+ voipLink->setRegistrationState(VoIPLink::Error);
+ break;
+ }
+ voipLink->setRegister(false);
+ } else {
+ // Registration/Unregistration is success
+
+ if(voipLink->isRegister())
+ voipLink->setRegistrationState(VoIPLink::Registered);
+ else {
+ voipLink->setRegistrationState(VoIPLink::Unregistered);
+ voipLink->setRegister(false);
+ }
+ }
+ } else {
+ voipLink->setRegistrationState(VoIPLink::ErrorAuth);
+ voipLink->setRegister(false);
+ }
+
+ }
+
+ pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata){
+
+ pj_status_t status;
+ pj_str_t reason;
+ unsigned options = 0;
+ pjsip_dialog* dialog;
+ pjsip_tx_data *tdata;
+ //pjmedia_sdp_session *r_sdp;
+ AccountID account_id;
+ pjsip_uri *uri;
+ pjsip_sip_uri *sip_uri;
+ std::string userName, server, caller, callerServer, peerNumber;
+
+ // voicemail part
+ std::string method_name;
+ std::string request;
+
+ // Handle the incoming call invite in this function
+ _debug("UserAgent: Callback on_rx_request is involved!\n");
+
+ /* First, let's got the username and server name from the invite.
+ * We will use them to detect which account is the callee.
+ */
+ uri = rdata->msg_info.to->uri;
+ sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
+
+ userName = std::string(sip_uri->user.ptr, sip_uri->user.slen);
+ server = std::string(sip_uri->host.ptr, sip_uri->host.slen) ;
+
+ // Get the account id of callee from username and server
+ account_id = Manager::instance().getAccountIdFromNameAndServer(userName, server);
+ if(account_id == AccountNULL) {
+ _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
+ return false;
+ }
+
+ _debug("UserAgent: The receiver is : %s@%s\n", userName.data(), server.data());
+ _debug("UserAgent: The callee account id is %s\n", account_id.c_str());
+
+ /* Now, it is the time to find the information of the caller */
+ uri = rdata->msg_info.from->uri;
+ sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
+
+ caller = sip_uri->user.ptr;
+ callerServer = sip_uri->host.ptr;
+ peerNumber = caller + "@" + callerServer;
+
+ // Get the server voicemail notification
+ // Catch the NOTIFY message
+ if( rdata->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD )
+ {
+ method_name = "NOTIFY";
+ // Retrieve all the message. Should contains only the method name but ...
+ request = rdata->msg_info.msg->line.req.method.name.ptr;
+ // Check if the message is a notification
+ if( request.find( method_name ) != (size_t)-1 ) {
+ set_voicemail_info( account_id, rdata->msg_info.msg->body );
+ }
+ pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
+ return true;
+ }
+
+ // Respond statelessly any non-INVITE requests with 500
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
+ pj_strdup2(_pool, &reason, "user agent unable to handle this request ");
+ pjsip_endpt_respond_stateless( _endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, &reason, NULL,
+ NULL);
+ return true;
+ }
+ }
+
+ // Verify that we can handle the request
+ status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, _endpt, NULL);
+ if (status != PJ_SUCCESS) {
+ pj_strdup2(_pool, &reason, "user agent unable to handle this INVITE ");
+ pjsip_endpt_respond_stateless( _endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, &reason, NULL,
+ NULL);
+ return true;
+ }
+
+ // Generate a new call ID for the incoming call!
+ CallID id = Manager::instance().getNewCallID();
+
+ _debug("UserAgent: The call id of the incoming call is %s\n", id.c_str());
+ SIPCall* call = new SIPCall(id, Call::Incoming);
+ if (!call) {
+ _debug("UserAgent: unable to create an incoming call");
+ return false;
+ }
+
+ // Set the codec map, IP, peer number and so on... for the SIPCall object
+ setCallAudioLocal(call);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ call->setConnectionState(Call::Progressing);
+ call->setIp("127.0.0.1");
+ call->setPeerNumber(peerNumber);
+
+ /* Call the SIPCallInvite function to generate the local sdp,
+ * remote sdp and negociator.
+ * This function is also used to set the parameters of audio RTP, including:
+ * local IP and port number
+ * remote IP and port number
+ * possilbe audio codec will be used in this call
+ */
+ if (call->SIPCallInvite(rdata, _pool)) {
+
+ // Notify UI there is an incoming call
+ if (Manager::instance().incomingCall(call, account_id)) {
+ // Add this call to the callAccountMap in ManagerImpl
+ Manager::instance().getAccountLink(account_id)->addCall(call);
+ _debug("UserAgent: Notify UI success!\n");
+ } else {
+ // Fail to notify UI
+ delete call;
+ call = NULL;
+ _debug("UserAgent: Fail to notify UI!\n");
+ return false;
+ }
+ } else {
+ // Fail to collect call information
+ delete call;
+ call = NULL;
+ _debug("UserAgent: Call SIPCallInvite failed!\n");
+ return false;
+ }
+
+ /* Create the local dialog (UAS) */
+ status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dialog);
+ if (status != PJ_SUCCESS) {
+ pjsip_endpt_respond_stateless( _endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, &reason, NULL,
+ NULL);
+ return true;
+ }
+
+ // Specify media capability during invite session creation
+ pjsip_inv_session *inv;
+ status = pjsip_inv_create_uas(dialog, rdata, call->getLocalSDPSession(), 0, &inv);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ // Associate the call in the invite session
+ inv->mod_data[_mod_ua.id] = call;
+
+ // Send a 180/Ringing response
+ status = pjsip_inv_initial_answer(inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjsip_inv_send_msg(inv, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ // Associate invite session to the current call
+ call->setInvSession(inv);
+
+ // Update the connection state
+ call->setConnectionState(Call::Ringing);
+
+ /* Done */
+ return true;
+
+ }
+
+ pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata UNUSED) {
+ _debug("mod_on_rx_response\n");
+ return PJ_SUCCESS;
+ }
+
+void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
{
- //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
- if (!call) {
- _debug("! SIP Failure: unknown call\n");
- return;
- }
- //call->setDid(event->did);
-
- if (call->getConnectionState() != Call::Connected) {
- //call->SIPCallAnswered(event);
- call->SIPCallAnsweredWithoutHold(rdata);
-
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ SIPCall *existing_call;
+ const pj_str_t str_refer_to = { (char*)"Refer-To", 8};
+ const pj_str_t str_refer_sub = { (char*)"Refer-Sub", 9 };
+ const pj_str_t str_ref_by = { (char*)"Referred-By", 11 };
+ pjsip_generic_string_hdr *refer_to;
+ pjsip_generic_string_hdr *refer_sub;
+ pjsip_hdr *ref_by_hdr;
+ pj_bool_t no_refer_sub = PJ_FALSE;
+ char *uri;
+ std::string tmp;
+ pjsip_status_code code;
+ pjsip_evsub *sub;
+
+ existing_call = (SIPCall *) inv->mod_data[_mod_ua.id];
+
+ /* Find the Refer-To header */
+ refer_to = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL);
+
+ if (refer_to == NULL) {
+ /* Invalid Request.
+ * No Refer-To header!
+ */
+ _debug("UserAgent: Received REFER without Refer-To header!\n");
+ pjsip_dlg_respond( inv->dlg, rdata, 400, NULL, NULL, NULL);
+ return;
+ }
- Manager::instance().peerAnsweredCall(call->getCallId());
- if (Manager::instance().isCurrentCall(call->getCallId())) {
- _debug("* SIP Info: Starting AudioRTP when answering\n");
- if ( _audiortp->createNewSession(call) < 0) {
- _debug("RTP Failure: unable to create new session\n");
- } else {
- call->setAudioStart(true);
- }
+ /* Find optional Refer-Sub header */
+ refer_sub = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_sub, NULL);
+
+ if (refer_sub) {
+ if (!pj_strnicmp2(&refer_sub->hvalue, "true", 4)==0)
+ no_refer_sub = PJ_TRUE;
+ }
+
+ /* Find optional Referred-By header (to be copied onto outgoing INVITE
+ * request.
+ */
+ ref_by_hdr = (pjsip_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by,
+ NULL);
+
+ /* Notify callback */
+ code = PJSIP_SC_ACCEPTED;
+
+ _debug("UserAgent: Call to %.*s is being transfered to %.*s\n",
+ (int)inv->dlg->remote.info_str.slen,
+ inv->dlg->remote.info_str.ptr,
+ (int)refer_to->hvalue.slen,
+ refer_to->hvalue.ptr);
+
+ if (no_refer_sub) {
+ /*
+ * Always answer with 2xx.
+ */
+ pjsip_tx_data *tdata;
+ const pj_str_t str_false = { (char*)"false", 5};
+ pjsip_hdr *hdr;
+
+ status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL,
+ &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
+ return;
+ }
+
+ /* Add Refer-Sub header */
+ hdr = (pjsip_hdr*)
+ pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
+ &str_false);
+ pjsip_msg_add_hdr(tdata->msg, hdr);
+
+
+ /* Send answer */
+ status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata),
+ tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
+ return;
+ }
+
+ /* Don't have subscription */
+ sub = NULL;
+
+ } else {
+ struct pjsip_evsub_user xfer_cb;
+ pjsip_hdr hdr_list;
+
+ /* Init callback */
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_svr_cb;
+
+ /* Init addiTHIS_FILE, THIS_FILE, tional header list to be sent with REFER response */
+ pj_list_init(&hdr_list);
+
+ /* Create transferee event subscription */
+ status = pjsip_xfer_create_uas( inv->dlg, &xfer_cb, rdata, &sub);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create xfer uas -- %d\n", status);
+ pjsip_dlg_respond( inv->dlg, rdata, 500, NULL, NULL, NULL);
+ return;
+ }
+
+ /* If there's Refer-Sub header and the value is "true", send back
+ * Refer-Sub in the response with value "true" too.
+ */
+ if (refer_sub) {
+ const pj_str_t str_true = { (char*)"true", 4 };
+ pjsip_hdr *hdr;
+
+ hdr = (pjsip_hdr*)
+ pjsip_generic_string_hdr_create(inv->dlg->pool,
+ &str_refer_sub,
+ &str_true);
+ pj_list_push_back(&hdr_list, hdr);
+
+ }
+
+ /* Accept the REFER request, send 2xx. */
+ pjsip_xfer_accept(sub, rdata, code, &hdr_list);
+
+ /* Create initial NOTIFY request */
+ status = pjsip_xfer_notify( sub, PJSIP_EVSUB_STATE_ACTIVE,
+ 100, NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create NOTIFY to REFER -- %d", status);
+ return;
+ }
+
+ /* Send initial NOTIFY request */
+ status = pjsip_xfer_send_request( sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send NOTIFY to REFER -- %d\n", status);
+ return;
+ }
+ }
+
+ /* We're cheating here.
+ * We need to get a null terminated string from a pj_str_t.
+ * So grab the pointer from the hvalue and NULL terminate it, knowing
+ * that the NULL position will be occupied by a newline.
+ */
+ uri = refer_to->hvalue.ptr;
+ uri[refer_to->hvalue.slen] = '\0';
+
+ /* Now make the outgoing call. */
+ tmp = std::string(uri);
+
+ if(existing_call == NULL) {
+ _debug("UserAgent: Call doesn't exist!\n");
+ return;
}
- } else {
- _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
- //call->SIPCallAnswered(event);
- }
+
+ AccountID accId = Manager::instance().getAccountFromCall(existing_call->getCallId());
+ CallID newCallId = Manager::instance().getNewCallID();
+
+ if(!Manager::instance().outgoingCall(accId, newCallId, tmp)) {
+
+ /* Notify xferer about the error (if we have subscription) */
+ if (sub) {
+ status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED,
+ 500, NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to create NOTIFY to REFER -- %d\n", status);
+ return;
+ }
+ status = pjsip_xfer_send_request(sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Unable to send NOTIFY to REFER -- %d\n", status);
+ return;
+ }
+ }
+ return;
+ }
+
+ SIPCall* newCall;
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if(link) {
+ newCall = dynamic_cast<SIPCall *>(link->getCall(newCallId));
+ if(!newCall) {
+ _debug("UserAgent: can not find the call from sipvoiplink!\n");
+ return;
+ }
+ }
+
+ if (sub) {
+ /* Put the server subscription in inv_data.
+ * Subsequent state changed in pjsua_inv_on_state_changed() will be
+ * reported back to the server subscription.
+ */
+ newCall->setXferSub(sub);
+
+ /* Put the invite_data in the subscription. */
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id,
+ newCall);
+ }
}
-SIPCall*
-SIPVoIPLink::getSIPCall(const CallID& id)
-{
- Call* call = getCall(id);
- if (call) {
- return dynamic_cast<SIPCall*>(call);
- }
- return NULL;
+
+
+void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event){
+
+ PJ_UNUSED_ARG(event);
+
+ _debug("UserAgent: Transfer callback is involved!\n");
+ /*
+ * When subscription is accepted (got 200/OK to REFER), check if
+ * subscription suppressed.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
+
+ pjsip_rx_data *rdata;
+ pjsip_generic_string_hdr *refer_sub;
+ const pj_str_t REFER_SUB = {(char*)"Refer-Sub", 9 };
+
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
+ _mod_ua.id));
+
+ /* Must be receipt of response message */
+ pj_assert(event->type == PJSIP_EVENT_TSX_STATE &&
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Find Refer-Sub header */
+ refer_sub = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
+ &REFER_SUB, NULL);
+
+ /* Check if subscription is suppressed */
+ if (refer_sub && pj_stricmp2(&refer_sub->hvalue, "false")==0) {
+ /* Since no subscription is desired, assume that call has been
+ * transfered successfully.
+ */
+ if (link) {
+ // It's the time to stop the RTP
+ link->transferStep2();
+ }
+
+ /* Yes, subscription is suppressed.
+ * Terminate our subscription now.
+ */
+ _debug("UserAgent: Xfer subscription suppressed, terminating event subcription...\n");
+ pjsip_evsub_terminate(sub, PJ_TRUE);
+
+ } else {
+ /* Notify application about call transfer progress.
+ * Initially notify with 100/Accepted status.
+ */
+ _debug("UserAgent: Xfer subscription 100/Accepted received...\n");
+ }
+ }
+ /*
+ * On incoming NOTIFY, notify application about call transfer progress.
+ */
+ else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
+ pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
+ {
+ pjsip_msg *msg;
+ pjsip_msg_body *body;
+ pjsip_status_line status_line;
+ pj_bool_t is_last;
+ pj_bool_t cont;
+ pj_status_t status;
+
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
+ _mod_ua.id));
+
+ /* When subscription is terminated, clear the xfer_sub member of
+ * the inv_data.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ _debug("UserAgent: Xfer client subscription terminated\n");
+
+ }
+
+ if (!link || !event) {
+ /* Application is not interested with call progress status */
+ _debug("UserAgent: Either link or event is empty!\n");
+ return;
+ }
+
+ // Get current call
+ SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(Manager::instance().getCurrentCallId()));
+ if(!call) {
+ _debug("UserAgent: Call doesn't exit!\n");
+ return;
+ }
+
+ /* This better be a NOTIFY request */
+ if (event->type == PJSIP_EVENT_TSX_STATE &&
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+ {
+ pjsip_rx_data *rdata;
+
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Check if there's body */
+ msg = rdata->msg_info.msg;
+ body = msg->body;
+ if (!body) {
+ _debug("UserAgent: Warning! Received NOTIFY without message body\n");
+ return;
+ }
+
+ /* Check for appropriate content */
+ if (pj_stricmp2(&body->content_type.type, "message") != 0 ||
+ pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
+ {
+ _debug("UserAgent: Warning! Received NOTIFY with non message/sipfrag content\n");
+ return;
+ }
+
+ /* Try to parse the content */
+ status = pjsip_parse_status_line((char*)body->data, body->len,
+ &status_line);
+ if (status != PJ_SUCCESS) {
+ _debug("UserAgent: Warning! Received NOTIFY with invalid message/sipfrag content\n");
+ return;
+ }
+
+ } else {
+ _debug("UserAgent: Set code to 500!\n");
+ status_line.code = 500;
+ status_line.reason = *pjsip_get_status_text(500);
+ }
+
+ /* Notify application */
+ is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
+ cont = !is_last;
+
+ if(status_line.code/100 == 2) {
+ _debug("UserAgent: Try to stop rtp!\n");
+ pjsip_tx_data *tdata;
+
+ status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_GONE, NULL, &tdata);
+ if(status != PJ_SUCCESS) {
+ _debug("UserAgent: Fail to create end session msg!\n");
+ } else {
+ status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+ if(status != PJ_SUCCESS)
+ _debug("UserAgent: Fail to send end session msg!\n");
+ }
+
+ link->transferStep2();
+ cont = PJ_FALSE;
+ }
+
+ if (!cont) {
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ }
+ }
+
}
-///////////////////////////////////////////////////////////////////////////////
-// Private functions
-///////////////////////////////////////////////////////////////////////////////
-pj_str_t SIPVoIPLink::string2PJStr(const std::string &value)
+void xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event)
{
- char tmp[256];
-
- strcpy(tmp, value.data());
- return pj_str(tmp);
+ PJ_UNUSED_ARG(event);
+
+ /*
+ * When subscription is terminated, clear the xfer_sub member of
+ * the inv_data.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ SIPCall *call;
+
+ call = (SIPCall*) pjsip_evsub_get_mod_data(sub, _mod_ua.id);
+ if (!call)
+ return;
+
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ call->setXferSub(NULL);
+
+ _debug("UserAgent: Xfer server subscription terminated\n");
+ }
}
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 865703e2151eb4345c40e178b5fa2187aa06230a..c00479a5b2983ebd8699f94061d87a101e527109 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -23,12 +23,24 @@
#define SIPVOIPLINK_H
#include "voiplink.h"
-#include "useragent.h"
+
+#include <pjsip.h>
+#include <pjlib-util.h>
+#include <pjlib.h>
+#include <pjnath/stun_config.h>
+#include <pjsip_simple.h>
+#include <pjsip_ua.h>
+#include <pjmedia/sdp.h>
+#include <pjmedia/sdp_neg.h>
class EventThread;
class SIPCall;
class AudioRtp;
+#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
+#define RANDOM_SIP_PORT rand() % 64000 + 1024
+#define PJ_LOG_LEVEL 5
+
/**
* @file sipvoiplink.h
* @brief Specific VoIPLink for SIP (SIP core for incoming and outgoing events)
@@ -44,6 +56,8 @@ class SIPVoIPLink : public VoIPLink
*/
SIPVoIPLink(const AccountID& accountID);
+ static SIPVoIPLink* instance( const AccountID& id );
+
/**
* Destructor
*/
@@ -61,10 +75,7 @@ class SIPVoIPLink : public VoIPLink
*/
bool init(void);
- /**
- * Delete link-related stuuf like calls
- */
- void terminate(void);
+ void terminate( void );
/**
* Event listener. Each event send by the call manager is received and handled from here
@@ -158,8 +169,10 @@ class SIPVoIPLink : public VoIPLink
* If set to true, we check for a firewall
* @param use true if we use STUN
*/
- void setUseStun(bool use) { _useStun = use; }
+ void setUseStun(bool use);
+ bool useStun( void );
+
/**
* The name of the STUN server
* @param server Server FQDN/IP
@@ -235,15 +248,7 @@ class SIPVoIPLink : public VoIPLink
*/
std::string getSipTo(const std::string& to_url);
- /**
- * Set audio (SDP) configuration for a call
- * localport, localip, localexternalport
- * @param call a SIPCall valid pointer
- * @return bool True
- */
- bool setCallAudioLocal(SIPCall* call);
-
- /**
+ /**
* Tell the user that the call was answered
* @param
*/
@@ -268,6 +273,10 @@ class SIPVoIPLink : public VoIPLink
*/
void SIPCallReleased(SIPCall *call);
+ bool isInitDone() { return _initDone; }
+
+ void setInitState( bool state ){ _initDone = state; }
+
/**
* SIPCall accessor
* @param id The call identifier
@@ -275,31 +284,66 @@ class SIPVoIPLink : public VoIPLink
*/
SIPCall* getSIPCall(const CallID& id);
- /** Tell if the initialisation was done */
- bool _initDone;
-
- /** when we init the listener, how many times we try to bind a port? */
+ /** when we init the listener, how many times we try to bind a port? */
int _nbTryListenAddr;
- /** Do we use stun? */
- bool _useStun;
+ /** Starting sound */
+ AudioRtp* _audiortp;
+
+ pj_str_t string2PJStr(const std::string &value);
- /** What is the stun server? */
- std::string _stunServer;
+private:
+ static SIPVoIPLink* _instance;
+
+ int getModId();
+ /**
+ * Initialize the PJSIP library
+ * Must be called before any other calls to the SIP layer
+ *
+ * @return bool True on success
+ */
+ bool pjsip_init();
+
+ /**
+ * Delete link-related stuuf like calls
+ */
+ bool pjsip_shutdown(void);
+
+ pj_status_t stunServerResolve();
+ /** Create SIP UDP Listener */
+ int createUDPServer();
+
+ bool loadSIPLocalIP();
+
+ std::string getLocalIP() {return _localExternAddress;}
+
+ pjsip_regc *_regc;
+
+ /** For registration use only */
+ int _regPort;
+
+ /** Tell if the initialisation has been done */
+ bool _initDone;
+
+ /* Flag to check if the STUN server is valid or not */
+ bool validStunServer;
+
+ /** The current STUN server address */
+ std::string _stunServer;
+
/** Local Extern Address is the IP address seen by peers for SIP listener */
std::string _localExternAddress;
+ std::string _localIPAddress;
/** Local Extern Port is the port seen by peers for SIP listener */
- unsigned int _localExternPort;
+ unsigned int _localExternPort;
+ unsigned int _localPort;
- /** Starting sound */
- AudioRtp* _audiortp;
-
- pj_str_t string2PJStr(const std::string &value);
+ /** Threading object */
+ EventThread* _evThread;
+ ost::Mutex _mutexSIP;
-private:
- pjsip_regc *_regc;
bool _bRegister;
};
diff --git a/src/useragent.cpp b/src/useragent.cpp
index 6e0e104cec4ada90606d980fbf15212e2870e450..59a370dfab70abe8b6f6f911cdbad76fd6da3948 100644
--- a/src/useragent.cpp
+++ b/src/useragent.cpp
@@ -32,230 +32,13 @@
UserAgent *UserAgent::_current;
UserAgent::UserAgent():_endpt(NULL) ,_sock(NULL), _cp(), _pool(NULL), _mutex(NULL), _mod(), _useStun(false), _stunHost(),
- _stunServer(""), _localExternAddress(""), _localIPAddress("127.0.0.1"), _localExternPort(0), _localPort(0), _regPort(DEFAULT_SIP_PORT), _thread(NULL) {
- //_useStun = false;
- //_localIPAddress = "127.0.0.1";
- UserAgent::_current = this;
-}
-
-UserAgent::~UserAgent() {
- _debug("UserAgent: In dtor!\n");
- sipDestory();
-}
-
-pj_status_t UserAgent::sipCreate() {
-
- pj_status_t status;
-
- // Init PJLIB: must be called before any call to the pjsip library
- status = pj_init();
- // Use pjsip macros for sanity check
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init PJLIB-UTIL library
- status = pjlib_util_init();
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Set the pjsip log level
- pj_log_set_level( PJ_LOG_LEVEL );
-
- // Init PJNATH
- status = pjnath_init();
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Create a pool factory to allocate memory
- pj_caching_pool_init(&_cp, &pj_pool_factory_default_policy, 0);
-
- // Create memory pool for application.
- _pool = pj_pool_create(&_cp.factory, "sflphone", 4000, 4000, NULL);
-
- if (!_pool) {
- _debug("UserAgent: Could not initialize memory pool\n");
- return PJ_ENOMEM;
- }
-
- // Create a recursive mutex. Simple wrapper for pj_mutex_create
- status = pj_mutex_create_recursive(_pool, "sflphone", &_mutex);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Create the SIP endpoint
- status = pjsip_endpt_create(&_cp.factory, pj_gethostname()->ptr, &_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- return PJ_SUCCESS;
-}
-
-pj_status_t UserAgent::sipInit() {
-
- pj_status_t status;
- int errPjsip = 0;
- int port;
-
- validStunServer = true;
- /* Init SIP UA: */
-
- //FIXME! DNS initialize here! */
-
- /* Start resolving STUN server */
- // if we useStun and we failed to receive something on port 5060, we try a random port
- // If use STUN server, firewall address setup
- if (!loadSIPLocalIP()) {
- _debug("UserAgent: Unable to determine network capabilities\n");
- return false;
- }
- errPjsip = 0;
- port = _regPort;
-
- //_debug("stun host is %s\n", _stunHost.ptr);
- if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
- port = RANDOM_SIP_PORT;
- if (!Manager::instance().behindNat(_stunServer, port)) {
- _debug("UserAgent: Unable to check NAT setting\n");
- validStunServer = false;
- return false; // hoho we can't use the random sip port too...
- }
- }
-
- _localPort = port;
- if (_useStun) {
- // set by last behindNat() call (ish)...
- stunServerResolve();
- _localExternAddress = Manager::instance().getFirewallAddress();
- _localExternPort = Manager::instance().getFirewallPort();
- errPjsip = createUDPServer();
- if (errPjsip != 0) {
- _debug("UserAgent: Could not initialize SIP listener on port %d\n", port);
- return errPjsip;
- }
- } else {
- _localExternAddress = _localIPAddress;
- _localExternPort = _localPort;
- errPjsip = createUDPServer();
- if (errPjsip != 0) {
- _debug("UserAgent: Could not initialize SIP listener on port %d\n", _localExternPort);
- _localExternPort = _localPort = RANDOM_SIP_PORT;
- _debug("UserAgent: Try to initialize SIP listener on port %d\n", _localExternPort);
- errPjsip = createUDPServer();
- if (errPjsip != 0) {
- _debug("UserAgent: Fail to initialize SIP listener on port %d\n", _localExternPort);
- return errPjsip;
- }
- }
- }
-
- _debug("UserAgent: SIP Init -- listening on port %d\n", _localExternPort);
-
- // Initialize transaction layer
- status = pjsip_tsx_layer_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize UA layer module
- status = pjsip_ua_init_module(_endpt, NULL);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize Replaces support. See the Replaces specification in RFC 3891
- status = pjsip_replaces_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize 100rel support
- status = pjsip_100rel_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize and register sflphone module
- {
- const pjsip_module mod_initializer ={
- NULL, NULL, // prev, next.
- { (char*)"mod-sflphone", 9}, // Name.
- -1, // Id
- PJSIP_MOD_PRIORITY_APPLICATION, // Priority
- NULL, // load()
- NULL, // start()
- NULL, // stop()
- NULL, // unload()
- &mod_on_rx_request, // on_rx_request()
- &mod_on_rx_response, // on_rx_response()
- NULL, // on_tx_request.
- NULL, // on_tx_response()
- NULL, // on_tsx_state()
- };
-
- _mod = mod_initializer;
-
- status = pjsip_endpt_register_module(_endpt, &_mod);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- }
-
- // Init the event subscription module.
- // It extends PJSIP by supporting SUBSCRIBE and NOTIFY methods
- status = pjsip_evsub_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init presence module.
- // TODO We probably do not need that extension
- status = pjsip_pres_init_module(_endpt, pjsip_evsub_instance());
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init PUBLISH module
- // Provide an implementation of SIP Extension for Event State Publication (RFC 3903)
- // TODO Check if it is necessary
- status = pjsip_publishc_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Init xfer/REFER module
- status = pjsip_xfer_init_module(_endpt);
- PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
-
- // Initialize invite session module
- // These callbacks will be called on incoming requests, media session state, etc.
- {
- pjsip_inv_callback inv_cb;
-
- // Init the callback for INVITE session:
- pj_bzero(&inv_cb, sizeof (inv_cb));
-
- inv_cb.on_state_changed = &call_on_state_changed;
- inv_cb.on_new_session = &call_on_forked;
- inv_cb.on_media_update = &call_on_media_update;
- inv_cb.on_tsx_state_changed = &call_on_tsx_changed;
-
- _debug("UserAgent: VOIP callbacks initialized\n");
-
- // Initialize session invite module
- status = pjsip_inv_usage_init(_endpt, &inv_cb);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ _stunServer(""), _localExternAddress(""), _localIPAddress("127.0.0.1"), _localExternPort(0), _localPort(0), _regPort(DEFAULT_SIP_PORT), _thread(NULL) {
+ //_useStun = false;
+ //_localIPAddress = "127.0.0.1";
+ UserAgent::_current = this;
}
- // Add endpoint capabilities (INFO, OPTIONS, etc) for this UA
- {
- pj_str_t allowed[] = {
- {(char*)"INFO", 4},
- {(char*)"REGISTER", 8}
- }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}, {"OPTIONS", 7},
- pj_str_t accepted = {(char*)"application/sdp", 15};
-
- // Register supported methods
- pjsip_endpt_add_capability(_endpt, &_mod, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
-
- // Register "application/sdp" in ACCEPT header
- pjsip_endpt_add_capability(_endpt, &_mod, PJSIP_H_ACCEPT, NULL, 1, &accepted);
- }
-
- _debug("UserAgent: pjsip version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
-
- Manager::instance().setSipThreadStatus(false);
-
- // Create the secondary thread to poll sip events
- status = pj_thread_create(_pool, "sflphone", &start_thread, NULL, PJ_THREAD_DEFAULT_STACK_SIZE, 0,
- &_thread);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
- /* Done! */
- return PJ_SUCCESS;
-
-}
-
void UserAgent::sipDestory() {
/* Signal threads to quit: */
Manager::instance().setSipThreadStatus(true);
@@ -319,7 +102,7 @@ void UserAgent::busy_sleep(unsigned msec)
tv.sec = 0;
tv.msec = 10;
pj_time_val_normalize(&tv);
-
+
do {
pjsip_endpt_handle_events(_endpt, &tv);
pj_gettimeofday(&now);
@@ -327,128 +110,7 @@ void UserAgent::busy_sleep(unsigned msec)
#endif
}
-bool UserAgent::addAccount(AccountID id, pjsip_regc **regc2, const std::string& server, const std::string& user, const std::string& passwd, const int& timeout UNUSED) {
-
- pj_status_t status;
- AccountID *currentId = new AccountID(id);
- char contactTmp[256];
- pjsip_regc *regc;
- pj_str_t svr;
- pj_str_t aor;
- pj_str_t contact;
- pjsip_tx_data *tdata;
-
- //pj_mutex_lock(_mutex);
- std::string tmp;
-
-
- SIPAccount *account;
-
- if (!validStunServer) {
-
- SIPVoIPLink *voipLink;
- voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(id));
- Manager::instance().getAccountLink(id)->setRegistrationState(VoIPLink::ErrorExistStun);
- voipLink->setRegister(false);
- return false;
- }
-
- status = pjsip_regc_create(_endpt, (void *) currentId, ®c_cb, ®c);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create regc.\n");
- return false;
- }
-
- tmp = "sip:" + server;
- pj_strdup2(_pool, &svr, tmp.data());
-
- tmp = "<sip:" + user + "@" + server + ">";
- pj_strdup2(_pool, &aor, tmp.data());
-
-
- sprintf(contactTmp, "<sip:%s@%s:%d>", user.data(), _localExternAddress.data(), _localExternPort);
- pj_strdup2(_pool, &contact, contactTmp);
-
- //_debug("UserAgent: Get in %s %d %s\n", svr.ptr, svr.slen, aor.ptr);
- status = pjsip_regc_init(regc, &svr, &aor, &aor, 1, &contact, 600); //timeout);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to initialize regc. %d\n", status); //, regc->str_srv_url.ptr);
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
-
- account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount(id));
- pjsip_cred_info *cred = account->getCredInfo();
- if(!cred)
- cred = new pjsip_cred_info();
-
- pj_bzero(cred, sizeof (pjsip_cred_info));
- pj_strdup2(_pool, &cred->username, user.data());
- cred->data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
- pj_strdup2(_pool, &cred->data, passwd.data());
- pj_strdup2(_pool, &cred->realm, "*");
- pj_strdup2(_pool, &cred->scheme, "digest");
- pjsip_regc_set_credentials(regc, 1, cred);
-
- account->setCredInfo(cred);
-
- status = pjsip_regc_register(regc, PJ_TRUE, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to register regc.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- status = pjsip_regc_send(regc, tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send regc request.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- account->setUserName(user);
- account->setServer(server);
- account->setContact(contactTmp);
-
- // associate regc with account
- *regc2 = regc;
-
- //pj_mutex_unlock(_mutex);
-
- return true;
-}
-
-bool UserAgent::removeAccount(pjsip_regc *regc)
-{
- pj_status_t status = 0;
- pjsip_tx_data *tdata = NULL;
-
- //pj_mutex_lock(_mutex);
- if(regc) {
- status = pjsip_regc_unregister(regc, &tdata);
- if(status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to unregister regc.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- status = pjsip_regc_send( regc, tdata );
- if(status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send regc request.\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
- } else {
- _debug("UserAgent: regc is null!\n");
- //pj_mutex_unlock(_mutex);
- return false;
- }
-
- //pj_mutex_unlock(_mutex);
- return true;
-}
pj_str_t UserAgent::buildContact(char *userName) {
//pj_str_t contact;
@@ -461,88 +123,6 @@ pj_str_t UserAgent::buildContact(char *userName) {
return pj_str(tmp);
}
-pj_status_t UserAgent::stunServerResolve() {
- pj_str_t stun_adr;
- pj_hostent he;
- pj_stun_config stunCfg;
- pj_status_t stun_status;
- pj_sockaddr stun_srv;
-
- // Initialize STUN configuration
- pj_stun_config_init(&stunCfg, &_cp.factory, 0, pjsip_endpt_get_ioqueue(_endpt), pjsip_endpt_get_timer_heap(_endpt));
-
- stun_status = PJ_EPENDING;
-
- // Init STUN socket
- size_t pos = _stunServer.find(':');
- if(pos == std::string::npos) {
- pj_strdup2(_pool, &stun_adr, _stunServer.data());
- stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) 3478);
- } else {
- std::string serverName = _stunServer.substr(0, pos);
- std::string serverPort = _stunServer.substr(pos + 1);
- int nPort = atoi(serverPort.data());
- pj_strdup2(_pool, &stun_adr, serverName.data());
- stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) nPort);
- }
-
- if (stun_status != PJ_SUCCESS) {
- _debug("UserAgent: Unresolved stun server!\n");
- stun_status = pj_gethostbyname(&stun_adr, &he);
-
- if (stun_status == PJ_SUCCESS) {
- pj_sockaddr_in_init(&stun_srv.ipv4, NULL, 0);
- stun_srv.ipv4.sin_addr = *(pj_in_addr*) he.h_addr;
- stun_srv.ipv4.sin_port = pj_htons((pj_uint16_t) 3478);
- }
- }
-
- return stun_status;
-}
-
-int UserAgent::createUDPServer() {
- pj_status_t status;
- //pj_str_t ipAddr;
- pj_sockaddr_in bound_addr;
- pjsip_host_port a_name;
- char tmpIP[32];
-
- // Init bound address to ANY
- pj_memset(&bound_addr, 0, sizeof (bound_addr));
- bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
-
- // Create UDP server socket
- status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &_sock);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) UDP socket() error\n", status);
- return status;
- }
-
- status = pj_sock_bind_in(_sock, pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t) _localPort);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) UDP bind() error\n", status);
- pj_sock_close(_sock);
- return status;
- }
-
- _debug("UserAgent: Use IP: %s\n", _localExternAddress.data());
-
- // Create UDP-Server (default port: 5060)
- strcpy(tmpIP, _localExternAddress.data());
- pj_strdup2(_pool, &a_name.host, tmpIP);
- a_name.port = (pj_uint16_t) _localExternPort;
-
- status = pjsip_udp_transport_attach(_endpt, _sock, &a_name, 1, NULL);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) Unable to start UDP transport!\n", status);
- return -1;
- } else {
- _debug("UserAgent: UDP server listening on port %d\n", _localExternPort);
- }
-
- return 0;
-}
-
void UserAgent::setStunServer(const char *server) {
if(server != NULL) {
_useStun = true;
@@ -553,88 +133,6 @@ void UserAgent::setStunServer(const char *server) {
}
}
-void UserAgent::regc_cb(struct pjsip_regc_cbparam *param) {
-
- AccountID *id = static_cast<AccountID *> (param->token);
- SIPVoIPLink *voipLink;
-
- _debug("UserAgent: Account ID is %s, Register result: %d, Status: %d\n", id->data(), param->status, param->code);
- voipLink = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(*id));
- if(!voipLink)
- return;
-
- if (param->status == PJ_SUCCESS) {
- if (param->code < 0 || param->code >= 300) {
- /* Sometimes, the status is OK, but we still failed.
- * So checking the code for real result
- */
- _debug("UserAgent: The error is: %d\n", param->code);
- switch(param->code) {
- case 408:
- case 606:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorConfStun);
- break;
- case 503:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorHost);
- break;
- case 401:
- case 403:
- case 404:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorAuth);
- break;
- default:
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Error);
- break;
- }
- voipLink->setRegister(false);
- } else {
- // Registration/Unregistration is success
-
- if(voipLink->isRegister())
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Registered);
- else {
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Unregistered);
- voipLink->setRegister(false);
- }
- }
- } else {
- Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::ErrorAuth);
- voipLink->setRegister(false);
- }
-}
-
-bool
-UserAgent::loadSIPLocalIP() {
- bool returnValue = true;
- if (_localIPAddress == "127.0.0.1") {
- pj_sockaddr ip_addr;
- if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
- // Update the registration state if no network capabilities found
- _debug("UserAgent: Get host ip failed!\n");
- returnValue = false;
- } else {
- _localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
- _debug("UserAgent: Checking network, setting local IP address to: %s\n", _localIPAddress.data());
- }
- }
- return returnValue;
-}
-
-/* Thread entry point function. */
-int UserAgent::start_thread(void *arg) {
-
- PJ_UNUSED_ARG(arg);
-
- // FIXME! maybe we should add a flag for exiting!
- // TODO Add the flag. We have to stop the thread when destroying the instance
- while (!Manager::instance().getSipThreadStatus()) {
- pj_time_val timeout = {0, 10};
- pjsip_endpt_handle_events(getInstance()->getEndPoint(), &timeout);
- }
-
- return 0;
-}
-
void UserAgent::set_voicemail_info( AccountID account, pjsip_msg_body *body ){
int voicemail, pos_begin, pos_end;
@@ -657,17 +155,17 @@ void UserAgent::set_voicemail_info( AccountID account, pjsip_msg_body *body ){
// So our voicemail number between the both index
try {
- voicemail_str = msg_body.substr(pos_begin + voice_str.length(), pos_end - ( pos_begin + voice_str.length()));
- std::cout << "voicemail number : " << voicemail_str << std::endl;
- voicemail = atoi( voicemail_str.c_str() );
+ voicemail_str = msg_body.substr(pos_begin + voice_str.length(), pos_end - ( pos_begin + voice_str.length()));
+ std::cout << "voicemail number : " << voicemail_str << std::endl;
+ voicemail = atoi( voicemail_str.c_str() );
}
catch( std::out_of_range& e ){
- std::cerr << e.what() << std::endl;
+ std::cerr << e.what() << std::endl;
}
// We need now to notify the manager
if( voicemail != 0 )
- Manager::instance().startVoiceMessageNotification(account, voicemail);
+ Manager::instance().startVoiceMessageNotification(account, voicemail);
}
@@ -700,8 +198,8 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
// Get the account id of callee from username and server
account_id = Manager::instance().getAccountIdFromNameAndServer(userName, server);
if(account_id == AccountNULL) {
- _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
- return PJ_FALSE;
+ _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
+ return PJ_FALSE;
}
_debug("UserAgent: The receiver is : %s@%s\n", userName.data(), server.data());
_debug("UserAgent: The callee account id is %s\n", account_id.c_str());
@@ -709,25 +207,25 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
/* Now, it is the time to find the information of the caller */
uri = rdata->msg_info.from->uri;
sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
-
+
std::string caller = std::string(sip_uri->user.ptr, sip_uri->user.slen);
std::string callerServer = std::string(sip_uri->host.ptr, sip_uri->host.slen);
std::string peerNumber = caller + "@" + callerServer;
-
-
+
+
// Get the server voicemail notification
// Catch the NOTIFY message
if( rdata->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD )
{
- method_name = "NOTIFY";
- // Retrieve all the message. Should contains only the method name but ...
- request = rdata->msg_info.msg->line.req.method.name.ptr;
- // Check if the message is a notification
- if( request.find( method_name ) != (size_t)-1 ) {
- set_voicemail_info( account_id, rdata->msg_info.msg->body );
- }
+ method_name = "NOTIFY";
+ // Retrieve all the message. Should contains only the method name but ...
+ request = rdata->msg_info.msg->line.req.method.name.ptr;
+ // Check if the message is a notification
+ if( request.find( method_name ) != (size_t)-1 ) {
+ set_voicemail_info( account_id, rdata->msg_info.msg->body );
+ }
pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, PJSIP_SC_OK, NULL, NULL, NULL);
- return PJ_SUCCESS;
+ return PJ_SUCCESS;
}
// Respond statelessly any non-INVITE requests with 500
@@ -739,7 +237,7 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
return PJ_TRUE;
}
}
-
+
// Verify that we can handle the request
status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, getInstance()->getEndPoint(), NULL);
if (status != PJ_SUCCESS) {
@@ -765,7 +263,7 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
call->setConnectionState(Call::Progressing);
call->setIp(getInstance()->getLocalIP());
call->setPeerNumber(peerNumber);
-
+
/* Call the SIPCallInvite function to generate the local sdp,
* remote sdp and negociator.
* This function is also used to set the parameters of audio RTP, including:
@@ -774,7 +272,7 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
* possilbe audio codec will be used in this call
*/
if (call->SIPCallInvite(rdata, getInstance()->getAppPool())) {
-
+
// Notify UI there is an incoming call
if (Manager::instance().incomingCall(call, account_id)) {
// Add this call to the callAccountMap in ManagerImpl
@@ -802,7 +300,7 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
NULL);
return PJ_TRUE;
}
-
+
// Specify media capability during invite session creation
pjsip_inv_session *inv;
status = pjsip_inv_create_uas(dialog, rdata, call->getLocalSDPSession(), 0, &inv);
@@ -810,7 +308,7 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
// Associate the call in the invite session
inv->mod_data[getInstance()->getModId()] = call;
-
+
// Send a 180/Ringing response
status = pjsip_inv_initial_answer(inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
@@ -827,107 +325,8 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
return PJ_SUCCESS;
}
-bool UserAgent::setCallAudioLocal(SIPCall* call) {
- // Firstly, we use the local IP and port number
- unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
- unsigned int callLocalExternAudioPort = callLocalAudioPort;
-
- if (_useStun) {
- // If use Stun server, modify them
- if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
- callLocalExternAudioPort = Manager::instance().getFirewallPort();
- }
- }
- _debug("UserAgent: Setting local audio port to: %d\n", callLocalAudioPort);
- _debug("UserAgent: Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
-
- // Set local audio port for SIPCall(id)
- call->setLocalIp(_localIPAddress);
- call->setLocalAudioPort(callLocalAudioPort);
- call->setLocalExternAudioPort(callLocalExternAudioPort);
-
- return true;
-}
-
-int UserAgent::answer(SIPCall *call) {
- pj_status_t status;
- pjsip_tx_data *tdata;
-
- // User answered the incoming call, tell peer this news
- if (call->startNegociation(_pool)) {
- // Create and send a 200(OK) response
- _debug("UserAgent: Negociation success!\n");
- status = pjsip_inv_answer(call->getInvSession(), PJSIP_SC_OK, NULL, NULL, &tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
-
- return 0;
- }
-
- return 1;
-}
-
-bool UserAgent::makeOutgoingCall(const std::string& strTo, SIPCall* call, const AccountID& id) {
- pj_status_t status;
- pjsip_dialog *dialog;
- pjsip_tx_data *tdata;
- pj_str_t from, to, contact;
-
- _debug("*******************AccountId is %s\n", id.data());
- // Get the basic information about the callee account
- SIPAccount* account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
-
- // Generate the from URI
- std::string strFrom = "sip:" + account->getUserName() + "@" + account->getServer();
-
- _debug("UserAgent: Make a new call from:%s to %s. Contact is %s\n",
- strFrom.data(), strTo.data(), account->getContact().data());
-
- // pjsip need the from and to information in pj_str_t format
- pj_strdup2(_pool, &from, strFrom.data());
- pj_strdup2(_pool, &to, strTo.data());
- pj_strdup2(_pool, &contact, account->getContact().data());
-
- // create the dialog (UAC)
- status = pjsip_dlg_create_uac(pjsip_ua_instance(), &from,
- &contact,
- &to,
- NULL,
- &dialog);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-
- setCallAudioLocal(call);
- call->setIp(getInstance()->getLocalIP());
-
- // Building the local SDP offer
- call->createInitialOffer(_pool);
-
- // Create the invite session for this call
- pjsip_inv_session *inv;
- status = pjsip_inv_create_uac(dialog, call->getLocalSDPSession(), 0, &inv);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-
- // Set auth information
- pjsip_auth_clt_set_credentials(&dialog->auth_sess, 1, account->getCredInfo());
- // Associate current call in the invite session
- inv->mod_data[_mod.id] = call;
- status = pjsip_inv_invite(inv, &tdata);
- PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
-
- // Associate current invite session in the call
- call->setInvSession(inv);
-
- status = pjsip_inv_send_msg(inv, tdata);
- //PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- if(status != PJ_SUCCESS) {
- return false;
- }
-
- return true;
-}
void UserAgent::call_on_forked(pjsip_inv_session *inv, pjsip_event *e) {
PJ_UNUSED_ARG(inv);
@@ -946,8 +345,8 @@ void UserAgent::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *t
tsx->state, tsx->role, tsx->status_code, (int)tsx->method.name.slen, tsx->method.name.ptr);
if(pj_strcmp2(&tsx->method.name, "INFO") == 0) {
- // Receive a INFO message, ingore it!
- return;
+ // Receive a INFO message, ingore it!
+ return;
}
//Retrieve the body message
@@ -971,7 +370,7 @@ void UserAgent::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *t
if (link)
link->SIPCallAnswered(call, rdata);
} else if (tsx->status_code / 100 == 5) {
- _debug("UserAgent: 5xx error message received\n");
+ _debug("UserAgent: 5xx error message received\n");
}
break;
case PJSIP_TSX_STATE_PROCEEDING:
@@ -990,7 +389,7 @@ void UserAgent::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *t
}
break;
case PJSIP_TSX_STATE_COMPLETED:
- if (tsx->status_code == 407 || tsx->status_code == 401) //FIXME
+ if (tsx->status_code == 407 || tsx->status_code == 401) //FIXME
break;
if (tsx->status_code / 100 == 6 || tsx->status_code / 100 == 4) {
// We get error message of outgoing call from server
@@ -1010,7 +409,7 @@ void UserAgent::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *t
default:
break;
} // end of switch
-
+
} else {
switch (tsx->state) {
case PJSIP_TSX_STATE_TRYING:
@@ -1060,11 +459,11 @@ void UserAgent::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *t
void UserAgent::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
PJ_UNUSED_ARG(inv);
-
+
SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[getInstance()->getModId()]);
if(!call)
return;
-
+
/* If this is an outgoing INVITE that was created because of
* REFER/transfer, send NOTIFY to transferer.
*/
@@ -1110,8 +509,8 @@ void UserAgent::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
pj_status_t status;
status = pjsip_xfer_notify( call->getXferSub(),
- ev_state, st_code,
- NULL, &tdata);
+ ev_state, st_code,
+ NULL, &tdata);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Unable to create NOTIFY -- %d\n", status);
} else {
@@ -1125,203 +524,6 @@ void UserAgent::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
}
-bool UserAgent::onhold(SIPCall *call) {
-
- pj_status_t status;
- pjsip_tx_data *tdata;
- pjmedia_sdp_attr *attr;
- pjmedia_sdp_session* local_sdp;
-
- local_sdp = call->getLocalSDPSession();
- if( local_sdp == NULL ){
- _debug("! SIP Failure: unable to find local_sdp\n");
- return false;
- }
-
- /* Create re-INVITE with new offer */
- // Remove all the attributes with the specified name
- pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendrecv");
- attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
- pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
-
- status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp, &tdata);
- if( status != PJ_SUCCESS )
- {
- _debug("On hold: creation of the Re-invite request failed\n");
- return false;
- }
- /* Send the request */
- status = pjsip_inv_send_msg( call->getInvSession(), tdata);
-
- return (status == PJ_SUCCESS);
-}
-
-bool UserAgent::offhold(SIPCall *call) {
-
- pj_status_t status;
- pjsip_tx_data *tdata;
- pjmedia_sdp_attr *attr;
- pjmedia_sdp_session* local_sdp;
-
- local_sdp = call->getLocalSDPSession();
- if( local_sdp == NULL ){
- _debug("! SIP Failure: unable to find local_sdp\n");
- return false;
- }
-
- /* Create re-INVITE with new offer */
- // Remove all the attributes with the specified name
- pjmedia_sdp_media_remove_all_attr(local_sdp->media[0], "sendonly");
- attr = pjmedia_sdp_attr_create(_pool, "sendrecv", NULL);
- pjmedia_sdp_media_add_attr(local_sdp->media[0], attr);
-
- status = pjsip_inv_reinvite( call->getInvSession(), NULL, local_sdp , &tdata);
- if( status != PJ_SUCCESS )
- {
- _debug("Off hold: creation of the Re-invite request failed\n");
- return false;
- }
-
- /* Send the request */
- status = pjsip_inv_send_msg( call->getInvSession(), tdata);
-
- return (status == PJ_SUCCESS);
-}
-
-bool UserAgent::hangup(SIPCall* call) {
- pj_status_t status;
- pjsip_tx_data *tdata = NULL;
-
- // User hangup current call. Notify peer
- status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
- if(status != PJ_SUCCESS)
- return false;
-
- _debug("UserAgent: Before send msg!\n");
-
- if(tdata == NULL)
- return true;
-
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- if(status != PJ_SUCCESS)
- return false;
-
- call->getInvSession()->mod_data[getInstance()->getModId()] = NULL;
- return true;
-}
-
-bool UserAgent::refuse(SIPCall* call)
-{
- pj_status_t status;
- pjsip_tx_data *tdata;
-
- // User refuse current call. Notify peer
- status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata); //603
- if(status != PJ_SUCCESS)
- return false;
-
- status = pjsip_inv_send_msg(call->getInvSession(), tdata);
- if(status != PJ_SUCCESS)
- return false;
-
- call->getInvSession()->mod_data[getInstance()->getModId()] = NULL;
- return true;
-}
-
-
-bool UserAgent::carryingDTMFdigits(SIPCall* call, char *msgBody)
-{
- pj_status_t status;
- pjsip_tx_data *tdata;
- pj_str_t methodName, content;
- pjsip_method method;
- pjsip_media_type ctype;
-
- pj_strdup2(_pool, &methodName, "INFO");
- pjsip_method_init_np(&method, &methodName);
-
- /* Create request message. */
- status = pjsip_dlg_create_request( call->getInvSession()->dlg, &method,
- -1, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create INFO request -- %d\n", status);
- return false;
- }
-
- /* Get MIME type */
- pj_strdup2(_pool, &ctype.type, "application");
- pj_strdup2(_pool, &ctype.subtype, "dtmf-relay");
-
- /* Create "application/dtmf-relay" message body. */
- pj_strdup2(_pool, &content, msgBody);
- tdata->msg->body = pjsip_msg_body_create( tdata->pool, &ctype.type,
- &ctype.subtype, &content);
- if (tdata->msg->body == NULL) {
- _debug("UserAgent: Unable to create msg body!\n");
- pjsip_tx_data_dec_ref(tdata);
- return false;
- }
-
- /* Send the request. */
- status = pjsip_dlg_send_request( call->getInvSession()->dlg, tdata,
- _mod.id, NULL);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send MESSAGE request -- %d\n", status);
- return false;
- }
-
- return true;
-
-}
-
-bool UserAgent::transfer(SIPCall *call, const std::string& to)
-{
- pjsip_evsub *sub;
- pjsip_tx_data *tdata;
- struct pjsip_evsub_user xfer_cb;
- pj_status_t status;
- pj_str_t dest;
-
- pj_strdup2(_pool, &dest, to.data());
-
- /* Create xfer client subscription. */
- pj_bzero(&xfer_cb, sizeof(xfer_cb));
- xfer_cb.on_evsub_state = &xfer_func_cb;
-
- status = pjsip_xfer_create_uac(call->getInvSession()->dlg, &xfer_cb, &sub);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create xfer -- %d\n", status);
- return false;
- }
-
- /* Associate this voiplink of call with the client subscription
- * We can not just associate call with the client subscription
- * because after this function, we can not find the cooresponding
- * voiplink from the call any more. But the voiplink is useful!
- */
- AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
- SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
- pjsip_evsub_set_mod_data(sub, _mod.id, link);
-
- /*
- * Create REFER request.
- */
- status = pjsip_xfer_initiate(sub, &dest, &tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to create REFER request -- %d\n", status);
- return false;
- }
-
- /* Send. */
- status = pjsip_xfer_send_request(sub, tdata);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: Unable to send REFER request -- %d\n", status);
- return false;
- }
-
- return true;
-}
-
void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
{
PJ_UNUSED_ARG(event);
@@ -1338,17 +540,17 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
const pj_str_t REFER_SUB = {(char*)"Refer-Sub", 9 };
SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
- getInstance()->getModId()));
+ getInstance()->getModId()));
/* Must be receipt of response message */
pj_assert(event->type == PJSIP_EVENT_TSX_STATE &&
- event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
rdata = event->body.tsx_state.src.rdata;
/* Find Refer-Sub header */
refer_sub = (pjsip_generic_string_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
- &REFER_SUB, NULL);
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
+ &REFER_SUB, NULL);
/* Check if subscription is suppressed */
if (refer_sub && pj_stricmp2(&refer_sub->hvalue, "false")==0) {
@@ -1377,7 +579,7 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
* On incoming NOTIFY, notify application about call transfer progress.
*/
else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
- pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
+ pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
{
pjsip_msg *msg;
pjsip_msg_body *body;
@@ -1387,7 +589,7 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
pj_status_t status;
SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
- getInstance()->getModId()));
+ getInstance()->getModId()));
/* When subscription is terminated, clear the xfer_sub member of
* the inv_data.
@@ -1410,10 +612,10 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
_debug("UserAgent: Call doesn't exit!\n");
return;
}
-
+
/* This better be a NOTIFY request */
if (event->type == PJSIP_EVENT_TSX_STATE &&
- event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
{
pjsip_rx_data *rdata;
@@ -1429,7 +631,7 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
/* Check for appropriate content */
if (pj_stricmp2(&body->content_type.type, "message") != 0 ||
- pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
+ pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
{
_debug("UserAgent: Warning! Received NOTIFY with non message/sipfrag content\n");
return;
@@ -1437,7 +639,7 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
/* Try to parse the content */
status = pjsip_parse_status_line((char*)body->data, body->len,
- &status_line);
+ &status_line);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Warning! Received NOTIFY with invalid message/sipfrag content\n");
return;
@@ -1452,11 +654,11 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
/* Notify application */
is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
cont = !is_last;
-
+
if(status_line.code/100 == 2) {
_debug("UserAgent: Try to stop rtp!\n");
pjsip_tx_data *tdata;
-
+
status = pjsip_inv_end_session(call->getInvSession(), PJSIP_SC_GONE, NULL, &tdata);
if(status != PJ_SUCCESS) {
_debug("UserAgent: Fail to create end session msg!\n");
@@ -1465,16 +667,16 @@ void UserAgent::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
if(status != PJ_SUCCESS)
_debug("UserAgent: Fail to send end session msg!\n");
}
-
+
link->transferStep2();
cont = PJ_FALSE;
}
-
+
if (!cont) {
pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
}
}
-
+
}
void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
@@ -1522,17 +724,17 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
* request.
*/
ref_by_hdr = (pjsip_hdr*)
- pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by,
- NULL);
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by,
+ NULL);
/* Notify callback */
code = PJSIP_SC_ACCEPTED;
_debug("UserAgent: Call to %.*s is being transfered to %.*s\n",
- (int)inv->dlg->remote.info_str.slen,
- inv->dlg->remote.info_str.ptr,
- (int)refer_to->hvalue.slen,
- refer_to->hvalue.ptr);
+ (int)inv->dlg->remote.info_str.slen,
+ inv->dlg->remote.info_str.ptr,
+ (int)refer_to->hvalue.slen,
+ refer_to->hvalue.ptr);
if (no_refer_sub) {
/*
@@ -1543,7 +745,7 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
pjsip_hdr *hdr;
status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL,
- &tdata);
+ &tdata);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
return;
@@ -1551,14 +753,14 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
/* Add Refer-Sub header */
hdr = (pjsip_hdr*)
- pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
- &str_false);
+ pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
+ &str_false);
pjsip_msg_add_hdr(tdata->msg, hdr);
-
+
/* Send answer */
status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata),
- tdata);
+ tdata);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Unable to create 2xx response to REFER -- %d\n", status);
return;
@@ -1594,9 +796,9 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
pjsip_hdr *hdr;
hdr = (pjsip_hdr*)
- pjsip_generic_string_hdr_create(inv->dlg->pool,
- &str_refer_sub,
- &str_true);
+ pjsip_generic_string_hdr_create(inv->dlg->pool,
+ &str_refer_sub,
+ &str_true);
pj_list_push_back(&hdr_list, hdr);
}
@@ -1606,7 +808,7 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
/* Create initial NOTIFY request */
status = pjsip_xfer_notify( sub, PJSIP_EVSUB_STATE_ACTIVE,
- 100, NULL, &tdata);
+ 100, NULL, &tdata);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Unable to create NOTIFY to REFER -- %d", status);
return;
@@ -1635,16 +837,16 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
_debug("UserAgent: Call doesn't exist!\n");
return;
}
-
+
AccountID accId = Manager::instance().getAccountFromCall(existing_call->getCallId());
CallID newCallId = Manager::instance().getNewCallID();
-
+
if(!Manager::instance().outgoingCall(accId, newCallId, tmp)) {
-
+
/* Notify xferer about the error (if we have subscription) */
if (sub) {
status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED,
- 500, NULL, &tdata);
+ 500, NULL, &tdata);
if (status != PJ_SUCCESS) {
_debug("UserAgent: Unable to create NOTIFY to REFER -- %d\n", status);
return;
@@ -1677,7 +879,7 @@ void UserAgent::onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
/* Put the invite_data in the subscription. */
pjsip_evsub_set_mod_data(sub, _mod.id,
- newCall);
+ newCall);
}
}
@@ -1692,11 +894,11 @@ void UserAgent::xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event)
if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
SIPCall *call;
- call = (SIPCall*) pjsip_evsub_get_mod_data(sub, getInstance()->getModId());
+ call = (SIPCall*) pjsip_evsub_get_mod_data(sub, _mod_ua.id);
if (!call)
return;
- pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
+ pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
call->setXferSub(NULL);
_debug("UserAgent: Xfer server subscription terminated\n");
diff --git a/src/useragent.h b/src/useragent.h
index b37257aeae33bd46bf9fae99a5a841fc655e013e..728b074cd70eb9fb14d09256cfabeaa36cd653c0 100644
--- a/src/useragent.h
+++ b/src/useragent.h
@@ -56,20 +56,8 @@ private:
pjsip_module _options_handler;
bool _useStun;
pj_str_t _stunHost;
- std::string _stunServer;
- bool validStunServer;
- /** Local Extern Address is the IP address seen by peers for SIP listener */
- std::string _localExternAddress;
- std::string _localIPAddress;
-
- /** Local Extern Port is the port seen by peers for SIP listener */
- unsigned int _localExternPort;
- unsigned int _localPort;
-
- /** For registration use only */
- int _regPort;
-
+
pj_thread_t *_thread;
static UserAgent *_current;
@@ -90,9 +78,6 @@ public:
void sipDestory();
- /** Create SIP UDP Listener */
- int createUDPServer();
-
/** Set whether it will use stun server */
void setStunServer(const char *server);
@@ -111,7 +96,6 @@ public:
bool loadSIPLocalIP();
- pj_status_t stunServerResolve();
pjsip_endpoint* getEndPoint() {return _endpt;}