diff --git a/src/Makefile.am b/src/Makefile.am
index 77070875f7a4720f0fa2aefd84cfa8714951ea02..014decc04471c121666f98edd6c1e36a028b80c2 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -13,10 +13,10 @@ endif
SUBDIRS = audio config gui $(ZEROCONFDIR)
-sflphoned_SOURCES = call.cpp eventthread.cpp main.cpp sipvoiplink.cpp voIPLink.cpp \
- sipcall.cpp managerimpl.cpp \
- observer.cpp \
- account.cpp sipaccount.cpp aixaccount.cpp accountcreator.cpp aixvoiplink.cpp
+sflphoned_SOURCES = eventthread.cpp main.cpp voIPLink.cpp \
+ managerimpl.cpp observer.cpp \
+ account.cpp sipaccount.cpp iaxaccount.cpp accountcreator.cpp iaxvoiplink.cpp \
+ sipvoiplink.cpp call.cpp sipcall.cpp
sflphoned_CXXFLAGS = -DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone\" $(ZEROCONFFLAGS)
@@ -38,6 +38,8 @@ libsflphone_la_LIBADD = \
libsflphone_la_SOURCES =
noinst_LTLIBRARIES = libsflphone.la
-noinst_HEADERS = managerimpl.h manager.h global.h observer.h eventthread.h sipvoiplink.h user_cfg.h \
- call.h voIPLink.h sipcall.h \
- account.h sipaccount.h aixaccount.h accountcreator.h aixvoiplink.h
+noinst_HEADERS = managerimpl.h manager.h global.h observer.h eventthread.h user_cfg.h \
+ voIPLink.h \
+ account.h sipaccount.h iaxaccount.h accountcreator.h iaxvoiplink.h \
+ sipvoiplink.h call.h sipcall.h
+
diff --git a/src/account.cpp b/src/account.cpp
index 20cb189460aa7348929e4deb2275e7750f949d36..a34da5011336d93a18f8daf92d65d0fdb5cefdbb 100644
--- a/src/account.cpp
+++ b/src/account.cpp
@@ -24,6 +24,7 @@ Account::Account(const AccountID& accountID) : _accountID(accountID)
_link = 0;
_shouldInitOnStart = false;
+ _shouldRegisterOnStart = false;
_enabled = false;
_registered = false;
}
diff --git a/src/account.h b/src/account.h
index 3cbdc738877ccd2e4aca98ac199cf8f78e9886a8..a2ab5f6932659818b34e4eb29bd87d3aa43342e4 100644
--- a/src/account.h
+++ b/src/account.h
@@ -28,6 +28,8 @@ typedef std::string AccountID;
#define AccountNULL ""
#define CONFIG_ACCOUNT_TYPE "Account.type"
#define CONFIG_ACCOUNT_ENABLE "Account.enable"
+#define CONFIG_ACCOUNT_AUTO_REGISTER "Account.autoregister"
+
/**
@author Yan Morin
@@ -39,13 +41,13 @@ class Account{
public:
Account(const AccountID& accountID);
- ~Account();
+ virtual ~Account();
/**
* Load the default properties for the account
*/
virtual void initConfig(Conf::ConfigTree& config) = 0;
-
+ virtual void loadConfig() = 0;
/**
* Get the voiplink pointer
@@ -77,6 +79,18 @@ public:
*/
virtual bool terminate() = 0;
+ /**
+ * Tell if we should init the account on start
+ * @return true if we must init the link
+ */
+ bool shouldInitOnStart() {return _shouldInitOnStart; }
+
+ /**
+ * Tell if we should init the account on start
+ * @return true if we must init the link
+ */
+ bool shouldRegisterOnStart() {return _shouldRegisterOnStart; }
+
private:
/**
* Create a unique voIPLink() depending on the protocol
@@ -102,6 +116,12 @@ protected:
*/
bool _shouldInitOnStart;
+ /**
+ * Tells if we should register automatically on startup
+ * Modified by the configuration
+ */
+ bool _shouldRegisterOnStart;
+
/**
* Tells if the link is enabled or not
* Modified by init/terminate
diff --git a/src/accountcreator.cpp b/src/accountcreator.cpp
index b013d09bfff229ebd3d793dbf8a3ad8d1d86399a..4bb4404c06320f1bab366a37e8df6e88889a55a2 100644
--- a/src/accountcreator.cpp
+++ b/src/accountcreator.cpp
@@ -18,7 +18,7 @@
*/
#include "accountcreator.h"
#include "sipaccount.h"
-#include "aixaccount.h"
+#include "iaxaccount.h"
AccountCreator::AccountCreator()
{
@@ -37,8 +37,8 @@ AccountCreator::createAccount(AccountType type, AccountID accountID)
return new SIPAccount(accountID);
break;
- case AIX_ACCOUNT:
- return new AIXAccount(accountID);
+ case IAX_ACCOUNT:
+ return new IAXAccount(accountID);
break;
}
return 0;
diff --git a/src/accountcreator.h b/src/accountcreator.h
index b38d78b0263a3d397909f1e84b415c2ec61f72bc..cb156358b19e136a9ed540eccce5bf12299572e8 100644
--- a/src/accountcreator.h
+++ b/src/accountcreator.h
@@ -33,7 +33,7 @@ public:
/**
* Public account type
*/
- enum AccountType {SIP_ACCOUNT, AIX_ACCOUNT };
+ enum AccountType {SIP_ACCOUNT, IAX_ACCOUNT };
/**
* Create a new account or null
diff --git a/src/aixaccount.cpp b/src/aixaccount.cpp
deleted file mode 100644
index f918a21fcc7aaa681ee678000cc9c453b48e9fa5..0000000000000000000000000000000000000000
--- a/src/aixaccount.cpp
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (C) 2006 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@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 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#include "aixaccount.h"
-#include "aixvoiplink.h"
-
-AIXAccount::AIXAccount(const AccountID& accountID)
- : Account(accountID)
-{
- createVoIPLink();
-}
-
-
-AIXAccount::~AIXAccount()
-{
-}
-
-/* virtual Account function implementation */
-bool
-AIXAccount::createVoIPLink()
-{
- if (!_link) {
- _link = new AIXVoIPLink();
- }
- return (_link != 0 ? true : false);
-}
-
-bool
-AIXAccount::registerAccount()
-{
- if (_link && !_registered) {
- _registered = (_link->setRegister() >= 0) ? true : false;
- }
- return _registered;
-}
-
-bool
-AIXAccount::unregisterAccount()
-{
- if (_link && _registered) {
- _registered = (_link->setUnregister() == 0) ? false : true;
- }
- return !_registered;
-}
-
-bool
-AIXAccount::init()
-{
- if (_link && !_enabled) {
- _link->init();
- _enabled = true;
- return true;
- }
- return false;
-}
-
-bool
-AIXAccount::terminate()
-{
- if (_link && _enabled) {
- _link->terminate();
- _enabled = false;
- return true;
- }
- return false;
-}
-
-void
-AIXAccount::initConfig(Conf::ConfigTree& config)
-{
- std::string section(_accountID);
- std::string type_str("string");
- std::string type_int("int");
-
- config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_TYPE, "AIX", type_str));
- config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_ENABLE, "1", type_int));
- config.addConfigTreeItem(section, Conf::ConfigTreeItem("AIX.Proxy", "", type_str));
-
-}
-
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index d4af191490db6293444d865050ab1d0b36bb3e3a..98d0fbc59d4e91d0720e8644cef1b7ba7e39c5c0 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -49,7 +49,7 @@ AudioRtp::~AudioRtp (void) {
}
int
-AudioRtp::createNewSession (SipCall *ca) {
+AudioRtp::createNewSession (SIPCall *ca) {
ost::MutexLock m(_threadMutex);
// something should stop the thread before...
@@ -90,7 +90,7 @@ AudioRtp::closeRtpSession () {
////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
-AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioLayer* driver, bool sym) {
+AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, AudioLayer* driver, bool sym) {
setCancel(cancelDeferred);
time = new ost::Time();
_ca = sipcall;
@@ -108,7 +108,7 @@ AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioLayer* driver, bool sym) {
// TODO: Change bind address according to user settings.
// TODO: this should be the local ip not the external (router) IP
- std::string localipConfig = "0.0.0.0"; // _ca->getLocalIp();
+ std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
ost::InetHostAddress local_ip(localipConfig.c_str());
//_debug("AudioRtpRTX ctor : Local IP:port %s:%d\tsymmetric:%d\n", local_ip.getHostname(), _ca->getLocalAudioPort(), _sym);
@@ -159,9 +159,9 @@ AudioRtpRTX::initAudioRtpSession (void)
if (_ca == 0) { return; }
//_debug("Init audio RTP session\n");
- ost::InetHostAddress remote_ip(_ca->getRemoteSdpAudioIp());
+ ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
if (!remote_ip) {
- _debug("AudioRTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteSdpAudioIp());
+ _debug("AudioRTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
return;
}
@@ -178,12 +178,12 @@ AudioRtpRTX::initAudioRtpSession (void)
}
if (!_sym) {
- if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort()) ) {
- _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getLocalAudioPort());
+ if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteAudioPort()) ) {
+ _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
return;
}
- if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
- _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteSdpAudioPort());
+ if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
+ _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
return;
}
@@ -202,7 +202,7 @@ AudioRtpRTX::initAudioRtpSession (void)
//_debug("AudioRTP Thread: Added session destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
- if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
+ if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
return;
}
@@ -451,6 +451,7 @@ AudioRtpRTX::run () {
audiolayer->flushMic();
audiolayer->startStream();
_start.post();
+ _debug("AudioRTP Start\n");
while (!testCancel()) {
////////////////////////////
// Send session
@@ -471,11 +472,11 @@ AudioRtpRTX::run () {
audiolayer->stopStream();
} catch(std::exception &e) {
_start.post();
- _debug("AudioRTP Thread, run: %s\n", e.what());
+ _debug("AudioRTP Stop: %s\n", e.what());
throw;
} catch(...) {
_start.post();
- _debugException("AudioRTP Thread, run()");
+ _debugException("AudioRTP Stop");
throw;
}
delete [] data_for_speakers_stereo; data_for_speakers_stereo = 0;
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index aab5eaa425cd3b4247c085bca519ee82394eaf75..5a787ee802d905efe1819a3254cec1ae55682756 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -29,21 +29,21 @@
#define RTP_FRAMES2SEND 160
class AudioLayer;
-class SipCall;
+class SIPCall;
///////////////////////////////////////////////////////////////////////////////
// Two pair of sockets
///////////////////////////////////////////////////////////////////////////////
class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
public:
- AudioRtpRTX (SipCall *, AudioLayer*, bool);
+ AudioRtpRTX (SIPCall *, AudioLayer*, bool);
~AudioRtpRTX();
ost::Time *time; // For incoming call notification
virtual void run ();
private:
- SipCall* _ca;
+ SIPCall* _ca;
AudioLayer* _audioDevice;
ost::RTPSession *_sessionSend;
ost::RTPSession *_sessionRecv;
@@ -72,7 +72,7 @@ public:
AudioRtp();
~AudioRtp();
- int createNewSession (SipCall *);
+ int createNewSession (SIPCall *);
void closeRtpSession ();
private:
diff --git a/src/call.cpp b/src/call.cpp
index 38332db002c852c0c406aab9565f77a8f9a0400e..f483ed883a3b90104fc44a5bca4885ad6826c639 100644
--- a/src/call.cpp
+++ b/src/call.cpp
@@ -1,271 +1,61 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+/*
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
+ * Author : Laurielle Lea <laurielle.lea@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 2 of the License, or
* (at your option) any later version.
- *
+ *
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
- *
+ *
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-
-#include <iostream>
-
#include "call.h"
-#include "voIPLink.h"
-
-Call::Call (CALLID id, Call::CallType type, VoIPLink* voiplink)
-{
- _state = NotExist;
- _type = Null;
- _id = id;
- _type = type;
- _voIPLink = voiplink;
- _flagNotAnswered = true;
-
- switch (_type) {
- case Outgoing:
- _voIPLink->newOutgoingCall(_id);
- break;
- case Incoming:
- _voIPLink->newIncomingCall(_id);
- break;
- default:
- break;
- }
-}
-Call::~Call (void)
+Call::Call(const CallID& id, Call::CallType type) : _id(id), _type(type)
{
+ _connectionState = Call::Disconnected;
+ _callState = Call::Inactive;
}
-CALLID
-Call::getId (void)
-{
- return _id;
-}
-void
-Call::setId (CALLID id)
+Call::~Call()
{
- _id = id;
}
void
-Call::setVoIPLink (VoIPLink* voIPLink)
-{
- _voIPLink = voIPLink;
-}
-
-VoIPLink*
-Call::getVoIPLink (void)
-{
- return _voIPLink;
-}
-
-std::string
-Call::getCallerIdName (void)
+Call::setConnectionState(ConnectionState state)
{
- return _callerIdName;
+ ost::MutexLock m(_callMutex);
+ _connectionState = state;
}
-void
-Call::setCallerIdName (const std::string& callerId_name)
+Call::ConnectionState
+Call::getConnectionState()
{
- _callerIdName = callerId_name;
+ ost::MutexLock m(_callMutex);
+ return _connectionState;
}
-std::string
-Call::getCallerIdNumber (void)
-{
- return _callerIdNumber;
-}
void
-Call::setCallerIdNumber (const std::string& callerId_number)
+Call::setState(CallState state)
{
- _callerIdNumber = callerId_number;
+ ost::MutexLock m(_callMutex);
+ _callState = state;
}
Call::CallState
-Call::getState (void)
-{
- return _state;
-}
-
-void
-Call::setState (Call::CallState state)
-{
- _state = state;
-}
-
-Call::CallType
-Call::getType (void)
+Call::getState()
{
- return _type;
+ ost::MutexLock m(_callMutex);
+ return _callState;
}
-void
-Call::setType (Call::CallType type)
-{
- _type = type;
-}
-
-bool
-Call::isBusy (void)
-{
- if (isAnswered() or isOffHold() or isOnMute() or isOffMute()) {
- return true;
- } else {
- return false;
- }
-}
-bool
-Call::isOnHold (void)
-{
- return (_state == OnHold) ? true : false;
-}
-
-bool
-Call::isOffHold (void)
-{
- return (_state == OffHold) ? true : false;
-}
-
-bool
-Call::isOnMute (void)
-{
- return (_state == MuteOn) ? true : false;
-}
-
-bool
-Call::isOffMute (void)
-{
- return (_state == MuteOff) ? true : false;
-}
-
-bool
-Call::isTransfered (void)
-{
- return (_state == Transfered) ? true : false;
-}
-
-bool
-Call::isHungup (void)
-{
- return (_state == Hungup) ? true : false;
-}
-
-bool
-Call::isRinging (void)
-{
- return (_state == Ringing) ? true : false;
-}
-
-bool
-Call::isRefused (void)
-{
- return (_state == Refused) ? true : false;
-}
-
-
-bool
-Call::isAnswered (void)
-{
- return (_state == Answered) ? true : false;
-}
-
-bool
-Call::isNotAnswered (void)
-{
- return (_state == Error || _state == NotExist || _state == Busy) ? true : false;
-}
-
-bool
-Call::isProgressing (void)
-{
- return (_state == Progressing) ? true : false;
-}
-
-bool
-Call::isOutgoingType (void)
-{
- return (_type == Outgoing) ? true : false;
-}
-
-bool
-Call::isIncomingType (void)
-{
- return (_type == Incoming) ? true : false;
-}
-
-int
-Call::outgoingCall(const std::string& to)
-{
- return _voIPLink->outgoingInvite(_id, to);
-}
-
-int
-Call::hangup (void)
-{
- int i = _voIPLink->hangup(_id);
- setState(Hungup);
- return i;
-}
-
-int
-Call::cancel (void)
-{
- int i = _voIPLink->cancel(_id);
- setState(Hungup);
- return i;
-}
-
-int
-Call::answer (void)
-{
- _flagNotAnswered = false;
- int i = _voIPLink->answer(_id);
- setState(Answered);
- return i;
-}
-
-int
-Call::onHold (void)
-{
- int i = _voIPLink->onhold(_id);
- setState(OnHold);
- return i;
-}
-
-int
-Call::offHold (void)
-{
- int i = _voIPLink->offhold(_id);
- setState(OffHold);
- return i;
-}
-
-int
-Call::transfer (const std::string& to)
-{
- int i = _voIPLink->transfer(_id, to);
- setState(Transfered);
- return i;
-}
-
-int
-Call::refuse (void)
-{
- int i = _voIPLink->refuse(_id);
- return i;
-}
diff --git a/src/call.h b/src/call.h
index d6cf60513f750c296819cf0248313aabf48e5399..1f260825e7dbc7b3adc34568b393f9dfbac1c4f7 100644
--- a/src/call.h
+++ b/src/call.h
@@ -1,120 +1,109 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+/*
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
+ * Author : Laurielle Lea <laurielle.lea@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 2 of the License, or
* (at your option) any later version.
- *
+ *
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
- *
+ *
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-
-#ifndef __CALL_H__
-#define __CALL_H__
+#ifndef CALL_H
+#define CALL_H
#include <string>
+#include <cc++/thread.h> // for mutex
+
+//TODO: remove this, it's only for call ID
+typedef std::string CallID;
+
+/**
+ @author Yan Morin <yan.morin@gmail.com>
+ A call is the base classes for protocol-based calls
+*/
+class Call{
+public:
+ enum CallType {Incoming, Outgoing};
+ enum ConnectionState {Disconnected, Trying, Progressing, Ringing, Connected };
+ enum CallState {Inactive, Active, Hold, Busy, Refused, Error};
-typedef unsigned int CALLID;
+ /**
+ * Constructor of a call
+ * @param id Unique identifier of the call
+ * @param type set definitely this call as incoming/outgoing
+ */
+ Call(const CallID& id, Call::CallType type);
+ virtual ~Call();
+ /**
+ * Return a reference on the call id
+ * @return call id
+ */
+ CallID& getCallId() {return _id; }
-class VoIPLink;
+ /**
+ * Set the peer number (destination on outgoing)
+ * not protected by mutex (when created)
+ * @param number peer number
+ */
+ void setPeerNumber(const std::string& number) { _peerNumber = number; }
+ const std::string& getPeerNumber() { return _peerNumber; }
-class Call {
-public:
- enum CallType {
- Null = 0,
- Incoming,
- Outgoing
- };
+ /**
+ * Set the peer name (caller in ingoing)
+ * not protected by mutex (when created)
+ * @param number peer number
+ */
+ void setPeerName(const std::string& name) { _peerName = name; }
+ const std::string& getPeerName() { return _peerName; }
+
+ /**
+ * Set the connection state of the call (protected by mutex)
+ */
+ void setConnectionState(ConnectionState state);
+ /**
+ * get the connection state of the call (protected by mutex)
+ */
+ ConnectionState getConnectionState();
- enum CallState {
- NotExist = 0,
- Busy,
- OnHold,
- OffHold,
- MuteOn,
- MuteOff,
- Transfered,
- Hungup,
- Answered,
- Ringing,
- Progressing,
- Refused, // for refuse incoming ringing call
- Error // when a error occur
- };
+ /**
+ * Set the state of the call (protected by mutex)
+ */
+ void setState(CallState state);
+ /**
+ * get the call state of the call (protected by mutex)
+ */
+ CallState getState();
- // Constructor
- Call(CALLID id, CallType type, VoIPLink* voiplink);
- // Destructor
- ~Call(void);
-
+protected:
+ /** Protect every attribute that can be changed by two threads */
+ ost::Mutex _callMutex;
- // Handle call-id
- CALLID getId (void);
- void setId (CALLID id);
-
- // Accessor and modifior of VoIPLink
- VoIPLink* getVoIPLink(void);
- void setVoIPLink (VoIPLink* voIPLink);
-
- // Handle id name and id number
- std::string getCallerIdName (void);
- void setCallerIdName (const std::string& callerId_name);
- std::string getCallerIdNumber (void);
- void setCallerIdNumber (const std::string& callerId_number);
-
- // Handle state
- CallState getState (void);
- void setState (CallState state);
-
- // Handle type of call (incoming or outoing)
- enum CallType getType (void);
- void setType (enum CallType type);
+private:
+ /** Unique ID of the call */
+ CallID _id;
- bool isNotAnswered(void);
- bool isBusy (void);
- bool isOnHold (void);
- bool isOffHold (void);
- bool isOnMute (void);
- bool isOffMute (void);
- bool isTransfered (void);
- bool isHungup (void);
- bool isRinging (void);
- bool isRefused (void);
- bool isAnswered (void);
- bool isProgressing (void);
- bool isOutgoingType (void);
- bool isIncomingType (void);
-
- int outgoingCall (const std::string& to);
- int hangup (void);
- int cancel (void);
- int answer (void);
- int onHold (void);
- int offHold (void);
- int transfer (const std::string& to);
- int refuse (void);
- void setFlagNotAnswered(bool value) { _flagNotAnswered = value; }
- bool getFlagNotAnswered() const { return _flagNotAnswered; }
+ /** Type of the call */
+ CallType _type;
+ /** Disconnected/Progressing/Trying/Ringing/Connected */
+ ConnectionState _connectionState;
+ /** Inactive/Active/Hold/Busy/Refused/Error */
+ CallState _callState;
-private:
- VoIPLink *_voIPLink;
- CALLID _id;
- enum CallState _state;
- enum CallType _type;
- std::string _callerIdName;
- std::string _callerIdNumber;
+ /** Name of the peer */
+ std::string _peerName;
- bool _flagNotAnswered;
+ /** Number of the peer */
+ std::string _peerNumber;
};
-#endif // __CALL_H__
+#endif
diff --git a/src/eventthread.cpp b/src/eventthread.cpp
index 6e445ab232c3adda875de21ae6842ca0743dd7f9..7edc957062885343e4a99961ca21c25cafe1150e 100644
--- a/src/eventthread.cpp
+++ b/src/eventthread.cpp
@@ -19,11 +19,11 @@
*/
#include "eventthread.h"
-#include "sipvoiplink.h"
+#include "voIPLink.h"
-EventThread::EventThread (SipVoIPLink* sip) : Thread ()
+EventThread::EventThread (VoIPLink* link) : Thread ()
{
- _sipthread = sip;
+ _linkthread = link;
setCancel(cancelDeferred);
}
@@ -39,7 +39,7 @@ void
EventThread::run (void)
{
while(!testCancel()) {
- _sipthread->getEvent();
+ _linkthread->getEvent();
}
}
diff --git a/src/eventthread.h b/src/eventthread.h
index 7508b1cfc33effb6911cb74ec83c028efaf7ff42..a9eba59aeaeecd8a2c22ccbf2bfc130b8797b606 100644
--- a/src/eventthread.h
+++ b/src/eventthread.h
@@ -1,4 +1,4 @@
-/**
+/*
* Copyright (C) 2004-2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
@@ -23,16 +23,23 @@
#include <cc++/thread.h>
-
-class SipVoIPLink;
+class VoIPLink;
+/**
+ * General thread to listen events continuously
+ */
class EventThread : public ost::Thread {
public:
- EventThread (SipVoIPLink*);
+ /**
+ * Build a thread that call getEvents
+ */
+ EventThread (VoIPLink*);
~EventThread (void);
virtual void run ();
+
private:
- SipVoIPLink* _sipthread;
+ /** VoIPLink is the object being called by getEvents() method */
+ VoIPLink* _linkthread;
};
#endif // __EVENT_THREAD_H__
diff --git a/src/gui/guiframework.cpp b/src/gui/guiframework.cpp
index d0a84605d0c85e73cc18b7852f3fb1c3128f88a0..49150ab4950f07992ab258dad3d5d87a3b3dd9cf 100644
--- a/src/gui/guiframework.cpp
+++ b/src/gui/guiframework.cpp
@@ -1,5 +1,5 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+/*
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
@@ -30,10 +30,10 @@ GuiFramework::GuiFramework ()
GuiFramework::~GuiFramework (void) {}
-int
-GuiFramework::outgoingCall(const std::string&, const std::string& to)
+bool
+GuiFramework::outgoingCall(const std::string& account, const CallID& id, const std::string& to)
{
- return Manager::instance().outgoingCall(to);
+ return Manager::instance().outgoingCall(account, id, to);
}
bool
@@ -42,64 +42,40 @@ GuiFramework::sendTextMessage(const std::string& account, const std::string& to,
return Manager::instance().sendTextMessage(account, to, message);
}
-int
-GuiFramework::hangupCall (CALLID id)
+bool
+GuiFramework::hangupCall (const CallID& id)
{
- if (Manager::instance().hangupCall(id) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().hangupCall(id);
}
-int
-GuiFramework::cancelCall (CALLID id)
+bool
+GuiFramework::cancelCall (const CallID& id)
{
- if (Manager::instance().cancelCall(id) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().cancelCall(id);
}
-int
-GuiFramework::answerCall (CALLID id)
+bool
+GuiFramework::answerCall(const CallID& id)
{
- if (Manager::instance().answerCall(id) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().answerCall(id);
}
-int
-GuiFramework::onHoldCall (CALLID id)
+bool
+GuiFramework::onHoldCall(const CallID& id)
{
- if (Manager::instance().onHoldCall(id) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().onHoldCall(id);
}
-int
-GuiFramework::offHoldCall (CALLID id)
+bool
+GuiFramework::offHoldCall(const CallID& id)
{
- if (Manager::instance().offHoldCall(id) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().offHoldCall(id);
}
-int
-GuiFramework::transferCall (CALLID id, const std::string& to)
+bool
+GuiFramework::transferCall(const CallID& id, const std::string& to)
{
- if (Manager::instance().transferCall(id, to) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().transferCall(id, to);
}
void
@@ -107,20 +83,17 @@ GuiFramework::mute()
{
Manager::instance().mute();
}
+
void
GuiFramework::unmute()
{
Manager::instance().unmute();
}
-int
-GuiFramework::refuseCall (CALLID id)
+bool
+GuiFramework::refuseCall (const CallID& id)
{
- if (Manager::instance().refuseCall(id) == 0) {
- return 1;
- } else {
- return 0;
- }
+ return Manager::instance().refuseCall(id);
}
bool
@@ -130,19 +103,19 @@ GuiFramework::saveConfig (void)
}
bool
-GuiFramework::registerVoIPLink (void)
+GuiFramework::registerVoIPLink(const AccountID& accountId)
{
- return Manager::instance().registerVoIPLink();
+ return Manager::instance().registerVoIPLink(accountId);
}
bool
-GuiFramework::unregisterVoIPLink (void)
+GuiFramework::unregisterVoIPLink (const AccountID& accountId)
{
- return Manager::instance().unregisterVoIPLink();
+ return Manager::instance().unregisterVoIPLink(accountId);
}
bool
-GuiFramework::sendDtmf (CALLID id, char code)
+GuiFramework::sendDtmf (const CallID& id, char code)
{
return Manager::instance().sendDtmf(id, code);
}
@@ -202,7 +175,7 @@ GuiFramework::getCallStatus(const std::string& sequenceId)
return Manager::instance().getCallStatus(sequenceId);
}
-CALLID
+const CallID&
GuiFramework::getCurrentId()
{
return Manager::instance().getCurrentCallId();
diff --git a/src/gui/guiframework.h b/src/gui/guiframework.h
index 18242b499899f0ea4dae66cb805661143e0385dc..9a45ac5ca019e3b79cbf1bc070780bc6076c1cf0 100644
--- a/src/gui/guiframework.h
+++ b/src/gui/guiframework.h
@@ -1,5 +1,5 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+/*
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
@@ -26,7 +26,8 @@
#include <string>
#include "server/argtokenizer.h"
#include "../observer.h"
-#include "../call.h"
+#include "../call.h" // for callid2
+#include "../account.h" // for account
class GuiFramework {
public:
@@ -34,43 +35,45 @@ public:
virtual ~GuiFramework (void);
/* Parent class to child class */
- virtual int incomingCall (CALLID id, const std::string& accountId, const std::string& from) = 0;
- virtual void peerAnsweredCall (CALLID id) = 0;
- virtual void peerRingingCall (CALLID id) = 0;
- virtual void peerHungupCall (CALLID id) = 0;
- virtual void incomingMessage(const std::string& message) = 0;
+ virtual bool incomingCall (const AccountID& accountId, const CallID& id, const std::string& from) = 0;
+
+ virtual void peerAnsweredCall (const CallID& id) = 0;
+ virtual void peerRingingCall (const CallID& id) = 0;
+ virtual void peerHungupCall (const CallID& id) = 0;
+ virtual void incomingMessage(const AccountID& accountId, const std::string& message) = 0;
virtual void displayStatus (const std::string& status) = 0;
virtual void displayConfigError (const std::string& error) = 0;
- virtual void displayTextMessage (CALLID id, const std::string& message) = 0;
- virtual void displayErrorText (CALLID id, const std::string& message) = 0;
+ virtual void displayTextMessage (const CallID& id, const std::string& message) = 0;
+ virtual void displayErrorText (const CallID& id, const std::string& message) = 0;
virtual void displayError (const std::string& error) = 0;
virtual void startVoiceMessageNotification (void) {}
virtual void stopVoiceMessageNotification (void) {}
- virtual void sendVoiceNbMessage(const std::string& nb_msg) = 0;
+ virtual void sendVoiceNbMessage(const AccountID& accountId, const std::string& nb_msg) = 0;
virtual void setup() = 0;
virtual void sendMessage(const std::string& code, const std::string& seqId, TokenList& arg) = 0;
virtual void sendCallMessage(const std::string& code,
- const std::string& sequenceId, CALLID id, TokenList arg) = 0;
- virtual void sendRegistrationState(bool state) = 0;
- virtual void callFailure(CALLID id) = 0;
+ const std::string& sequenceId, const CallID& id, TokenList arg) = 0;
+ virtual void sendRegistrationState(const AccountID& accountid, bool state) = 0;
+ virtual void callFailure(const CallID& id) = 0;
/* Child class to parent class */
- int outgoingCall(const std::string& account, const std::string& to);
- bool sendTextMessage(const std::string& account, const std::string& to, const std::string& message);
- int hangupCall (CALLID id);
- int cancelCall (CALLID id);
- int answerCall (CALLID id);
- int onHoldCall (CALLID id);
- int offHoldCall (CALLID id);
- int transferCall (CALLID id, const std::string& to);
+ bool outgoingCall(const AccountID& account, const CallID& id, const std::string& to);
+ bool answerCall(const CallID& id);
+
+ bool sendTextMessage(const AccountID& accountId, const std::string& to, const std::string& message);
+ bool hangupCall (const CallID& id);
+ bool cancelCall (const CallID& id);
+ bool onHoldCall (const CallID& id);
+ bool offHoldCall (const CallID& id);
+ bool transferCall (const CallID& id, const std::string& to);
void mute ();
void unmute ();
- int refuseCall (CALLID id);
+ bool refuseCall (const CallID& id);
bool saveConfig(void);
- bool registerVoIPLink(void);
- bool unregisterVoIPLink(void);
- bool sendDtmf (CALLID id, char code);
+ bool registerVoIPLink(const AccountID& accountId);
+ bool unregisterVoIPLink(const AccountID& accountId);
+ bool sendDtmf (const CallID& id, char code);
bool playDtmf (char code);
bool playTone ();
bool stopTone ();
@@ -92,7 +95,7 @@ public:
bool setSwitch(const std::string& switchName);
bool hasLoadedSetup();
- CALLID getCurrentId();
+ const CallID& getCurrentId();
bool getRegistrationState(std::string& stateCode, std::string& stateMessage);
protected:
diff --git a/src/gui/qt/ConfigurationPanel.ui.h b/src/gui/qt/ConfigurationPanel.ui.h
index a6ee2f33799eaafaa6762cbc1403c3c9a83a6ee4..fe7c445f705fae42709610012e54c487184114fb 100644
--- a/src/gui/qt/ConfigurationPanel.ui.h
+++ b/src/gui/qt/ConfigurationPanel.ui.h
@@ -1,4 +1,4 @@
-/**
+/*
* Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Jean-Philippe Barrette-LaPierre
@@ -133,43 +133,41 @@ ConfigurationPanel::generate()
.get(AUDIO_SECTION,
AUDIO_RINGTONE));
+ QString account = ACCOUNT_DEFAULT_NAME;
+ QString type = ConfigurationManager::instance().get(account, ACCOUNT_TYPE);
+ QString active = ConfigurationManager::instance().get(account, ACCOUNT_ENABLE);
+
// For signalisations tab
+ autoregister->setChecked(ConfigurationManager::instance()
+ .get(account,ACCOUNT_AUTO_REGISTER).toUInt());
+
fullName->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_FULL_NAME));
+ .get(account,SIGNALISATION_FULL_NAME));
userPart->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_USER_PART));
+ .get(account,SIGNALISATION_USER_PART));
username->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_AUTH_USER_NAME));
+ .get(account,SIGNALISATION_AUTH_USER_NAME));
password->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_PASSWORD));
+ .get(account,SIGNALISATION_PASSWORD));
hostPart->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_HOST_PART));
+ .get(account,SIGNALISATION_HOST_PART));
sipproxy->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_PROXY));
- autoregister->setChecked(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_AUTO_REGISTER).toUInt());
+ .get(account,SIGNALISATION_PROXY));
+ STUNserver->setText(ConfigurationManager::instance()
+ .get(account,SIGNALISATION_STUN_SERVER));
+ ((QRadioButton*)stunButtonGroup->find(ConfigurationManager::instance()
+ .get(account,SIGNALISATION_USE_STUN).toUInt()))->setChecked(true);
+
+ sendDTMFas->setCurrentItem(ConfigurationManager::instance()
+ .get(SIGNALISATION_SECTION,
+ SIGNALISATION_SEND_DTMF_AS).toUInt());
playTones->setChecked(ConfigurationManager::instance()
.get(SIGNALISATION_SECTION,
SIGNALISATION_PLAY_TONES).toUInt());
pulseLength->setValue(ConfigurationManager::instance()
.get(SIGNALISATION_SECTION,
SIGNALISATION_PULSE_LENGTH).toUInt());
- sendDTMFas->setCurrentItem(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_SEND_DTMF_AS).toUInt());
- STUNserver->setText(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_STUN_SERVER));
- ((QRadioButton*)stunButtonGroup->find(ConfigurationManager::instance()
- .get(SIGNALISATION_SECTION,
- SIGNALISATION_USE_STUN).toUInt()))->setChecked(true);
+
QRadioButton* device =
static_cast< QRadioButton * >(DriverChoice->find(ConfigurationManager::instance()
.get(AUDIO_SECTION,
@@ -185,27 +183,32 @@ ConfigurationPanel::generate()
// For saving settings at application 'save'
void ConfigurationPanel::saveSlot()
{
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ QString account = ACCOUNT_DEFAULT_NAME;
+ ConfigurationManager::instance().set(account,
SIGNALISATION_FULL_NAME,
fullName->text());
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ ConfigurationManager::instance().set(account,
SIGNALISATION_USER_PART,
userPart->text());
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ ConfigurationManager::instance().set(account,
SIGNALISATION_AUTH_USER_NAME,
username->text());
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ ConfigurationManager::instance().set(account,
SIGNALISATION_PASSWORD,
password->text());
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ ConfigurationManager::instance().set(account,
SIGNALISATION_HOST_PART,
hostPart->text());
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ ConfigurationManager::instance().set(account,
SIGNALISATION_PROXY,
sipproxy->text());
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
- SIGNALISATION_AUTO_REGISTER,
+ ConfigurationManager::instance().set(account,
+ ACCOUNT_AUTO_REGISTER,
QString::number(autoregister->isChecked()));
+ ConfigurationManager::instance().set(account,
+ SIGNALISATION_STUN_SERVER,
+ STUNserver->text());
+
ConfigurationManager::instance().set(SIGNALISATION_SECTION,
SIGNALISATION_PULSE_LENGTH,
QString::number(pulseLength->value()));
@@ -215,9 +218,6 @@ void ConfigurationPanel::saveSlot()
ConfigurationManager::instance().set(SIGNALISATION_SECTION,
SIGNALISATION_SEND_DTMF_AS,
QString::number(sendDTMFas->currentItem()));
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
- SIGNALISATION_STUN_SERVER,
- STUNserver->text());
if (codec1->currentText() != NULL) {
ConfigurationManager::instance().set(AUDIO_SECTION,
@@ -292,7 +292,8 @@ void ConfigurationPanel::changeTabSlot()
void ConfigurationPanel::useStunSlot(int id)
{
- ConfigurationManager::instance().set(SIGNALISATION_SECTION,
+ QString account = ACCOUNT_DEFAULT_NAME;
+ ConfigurationManager::instance().set(account,
SIGNALISATION_USE_STUN,
QString::number(id));
}
diff --git a/src/gui/qt/Session.cpp b/src/gui/qt/Session.cpp
index 02d80a6e1eb0502874b2eeb6de3649fe29aaff81..9491565ac684411d26cf1f1d2aab5cb1e14cd492 100644
--- a/src/gui/qt/Session.cpp
+++ b/src/gui/qt/Session.cpp
@@ -176,6 +176,6 @@ Session::getAccount(const QString &name) const
Account
Session::getDefaultAccount() const
{
- return Account(mId, QString("mydefaultaccount"));
+ return Account(mId, QString("SIP0"));
}
diff --git a/src/gui/qt/globals.h b/src/gui/qt/globals.h
index 6d446339806df99f15f6c0c004cf1cdcbe6656fc..80565bebd616ed9a650761af2180689955c38452 100644
--- a/src/gui/qt/globals.h
+++ b/src/gui/qt/globals.h
@@ -32,7 +32,6 @@
#define AUDIO_CODEC3 "Codecs.codec3"
#define AUDIO_RINGTONE "Rings.ringChoice"
-
#define SIGNALISATION_SECTION "VoIPLink"
#define SIGNALISATION_FULL_NAME "SIP.fullName"
#define SIGNALISATION_USER_PART "SIP.userPart"
@@ -40,12 +39,16 @@
#define SIGNALISATION_PASSWORD "SIP.password"
#define SIGNALISATION_HOST_PART "SIP.hostPart"
#define SIGNALISATION_PROXY "SIP.proxy"
-#define SIGNALISATION_AUTO_REGISTER "SIP.autoregister"
+#define SIGNALISATION_STUN_SERVER "STUN.STUNserver"
+#define SIGNALISATION_USE_STUN "STUN.useStun"
#define SIGNALISATION_PLAY_TONES "DTMF.playTones"
#define SIGNALISATION_PULSE_LENGTH "DTMF.pulseLength"
#define SIGNALISATION_SEND_DTMF_AS "DTMF.sendDTMFas"
-#define SIGNALISATION_STUN_SERVER "STUN.STUNserver"
-#define SIGNALISATION_USE_STUN "STUN.useStun"
+
+#define ACCOUNT_DEFAULT_NAME "SIP0"
+#define ACCOUNT_TYPE "Account.type"
+#define ACCOUNT_ENABLE "Account.enable"
+#define ACCOUNT_AUTO_REGISTER "Account.autoregister"
#define PREFERENCES_SECTION "Preferences"
#define PREFERENCES_THEME "Themes.skinChoice"
diff --git a/src/gui/server/guiserverimpl.cpp b/src/gui/server/guiserverimpl.cpp
index 50f99f8796293bfa5bbf262223e47176eb38ac3d..a050875fb8410b848879c27294c3159534042481 100644
--- a/src/gui/server/guiserverimpl.cpp
+++ b/src/gui/server/guiserverimpl.cpp
@@ -48,12 +48,13 @@ GUIServerImpl::exec() {
* remove
*/
void
-GUIServerImpl::insertSubCall(CALLID id, SubCall& subCall) {
- _callMap[id] = subCall;
+GUIServerImpl::insertSubCall(const CallID& id, const CallID& seq) {
+
+ _callMap[id] = seq;
}
void
-GUIServerImpl::removeSubCall(CALLID id) {
+GUIServerImpl::removeSubCall(const CallID& id) {
_callMap.erase(id);
}
@@ -61,32 +62,21 @@ GUIServerImpl::removeSubCall(CALLID id) {
* Retreive the sequenceId or send default sequenceId
*/
std::string
-GUIServerImpl::getSequenceIdFromId(CALLID id) {
+GUIServerImpl::getSequenceIdFromId(const CallID& id) {
CallMap::iterator iter = _callMap.find(id);
if (iter != _callMap.end()) {
- return iter->second.sequenceId();
+ return iter->second;
}
return _getEventsSequenceId;
}
-/**
- * Retreive the string callid from the id
- */
-std::string
-GUIServerImpl::getCallIdFromId(CALLID id) {
- CallMap::iterator iter = _callMap.find(id);
- if (iter != _callMap.end()) {
- return iter->second.callId();
- }
- throw std::runtime_error(_("No match for this id"));
-}
bool
GUIServerImpl::getCurrentCallId(std::string& callId) {
bool returnValue = false;
try {
- CALLID id = GuiFramework::getCurrentId();
- if (id!=0) {
- callId = getCallIdFromId(id);
+ CallID id = GuiFramework::getCurrentId();
+ if (id != "") {
+ callId = id;
returnValue = true;
}
} catch(...) {
@@ -95,19 +85,6 @@ GUIServerImpl::getCurrentCallId(std::string& callId) {
return returnValue;
}
-CALLID
-GUIServerImpl::getIdFromCallId(const std::string& callId)
-{
- CallMap::iterator iter = _callMap.begin();
- while (iter != _callMap.end()) {
- if (iter->second.callId()==callId) {
- return iter->first;
- }
- iter++;
- }
- throw std::runtime_error(_("No match for this CallId"));
-}
-
bool
GUIServerImpl::getEvents(const std::string& sequenceId)
{
@@ -165,96 +142,45 @@ GUIServerImpl::outgoingCall(const std::string& seq,
const std::string& callid,
const std::string& to)
{
- CALLID serverCallId = GuiFramework::outgoingCall(account, to);
- if ( serverCallId ) {
- SubCall subcall(seq, callid);
- insertSubCall(serverCallId, subcall);
- return true;
- } else {
- return false;
- }
+ insertSubCall(callid, seq);
+ return GuiFramework::outgoingCall(account, callid, to);
}
bool
GUIServerImpl::answerCall(const std::string& callId)
{
- try {
- CALLID id = getIdFromCallId(callId);
- if (GuiFramework::answerCall(id)) {
- return true;
- }
- } catch(...) {
- return false;
- }
- return false;
+ return GuiFramework::answerCall(callId);
}
bool
GUIServerImpl::refuseCall(const std::string& callId)
{
- try {
- CALLID id = getIdFromCallId(callId);
- if (GuiFramework::refuseCall(id)) {
- return true;
- }
- } catch(...) {
- return false;
- }
- return false;
+ return GuiFramework::refuseCall(callId);
}
bool
GUIServerImpl::transferCall(const std::string& callId, const std::string& to)
{
- try {
- CALLID id = getIdFromCallId(callId);
- if (GuiFramework::transferCall(id, to)) {
- return true;
- }
- } catch(...) {
- return false;
- }
- return false;
+ return GuiFramework::transferCall(callId, to);
}
bool
GUIServerImpl::holdCall(const std::string& callId)
{
- try {
- CALLID id = getIdFromCallId(callId);
- if (GuiFramework::onHoldCall(id)) {
- return true;
- }
- } catch(...) {
- return false;
- }
- return false;
+ return GuiFramework::onHoldCall(callId);
}
bool
GUIServerImpl::unholdCall(const std::string& callId)
{
- try {
- CALLID id = getIdFromCallId(callId);
- if (GuiFramework::offHoldCall(id)) {
- return true;
- }
- } catch(...) {
- return false;
- }
- return false;
+ return GuiFramework::offHoldCall(callId);
}
bool
GUIServerImpl::hangupCall(const std::string& callId)
{
- try {
- CALLID id = getIdFromCallId(callId);
- if (GuiFramework::hangupCall(id)) {
- _callMap.erase(id);
- return true;
- }
- } catch(...) {
- return false;
+ if ( GuiFramework::hangupCall(callId) ) {
+ removeSubCall(callId);
+ return true;
}
return false;
}
@@ -267,12 +193,10 @@ bool
GUIServerImpl::hangupAll()
{
bool result = true;
- CALLID id;
CallMap::iterator iter = _callMap.begin();
// try to hangup every call, even if one fail
while(iter!=_callMap.end()) {
- id = iter->first;
- if (!GuiFramework::hangupCall(id)) {
+ if (!GuiFramework::hangupCall(iter->first)) {
result = false;
}
iter++;
@@ -284,14 +208,7 @@ GUIServerImpl::hangupAll()
bool
GUIServerImpl::dtmfCall(const std::string& callId, const std::string& dtmfKey)
{
- try {
- CALLID id = getIdFromCallId(callId);
- char code = dtmfKey[0];
- return GuiFramework::sendDtmf(id, code);
- } catch(...) {
- return false;
- }
- return false;
+ return GuiFramework::sendDtmf(callId, dtmfKey[0]);
}
/**
@@ -307,62 +224,60 @@ GUIServerImpl::version()
}
-int
-GUIServerImpl::incomingCall (CALLID id, const std::string& accountId, const std::string& from)
+bool
+GUIServerImpl::incomingCall(const AccountID& accountId, const CallID& id, const std::string& from)
{
TokenList arg;
- std::ostringstream callId;
- callId << "s" << id;
- arg.push_back(callId.str());
+ arg.push_back(id);
arg.push_back(accountId);
arg.push_back(from);
arg.push_back("call");
- SubCall subcall(_getEventsSequenceId, callId.str());
-
- insertSubCall(id, subcall);
-
- _requestManager.sendResponse(ResponseMessage("001", _getEventsSequenceId,arg));
+ insertSubCall(id, _getEventsSequenceId);
+ _requestManager.sendResponse(ResponseMessage("001", _getEventsSequenceId, arg));
return 0;
}
void
-GUIServerImpl::incomingMessage(const std::string& message) {
- _requestManager.sendResponse(ResponseMessage("030", _getEventsSequenceId, message));
+GUIServerImpl::incomingMessage(const std::string& account, const std::string& message) {
+ TokenList arg;
+ arg.push_back(account);
+ arg.push_back(message);
+ _requestManager.sendResponse(ResponseMessage("030", _getEventsSequenceId, arg));
}
void
-GUIServerImpl::peerAnsweredCall (CALLID id)
+GUIServerImpl::peerAnsweredCall (const CallID& id)
{
CallMap::iterator iter = _callMap.find(id);
if ( iter != _callMap.end() ) {
- _requestManager.sendResponse(ResponseMessage("200", iter->second.sequenceId(), _("Established")));
+ _requestManager.sendResponse(ResponseMessage("200", iter->second, _("Established")));
}
}
void
-GUIServerImpl::peerRingingCall (CALLID id)
+GUIServerImpl::peerRingingCall (const CallID& id)
{
CallMap::iterator iter = _callMap.find(id);
if ( iter != _callMap.end() ) {
- _requestManager.sendResponse(ResponseMessage("151", iter->second.sequenceId(), _("Ringing")));
+ _requestManager.sendResponse(ResponseMessage("151", iter->second, _("Ringing")));
}
}
void
-GUIServerImpl::peerHungupCall (CALLID id)
+GUIServerImpl::peerHungupCall (const CallID& id)
{
CallMap::iterator iter = _callMap.find(id);
if ( iter != _callMap.end() ) {
TokenList tk;
- tk.push_back(iter->second.callId());
+ tk.push_back(id);
tk.push_back("hangup");
_requestManager.sendResponse(ResponseMessage("002", _getEventsSequenceId,tk));
// remove this call...
- _callMap.erase(id);
+ removeSubCall(id);
}
}
@@ -385,36 +300,23 @@ GUIServerImpl::displayConfigError (const std::string& error)
}
void
-GUIServerImpl::displayTextMessage (CALLID id, const std::string& message)
+GUIServerImpl::displayTextMessage (const CallID& id, const std::string& message)
{
- try {
- std::string callId = getCallIdFromId(id);
- TokenList tk;
- tk.push_back(callId);
- tk.push_back(message);
- tk.push_back("Text message");
- _requestManager.sendResponse(ResponseMessage("102", _getEventsSequenceId, tk));
- } catch(...) {
- TokenList tk;
- tk.push_back(message);
- tk.push_back("Text message");
- _requestManager.sendResponse(ResponseMessage("103", _getEventsSequenceId, tk));
- }
+ TokenList tk;
+ tk.push_back(id);
+ tk.push_back(message);
+ tk.push_back("Text message");
+ _requestManager.sendResponse(ResponseMessage("102", _getEventsSequenceId, tk));
}
void
-GUIServerImpl::displayErrorText (CALLID id, const std::string& message)
+GUIServerImpl::displayErrorText (const CallID& id, const std::string& message)
{
- try {
- std::string callId = getCallIdFromId(id);
- TokenList tk;
- tk.push_back(callId);
- tk.push_back(message);
- tk.push_back("Error");
- _requestManager.sendResponse(ResponseMessage("104", _getEventsSequenceId, tk));
- } catch(...) {
- displayError(message);
- }
+ TokenList tk;
+ tk.push_back(id);
+ tk.push_back(message);
+ tk.push_back("Error");
+ _requestManager.sendResponse(ResponseMessage("104", _getEventsSequenceId, tk));
}
void
@@ -427,18 +329,25 @@ GUIServerImpl::displayError (const std::string& error)
}
void
-GUIServerImpl::sendVoiceNbMessage(const std::string& nb_msg)
+GUIServerImpl::sendVoiceNbMessage(const AccountID& accountId, const std::string& nb_msg)
{
- _requestManager.sendResponse(ResponseMessage("020", _getEventsSequenceId, nb_msg));
+ TokenList tk;
+ tk.push_back(accountId);
+ tk.push_back(nb_msg);
+ _requestManager.sendResponse(ResponseMessage("020", _getEventsSequenceId, tk));
}
void
-GUIServerImpl::sendRegistrationState(bool state)
+GUIServerImpl::sendRegistrationState(const AccountID& accountid, bool state)
{
+ TokenList tk;
+ tk.push_back(accountid);
if (state == true) {
- _requestManager.sendResponse(ResponseMessage("003", _getEventsSequenceId, _("Registration succeed")));
+ tk.push_back(_("Registration succeed"));
+ _requestManager.sendResponse(ResponseMessage("003", _getEventsSequenceId, tk));
} else {
- _requestManager.sendResponse(ResponseMessage("004", _getEventsSequenceId, _("Registration failed")));
+ tk.push_back(_("Registration failed"));
+ _requestManager.sendResponse(ResponseMessage("004", _getEventsSequenceId, tk));
}
}
void
@@ -455,12 +364,11 @@ GUIServerImpl::sendMessage(const std::string& code, const std::string& seqId, To
void
GUIServerImpl::sendCallMessage(const std::string& code,
const std::string& sequenceId,
- CALLID id,
+ const CallID& id,
TokenList arg)
{
try {
- std::string callid = getCallIdFromId(id);
- arg.push_front(callid);
+ arg.push_front(id);
_requestManager.sendResponse(ResponseMessage(code, sequenceId, arg));
} catch(...) {
// no callid found
@@ -474,14 +382,15 @@ GUIServerImpl::update()
}
void
-GUIServerImpl::callFailure(CALLID id)
+GUIServerImpl::callFailure(const CallID& id)
{
CallMap::iterator iter = _callMap.find(id);
if ( iter != _callMap.end() ) {
TokenList tk;
- tk.push_back(iter->second.callId());
+ tk.push_back(id);
tk.push_back("Wrong number");
- _requestManager.sendResponse(ResponseMessage("504", iter->second.sequenceId(), tk));
+ _requestManager.sendResponse(ResponseMessage("504", iter->second, tk));
+ removeSubCall(id);
}
}
diff --git a/src/gui/server/guiserverimpl.h b/src/gui/server/guiserverimpl.h
index f237a3f3d0b19ccb9728796f4497795acad7077b..8b2a8ab8dae8cefd2b64d6f643ec4fafa2963198 100644
--- a/src/gui/server/guiserverimpl.h
+++ b/src/gui/server/guiserverimpl.h
@@ -23,14 +23,13 @@
#include <string>
#include <map>
-#include "subcall.h"
#include "requestmanager.h"
/** Session port for the daemon, default is DEFAULT_SESSION_PORT */
#define DEFAULT_SESSION_PORT 3999
-typedef std::map<CALLID, SubCall> CallMap;
+typedef std::map<CallID, std::string> CallMap;
class GUIServerImpl : public GuiFramework {
public:
@@ -42,26 +41,26 @@ public:
// exec loop
int exec(void);
- int incomingCall(CALLID id, const std::string& accountId, const std::string& from);
- void incomingMessage(const std::string& message);
+ bool incomingCall(const AccountID& accountId, const CallID& id, const std::string& from);
+ void incomingMessage(const AccountID& accountId, const std::string& message);
- void peerAnsweredCall (CALLID id);
- void peerRingingCall (CALLID id);
- void peerHungupCall (CALLID id);
+ void peerAnsweredCall (const CallID& id);
+ void peerRingingCall (const CallID& id);
+ void peerHungupCall (const CallID& id);
void displayStatus (const std::string& status);
void displayConfigError(const std::string& error);
- void displayTextMessage (CALLID id, const std::string& message);
- void displayErrorText (CALLID id, const std::string& message);
+ void displayTextMessage (const CallID& id, const std::string& message);
+ void displayErrorText (const CallID& id, const std::string& message);
void displayError (const std::string& error);
- void sendVoiceNbMessage(const std::string& nb_msg);
- void sendRegistrationState(bool state);
+ void sendVoiceNbMessage(const AccountID& accountid, const std::string& nb_msg);
+ void sendRegistrationState(const AccountID& accountid, bool state);
void setup();
void sendMessage(const std::string& code, const std::string& seqId,
TokenList& arg);
void sendCallMessage(const std::string& code, const std::string& sequenceId,
- CALLID id, TokenList arg);
- void callFailure(CALLID id);
+ const CallID& id, TokenList arg);
+ void callFailure(const CallID& id);
bool getEvents(const std::string& sequenceId);
bool sendGetEventsEnd();
@@ -90,17 +89,13 @@ public:
};
private:
- void insertSubCall(CALLID id, SubCall& subCall);
- void removeSubCall(CALLID id);
- std::string getSequenceIdFromId(CALLID id);
- std::string getCallIdFromId(CALLID id);
- CALLID getIdFromCallId(const std::string& callId);
-
+ void insertSubCall(const CallID& id, const std::string& seq);
+ void removeSubCall(const CallID& id);
+ std::string getSequenceIdFromId(const CallID& id);
/**
* This callMap is necessary because
- * ManagerImpl use callid-int
- * and the client use a callid-string
+ * because we want to retreive the seq associate to a call id
* and also a sequence number
*/
CallMap _callMap;
diff --git a/src/gui/server/request.cpp b/src/gui/server/request.cpp
index c4d84edfc6790a005edbeb14701c106275092475..124ec473a79a0f8e669e914b606dfa767ec58192 100644
--- a/src/gui/server/request.cpp
+++ b/src/gui/server/request.cpp
@@ -32,19 +32,19 @@ RequestCall::execute()
}
ResponseMessage
-RequestTextMessage::execute()
+RequestAnswer::execute()
{
- if ( GUIServer::instance().sendTextMessage(_account, _destination, _message) ) {
- return message("200", "Sending message");
+ if ( GUIServer::instance().answerCall(_callId) ) {
+ return message("200", _("OK"));
}
return message("500","Server Error");
}
ResponseMessage
-RequestAnswer::execute()
+RequestTextMessage::execute()
{
- if ( GUIServer::instance().answerCall(_callId) ) {
- return message("200", _("OK"));
+ if ( GUIServer::instance().sendTextMessage(_account, _destination, _message) ) {
+ return message("200", "Sending message");
}
return message("500","Server Error");
}
diff --git a/src/gui/server/requestconfig.cpp b/src/gui/server/requestconfig.cpp
index 5ce1f5431a01ae13cef5a9254af925dcb63e8d5d..134a26ab7b89b1688eaf4344493e71072eb96655 100644
--- a/src/gui/server/requestconfig.cpp
+++ b/src/gui/server/requestconfig.cpp
@@ -242,20 +242,42 @@ RequestVolumeMic::execute()
}
}
+RequestRegister::RequestRegister(const std::string &sequenceId, const TokenList& argList) : RequestGlobal(sequenceId,argList)
+{
+ TokenList::iterator iter = _argList.begin();
+ if (iter != _argList.end()) {
+ _accountId = *iter;
+ _argList.pop_front();
+ } else {
+ throw RequestConstructorException();
+ }
+}
+
ResponseMessage
RequestRegister::execute()
{
- if (GUIServer::instance().registerVoIPLink()) {
+ if (GUIServer::instance().registerVoIPLink(_accountId)) {
return message("200", _("OK"));
} else {
return message("500",_("Registration sending failed"));
}
}
+RequestUnregister::RequestUnregister(const std::string &sequenceId, const TokenList& argList) : RequestGlobal(sequenceId,argList)
+{
+ TokenList::iterator iter = _argList.begin();
+ if (iter != _argList.end()) {
+ _accountId = *iter;
+ _argList.pop_front();
+ } else {
+ throw RequestConstructorException();
+ }
+}
+
ResponseMessage
RequestUnregister::execute()
{
- if (GUIServer::instance().unregisterVoIPLink()) {
+ if (GUIServer::instance().unregisterVoIPLink(_accountId)) {
return message("200", _("OK"));
} else {
return message("500",_("Unregistration sending failed"));
diff --git a/src/gui/server/requestconfig.h b/src/gui/server/requestconfig.h
index c34cbb8202ddd77a08526c6d2550134067bd5a69..2edd772625abd0f487f040079bce5929d5f496a3 100644
--- a/src/gui/server/requestconfig.h
+++ b/src/gui/server/requestconfig.h
@@ -112,14 +112,18 @@ private:
class RequestRegister : public RequestGlobal {
public:
- RequestRegister(const std::string &sequenceId, const TokenList& argList) : RequestGlobal(sequenceId, argList) {}
+ RequestRegister(const std::string &sequenceId, const TokenList& argList);
ResponseMessage execute();
+private:
+ std::string _accountId;
};
class RequestUnregister : public RequestGlobal {
public:
- RequestUnregister(const std::string &sequenceId, const TokenList& argList) : RequestGlobal(sequenceId, argList) {}
+ RequestUnregister(const std::string &sequenceId, const TokenList& argList);
ResponseMessage execute();
+private:
+ std::string _accountId;
};
class RequestSwitch : public RequestGlobal {
public:
diff --git a/src/aixaccount.h b/src/iaxvoiplink.cpp
similarity index 64%
rename from src/aixaccount.h
rename to src/iaxvoiplink.cpp
index d5d54e289b792129da6d64f4d96ae9c7fa2f86ec..7b0729d5acc1e8b3cd268db4b418ed63e4bb1a42 100644
--- a/src/aixaccount.h
+++ b/src/iaxvoiplink.cpp
@@ -16,32 +16,16 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#ifndef AIXACCOUNT_H
-#define AIXACCOUNT_H
+#include "iaxvoiplink.h"
-#include "account.h"
-
-/**
- @author Yan Morin <yan.morin@gmail.com>
- An AIX Account specify AIX specific functions and objects (AIXCall/AIXVoIPLink)
-*/
-class AIXAccount : public Account
+IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
+ : VoIPLink(accountID)
{
-public:
- AIXAccount(const AccountID& accountID);
+}
- ~AIXAccount();
- /* virtual Account function implementation */
- void initConfig(Conf::ConfigTree& config);
- bool registerAccount();
- bool unregisterAccount();
- bool init();
- bool terminate();
+IAXVoIPLink::~IAXVoIPLink()
+{
+}
-private:
- /* virtual Account function implementation */
- bool createVoIPLink();
-};
-#endif
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
new file mode 100644
index 0000000000000000000000000000000000000000..fddd03a88a0e20f64825972220263a55c406b073
--- /dev/null
+++ b/src/iaxvoiplink.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2006 Savoir-Faire Linux inc.
+ * Author: Yan Morin <yan.morin@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 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+#ifndef IAXVOIPLINK_H
+#define IAXVOIPLINK_H
+
+#include "voIPLink.h"
+
+class AudioCodec;
+
+/**
+ @author Yan Morin <yan.morin@gmail.com>
+ VoIPLink contains a thread that listen to external events and
+ contains IAX Call related functions
+*/
+class IAXVoIPLink : public VoIPLink
+{
+public:
+ IAXVoIPLink(const AccountID& accountID);
+
+ ~IAXVoIPLink();
+
+ void getEvent (void) { }
+ bool init (void) { return false; }
+ bool checkNetwork (void) { return false; }
+ void terminate (void) { }
+
+ bool setRegister (void) { return false; }
+ bool setUnregister (void) { return false; }
+
+ Call* newOutgoingCall(const CallID& id, const std::string& toUrl) {return 0; }
+ bool answer(const CallID& id) {return false;}
+
+ bool hangup(const CallID& id) { return false; }
+ bool cancel(const CallID& id) { return false; }
+ bool onhold(const CallID& id) { return false; }
+ bool offhold(const CallID& id) { return false; }
+ bool transfer(const CallID& id, const std::string& to) { return false; }
+ bool refuse (const CallID& id) { return false; }
+ bool carryingDTMFdigits(const CallID& id, char code) { return false; }
+ bool sendMessage(const std::string& to, const std::string& body) { return false; }
+};
+
+#endif
diff --git a/src/main.cpp b/src/main.cpp
index fafad62cf8d4fc47ffadc77df73882c8e4abbe7c..bb74946f9c2904fc30614a687fb2f3fdfaf4340e 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -22,7 +22,7 @@
#include <cstring>
#include <iostream>
-#include "config.h"
+//#include "config.h"
#include "global.h"
#include "user_cfg.h"
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 6cb14fb00a6aaace0429caa100c86b091246c62c..3085ce1c8de60a116c303da573c9b907b3a71091 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -39,7 +39,6 @@
#include "accountcreator.h" // create new account
#include "voIPLink.h"
-#include "call.h"
#include "user_cfg.h"
#include "gui/guiframework.h"
@@ -54,8 +53,6 @@
#define fill_config_int(name, value) \
(_config.addConfigTreeItem(section, Conf::ConfigTreeItem(std::string(name), std::string(value), type_int)))
-#define DFT_VOIP_LINK 0
-
ManagerImpl::ManagerImpl (void)
{
// Init private variables
@@ -73,19 +70,18 @@ ManagerImpl::ManagerImpl (void)
_setupLoaded = false;
_gui = NULL;
+ // sound
_audiodriverPA = NULL;
_dtmfKey = 0;
-
- // Initialize after by init() -> initVolume()
- _spkr_volume = 0;
- _mic_volume = 0;
- _mic_volume_before_mute = 0;
+ _spkr_volume = 0; // Initialize after by init() -> initVolume()
+ _mic_volume = 0; // Initialize after by init() -> initVolume()
+ _mic_volume_before_mute = 0;
// Call
- _currentCallId = 0;
_nbIncomingWaitingCall=0;
_registerState = UNREGISTERED;
_hasTriedToRegister = false;
+
// initialize random generator for call id
srand (time(NULL));
@@ -156,8 +152,6 @@ ManagerImpl::init()
_dtmfKey = new DTMF(sampleRate, outChannel);
}
- _debugInit("Adding new VoIP Link");
-
// initRegisterVoIP 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();
@@ -169,16 +163,6 @@ void ManagerImpl::terminate()
unloadAccountMap();
- _debug("Removing calls\n");
- _mutex.enterMutex();
- for(CallVector::iterator pos = _callVector.begin();
- pos != _callVector.end();
- pos++) {
- delete *pos; *pos = NULL;
- }
- _callVector.clear();
- _mutex.leaveMutex();
-
_debug("Unload DTMF Key\n");
delete _dtmfKey;
@@ -189,230 +173,165 @@ void ManagerImpl::terminate()
delete _telephoneTone; _telephoneTone = 0;
}
-void
-ManagerImpl::setGui(GuiFramework* gui)
-{
- _gui = gui;
-}
-
-/**
- * Multi Thread with _mutex for callVector
- */
-Call *
-ManagerImpl::pushBackNewCall(CALLID id, Call::CallType type)
-{
- ost::MutexLock m(_mutex);
-
- Call* call = new Call(id, type, getAccountLink(ACCOUNT_SIP0));
- // Set the wanted voip-link (first of the list)
- _callVector.push_back(call);
- return call;
-}
-
-/**
- * Multi Thread with _mutex for callVector
- */
-Call*
-ManagerImpl::getCall(CALLID id)
-{
- //_debug("%10d: Getting call\n", id);
- Call* call = NULL;
- unsigned int size = _callVector.size();
- for (unsigned int i = 0; i < size; i++) {
- call = _callVector.at(i);
- if (call && call->getId() == id) {
- break;
- } else {
- call = NULL;
- }
- }
- return call;
+bool
+ManagerImpl::isCurrentCall(const CallID& callId) {
+ ost::MutexLock m(_currentCallMutex);
+ return (_currentCallId2 == callId ? true : false);
}
-/**
- * Multi Thread with _mutex for callVector
- */
-void
-ManagerImpl::deleteCall (CALLID id)
-{
- //_debug("%10d: Deleting call\n", id);
- CallVector::iterator iter = _callVector.begin();
- while(iter!=_callVector.end()) {
- Call *call = *iter;
- if (call != NULL && call->getId() == id) {
- if (call->getFlagNotAnswered() && call->isIncomingType() && call->getState() != Call::NotExist) {
- decWaitingCall();
- }
- delete (*iter); *iter = NULL;
- call = NULL;
- _callVector.erase(iter);
- return;
- }
- iter++;
+bool
+ManagerImpl::hasCurrentCall() {
+ ost::MutexLock m(_currentCallMutex);
+ if ( _currentCallId2 != "") {
+ return true;
}
+ return false;
}
-void
-ManagerImpl::setCurrentCallId(CALLID id)
-{
- //_debug("%10d: Setting current callid, old one was: %d\n", id, _currentCallId);
- _currentCallId = id;
+const CallID&
+ManagerImpl::getCurrentCallId() {
+ ost::MutexLock m(_currentCallMutex);
+ return _currentCallId2;
}
void
-ManagerImpl::removeCallFromCurrent(CALLID id)
-{
- if ( _currentCallId == id ) {
- //_debug("%10d: Setting current callid, old one was: %d\n", 0, _currentCallId);
- _currentCallId = 0;
- }
+ManagerImpl::switchCall(const CallID& id ) {
+ ost::MutexLock m(_currentCallMutex);
+ _currentCallId2 = id;
}
///////////////////////////////////////////////////////////////////////////////
// Management of events' IP-phone user
///////////////////////////////////////////////////////////////////////////////
-/**
- * Main thread
- */
-int
-ManagerImpl::outgoingCall (const std::string& to)
-{
- CALLID id = generateNewCallId();
- _debug("%10d: Outgoing Call\n", id);
- Call *call = pushBackNewCall(id, Call::Outgoing);
- ost::MutexLock m(_mutex);
- call->setState(Call::Progressing);
- call->setCallerIdNumber(to);
- if (call->outgoingCall(to) == 0) {
- return id;
+/* Main Thread */
+bool
+ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const std::string& to)
+{
+ if (!accountExists(accountid)) {
+ _debug("Outgoing Call: account doesn't exist\n");
+ return false;
+ }
+ if (getAccountFromCall(id) != AccountNULL) {
+ _debug("Outgoing Call: call id already exists\n");
+ return false;
+ }
+ _debug("Adding Outgoing Call %s on account %s\n", id.data(), accountid.data());
+ if ( getAccountLink(accountid)->newOutgoingCall(id, to) ) {
+ associateCallToAccount( id, accountid );
+ switchCall(id);
+ return true;
} else {
- return 0;
+ _debug("An error occur, the call was not created\n");
}
+ return false;
}
-/**
- * Main thread
- */
-bool
-ManagerImpl::sendTextMessage(const std::string&, const std::string& to, const std::string& message)
+//THREAD=Main : for outgoing Call
+bool
+ManagerImpl::answerCall(const CallID& id)
{
- return getAccountLink(ACCOUNT_SIP0)->sendMessage(to, message);
-}
+ stopTone();
-/**
- * User action (main thread)
- * Every Call
- */
-int
-ManagerImpl::hangupCall (CALLID id)
-{
- _debug("%10d: Hangup Call\n", id);
- stopTone();
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call == NULL) {
- return -1;
- }
- int result = -1;
- if (call->getState() != Call::Error) {
- result = call->hangup();
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("Answering Call: Call doesn't exists\n");
+ return false;
}
- deleteCall(id);
- // current call id or no line selected
- if (id == _currentCallId || _currentCallId == 0) {
- removeCallFromCurrent(id);
+
+ if (!getAccountLink(accountid)->answer(id)) {
+ // error when receiving...
+ removeCallAccount(id);
+ return false;
}
- return result;
+
+ // if it was waiting, it's waiting no more
+ removeWaitingCall(id);
+ switchCall(id);
+ return true;
}
-/**
- * User action (main thread)
- * Every Call
- * -1 : call not found
- * 0 : already in this state...
- */
-int
-ManagerImpl::cancelCall (CALLID id)
+//THREAD=Main
+bool
+ManagerImpl::sendTextMessage(const AccountID& accountId, const std::string& to, const std::string& message)
{
- _debug("%10d: Cancel Call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call == NULL) {
- return -1;
+ if (accountExists(accountId)) {
+ return getAccountLink(accountId)->sendMessage(to, message);
}
- int result = call->cancel();
- deleteCall(id);
- stopTone();
- return result;
+ return false;
}
-/**
- * User action (main thread)
- * Incoming Call
- */
-int
-ManagerImpl::answerCall (CALLID id)
+//THREAD=Main
+bool
+ManagerImpl::hangupCall(const CallID& id)
{
- _debug("%10d: Answer Call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call == NULL) {
- return -1;
- }
- if (call->getFlagNotAnswered()) {
- decWaitingCall();
- call->setFlagNotAnswered(false);
- }
- if (call->getState() != Call::OnHold) {
- switchCall(id);
+ stopTone();
+
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("Hangup Call: Call doesn't exists\n");
+ return false;
}
- stopTone(); // before answer, don't stop the audio stream after open it
- return call->answer();
+
+ bool returnValue = getAccountLink(accountid)->hangup(id);
+ removeCallAccount(id);
+ switchCall("");
+
+ return returnValue;
}
-/**
- * User action (main thread)
- * Every Call
- * @return 0 if it fails, -1 if not present
- */
-int
-ManagerImpl::onHoldCall (CALLID id)
+//THREAD=Main
+bool
+ManagerImpl::cancelCall (const CallID& id)
{
- _debug("%10d: On Hold Call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call == NULL) {
- return -1;
- }
- removeCallFromCurrent(id);
- if ( call->getState() == Call::OnHold || call->isNotAnswered()) {
- return 1;
+ stopTone();
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("Cancel Call: Call doesn't exists\n");
+ return false;
}
- return call->onHold();
+
+ bool returnValue = getAccountLink(accountid)->cancel(id);
+ // it could be a waiting call?
+ removeWaitingCall(id);
+ removeCallAccount(id);
+ switchCall("");
+
+ return returnValue;
}
-/**
- * User action (main thread)
- * Every Call
- */
-int
-ManagerImpl::offHoldCall (CALLID id)
+//THREAD=Main
+bool
+ManagerImpl::onHoldCall(const CallID& id)
{
- _debug("%10d: Off Hold Call\n", id);
- ost::MutexLock m(_mutex);
stopTone();
- Call* call = getCall(id);
- if (call == 0) {
- return -1;
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("On Hold Call: Call doesn't exists\n");
+ return false;
}
- if (call->getState() == Call::OffHold) {
- return 1;
+
+ bool returnValue = getAccountLink(accountid)->onhold(id);
+ removeWaitingCall(id);
+ switchCall("");
+
+ return returnValue;
+}
+
+//THREAD=Main
+bool
+ManagerImpl::offHoldCall(const CallID& id)
+{
+ stopTone();
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("OffHold Call: Call doesn't exists\n");
+ return false;
}
- setCurrentCallId(id);
- int returnValue = call->offHold();
- // start audio if it's ok
- if (returnValue != -1) {
+ bool returnValue = getAccountLink(accountid)->offhold(id);
+ switchCall(id);
+
+ if (returnValue) {
try {
getAudioDriver()->startStream();
} catch(...) {
@@ -422,37 +341,32 @@ ManagerImpl::offHoldCall (CALLID id)
return returnValue;
}
-/**
- * User action (main thread)
- * Every Call
- */
-int
-ManagerImpl::transferCall (CALLID id, const std::string& to)
+//THREAD=Main
+bool
+ManagerImpl::transferCall(const CallID& id, const std::string& to)
{
- _debug("%10d: Transfer Call to %s\n", id, to.c_str());
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call == 0) {
- return -1;
+ stopTone();
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("Transfer Call: Call doesn't exists\n");
+ return false;
}
- removeCallFromCurrent(id);
- return call->transfer(to);
+ bool returnValue = getAccountLink(accountid)->transfer(id, to);
+ removeWaitingCall(id);
+ removeCallAccount(id);
+ switchCall("");
+
+ return returnValue;
}
-/**
- * User action (main thread)
- * All Call
- */
+//THREAD=Main
void
ManagerImpl::mute() {
_mic_volume_before_mute = _mic_volume;
setMicVolume(0);
}
-/**
- * User action (main thread)
- * All Call
- */
+//THREAD=Main
void
ManagerImpl::unmute() {
if ( _mic_volume == 0 ) {
@@ -460,34 +374,24 @@ ManagerImpl::unmute() {
}
}
-/**
- * User action (main thread)
- * Call Incoming
- */
-int
-ManagerImpl::refuseCall (CALLID id)
+//THREAD=Main : Call:Incoming
+bool
+ManagerImpl::refuseCall (const CallID& id)
{
- _debug("%10d: Refuse Call\n", id);
- ost::MutexLock m(_mutex);
- Call *call = getCall(id);
- if (call == NULL) {
- return -1;
- }
-
- if ( call->getState() != Call::Progressing ) {
- return -1;
- }
-
- int refuse = call->refuse();
- removeCallFromCurrent(id);
- deleteCall(id);
stopTone();
- return refuse;
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("OffHold Call: Call doesn't exists\n");
+ return false;
+ }
+ bool returnValue = getAccountLink(accountid)->refuse(id);
+ removeWaitingCall(id);
+ removeCallAccount(id);
+ switchCall("");
+ return returnValue;
}
-/**
- * User action (main thread)
- */
+//THREAD=Main
bool
ManagerImpl::saveConfig (void)
{
@@ -499,73 +403,67 @@ ManagerImpl::saveConfig (void)
return _setupLoaded;
}
-/**
- * Main Thread
- */
+//THREAD=Main
bool
ManagerImpl::initRegisterVoIPLink()
{
- int returnValue = true;
- _debugInit("Initiate VoIP Link Registration\n");
- if (_hasTriedToRegister == false) {
- if ( getAccount(ACCOUNT_SIP0)->init() ) {
- // we call here, because it's long...
- // If network is available and exosip is start..
- if (getConfigInt(SIGNALISATION, AUTO_REGISTER) && _exist == 1) {
- registerVoIPLink();
- _hasTriedToRegister = true;
+ _debugInit("Initiate VoIP Links Registration\n");
+ AccountMap::iterator iter = _accountMap.begin();
+ while( iter != _accountMap.end() ) {
+ if ( iter->second) {
+ iter->second->loadConfig();
+ if ( iter->second->shouldInitOnStart() ) {
+ iter->second->init();
+ if (iter->second->shouldRegisterOnStart()) {
+ iter->second->registerAccount();
+ }
}
- } else {
- returnValue = false;
}
+ iter++;
}
- return returnValue;
+ return true;
}
-
-/**
- * Initialize action (main thread)
- * Note that Registration is only send if STUN is not activated
- * @return true if setRegister is call without failure, else return false
- */
+//THREAD=Main
bool
-ManagerImpl::registerVoIPLink (void)
+ManagerImpl::registerVoIPLink(const AccountID& accountId)
{
_debug("Register VoIP Link\n");
int returnValue = false;
- returnValue = getAccount(ACCOUNT_SIP0)->registerAccount();
- if (returnValue) {
- _registerState = REGISTERED;
- } else {
- _registerState = FAILED;
+ if (accountExists( accountId ) ) {
+ returnValue = getAccount(accountId)->registerAccount();
}
return returnValue;
}
-/**
- * Terminate action (main thread)
- * @return true if the unregister method is send correctly
- */
+//THREAD=Main
bool
-ManagerImpl::unregisterVoIPLink (void)
+ManagerImpl::unregisterVoIPLink(const AccountID& accountId)
{
_debug("Unregister VoIP Link\n");
- return getAccount(ACCOUNT_SIP0)->unregisterAccount();
+ int returnValue = false;
+ if (accountExists( accountId ) ) {
+ returnValue = getAccount(accountId)->registerAccount();
+ }
+ return returnValue;
}
-/**
- * User action (main thread)
- */
+//THREAD=Main
bool
-ManagerImpl::sendDtmf (CALLID id, char code)
+ManagerImpl::sendDtmf(const CallID& id, char code)
{
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("Send DTMF: call doesn't exists\n");
+ return false;
+ }
+
int sendType = getConfigInt(SIGNALISATION, SEND_DTMF_AS);
- int returnValue = false;
+ bool returnValue = false;
switch (sendType) {
case 0: // SIP INFO
playDtmf(code);
- getAccountLink(ACCOUNT_SIP0)->carryingDTMFdigits(id, code);
- returnValue = true;
+ returnValue = getAccountLink(accountid)->carryingDTMFdigits(id, code);
break;
case 1: // Audio way
@@ -578,10 +476,7 @@ ManagerImpl::sendDtmf (CALLID id, char code)
return returnValue;
}
-/**
- * User action (main thread)
- * Or sip event (dtmf body submit)
- */
+//THREAD=Main | VoIPLink
bool
ManagerImpl::playDtmf(char code)
{
@@ -595,7 +490,6 @@ ManagerImpl::playDtmf(char code)
// numbers of int = length in milliseconds / 1000 (number of seconds)
// = number of seconds * SAMPLING_RATE by SECONDS
-
AudioLayer* audiolayer = getAudioDriver();
// fast return, no sound, so no dtmf
@@ -634,264 +528,158 @@ ManagerImpl::playDtmf(char code)
return returnValue;
}
-///////////////////////////////////////////////////////////////////////////////
-// Management of event peer IP-phone
-////////////////////////////////////////////////////////////////////////////////
-/**
- * Multi-thread
- */
+
+
+// Multi-thread
bool
ManagerImpl::incomingCallWaiting() {
- ost::MutexLock m(_incomingCallMutex);
+ ost::MutexLock m(_waitingCallMutex);
return (_nbIncomingWaitingCall > 0) ? true : false;
}
void
-ManagerImpl::incWaitingCall() {
- ost::MutexLock m(_incomingCallMutex);
+ManagerImpl::addWaitingCall(const CallID& id) {
+ ost::MutexLock m(_waitingCallMutex);
+ _waitingCall.insert(id);
_nbIncomingWaitingCall++;
- //_debug("incWaitingCall: %d\n", _nbIncomingWaitingCall);
-}
-
-void
-ManagerImpl::decWaitingCall() {
- ost::MutexLock m(_incomingCallMutex);
- _nbIncomingWaitingCall--;
- //_debug("decWaitingCall: %d\n", _nbIncomingWaitingCall);
}
-
-/**
- * SipEvent Thread
- * Set the call info for incoming call
- */
void
-ManagerImpl::callSetInfo(CALLID id, const std::string& name, const std::string& number)
-{
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != 0) {
- call->setCallerIdName(name);
- call->setCallerIdNumber(number);
+ManagerImpl::removeWaitingCall(const CallID& id) {
+ ost::MutexLock m(_waitingCallMutex);
+ // should return more than 1 if it erase a call
+ if (_waitingCall.erase(id)) {
+ _nbIncomingWaitingCall--;
}
}
-/**
- * SipEvent Thread
- * ask if it can close the call
- */
bool
-ManagerImpl::callCanBeClosed(CALLID id) {
- bool returnValue = false;
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != NULL && call->getState() != Call::Progressing) {
- returnValue = true;
+ManagerImpl::isWaitingCall(const CallID& id) {
+ ost::MutexLock m(_waitingCallMutex);
+ CallIDSet::iterator iter = _waitingCall.find(id);
+ if (iter != _waitingCall.end()) {
+ return false;
}
- return returnValue;
+ return true;
}
-/**
- * SipEvent Thread
- * ask if it can answer the call
- */
-bool
-ManagerImpl::callCanBeAnswered(CALLID id) {
- bool returnValue = false;
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != NULL && ( call->getFlagNotAnswered() ||
- (call->getState()!=Call::OnHold && call->getState()!=Call::OffHold) )) {
- returnValue = true;
- }
- return returnValue;
-}
-/**
- * SipEvent Thread
- * ask if it can start the sound thread
- */
-bool
-ManagerImpl::callIsOnHold(CALLID id) {
- bool returnValue = false;
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != NULL && (call->getState()==Call::OnHold)) {
- returnValue = true;
- }
- return returnValue;
-}
-/**
- * SipEvent Thread
- */
-int
-ManagerImpl::incomingCall (CALLID id, const std::string& name, const std::string& number)
+///////////////////////////////////////////////////////////////////////////////
+// Management of event peer IP-phone
+////////////////////////////////////////////////////////////////////////////////
+// SipEvent Thread
+bool
+ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
{
- _debug("%10d: Incoming call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call == NULL) {
- return -1;
- }
- call->setType(Call::Incoming);
- call->setState(Call::Progressing);
-
- if ( _currentCallId == 0 ) {
- call->setFlagNotAnswered(false);
+ _debug("Incoming call\n");
+ associateCallToAccount(call->getCallId(), accountId);
+ if ( !hasCurrentCall() ) {
+ call->setConnectionState(Call::Ringing);
ringtone();
- switchCall(id);
+ switchCall(call->getCallId());
} else {
- incWaitingCall();
+ addWaitingCall(call->getCallId());
}
- // TODO: Account not yet implemented
- std::string accountId = "acc1";
- std::string from = name; call->setCallerIdName(name);
- call->setCallerIdNumber(number);
+ std::string from = call->getPeerName();
+ std::string number = call->getPeerNumber();
+
if ( !number.empty() ) {
from.append(" <");
from.append(number);
from.append(">");
}
- return _gui->incomingCall(id, accountId, from);
+ _gui->incomingCall(accountId, call->getCallId(), from);
+
+ return true;
}
-/**
- * SipEvent Thread
- * for outgoing message, send by SipEvent
- */
-void
-ManagerImpl::incomingMessage(const std::string& message) {
+//THREAD=VoIP
+void
+ManagerImpl::incomingMessage(const AccountID& accountId, const std::string& message) {
if (_gui) {
- _gui->incomingMessage(message);
+ _gui->incomingMessage(accountId, message);
}
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
-void
-ManagerImpl::peerAnsweredCall (CALLID id)
+//THREAD=VoIP CALL=Outgoing
+void
+ManagerImpl::peerAnsweredCall(const CallID& id)
{
- _debug("%10d: Peer Answered Call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != 0) {
- call->setFlagNotAnswered(false);
- call->setState(Call::Answered);
-
+ if (isCurrentCall(id)) {
stopTone();
- // switch current call
- switchCall(id);
- if (_gui) _gui->peerAnsweredCall(id);
}
+ if (_gui) _gui->peerAnsweredCall(id);
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
-int
-ManagerImpl::peerRingingCall (CALLID id)
+//THREAD=VoIP Call=Outgoing
+void
+ManagerImpl::peerRingingCall(const CallID& id)
{
- _debug("%10d: Peer Ringing Call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != 0) {
- call->setState(Call::Ringing);
-
- // ring
+ if (isCurrentCall(id)) {
ringback();
- if (_gui) _gui->peerRingingCall(id);
}
- return 1;
+ if (_gui) _gui->peerRingingCall(id);
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
-int
-ManagerImpl::peerHungupCall (CALLID id)
+//THREAD=VoIP Call=Outgoing/Ingoing
+void
+ManagerImpl::peerHungupCall(const CallID& id)
{
- _debug("%10d: Peer Hungup Call\n", id);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if ( call == NULL ) {
- return -1;
+ AccountID accountid = getAccountFromCall( id );
+ if (accountid == AccountNULL) {
+ _debug("peerHungupCall: Call doesn't exists\n");
+ return;
}
- if ( _currentCallId == id ) {
+ if (isCurrentCall(id)) {
stopTone();
+ switchCall("");
}
-
- if (_gui) _gui->peerHungupCall(id);
- deleteCall(id);
- call->setState(Call::Hungup);
-
- removeCallFromCurrent(id);
- return 1;
+ removeWaitingCall(id);
+ removeCallAccount(id);
}
-/**
- * Multi Thread
- */
+//THREAD=VoIP
void
-ManagerImpl::callBusy(CALLID id) {
- _debug("%10d: Call is busy\n", id);
- playATone(Tone::TONE_BUSY);
- ost::MutexLock m(_mutex);
- Call* call = getCall(id);
- if (call != 0) {
- call->setState(Call::Busy);
+ManagerImpl::callBusy(const CallID& id) {
+ _debug("Call busy\n");
+ if (isCurrentCall(id) ) {
+ playATone(Tone::TONE_BUSY);
+ switchCall("");
}
- deleteCall(id);
- call->setState(Call::Hungup);
-
- removeCallFromCurrent(id);
+ removeCallAccount(id);
+ removeWaitingCall(id);
}
-/**
- * Multi Thread
- */
+//THREAD=VoIP
void
-ManagerImpl::callFailure(CALLID id) {
- _debug("%10d: Call failed\n", id);
- playATone(Tone::TONE_BUSY);
- _mutex.enterMutex();
- Call* call = getCall(id);
- if (call != 0) {
- call->setState(Call::Error);
- }
- _mutex.leaveMutex();
+ManagerImpl::callFailure(const CallID& id)
+{
+ _debug("Call failed\n");
+ if (isCurrentCall(id) ) {
+ playATone(Tone::TONE_BUSY);
+ switchCall("");
+ }
if (_gui) {
_gui->callFailure(id);
}
- deleteCall(id);
- call->setState(Call::Hungup);
-
- removeCallFromCurrent(id);
+ removeCallAccount(id);
+ removeWaitingCall(id);
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
-ManagerImpl::displayTextMessage (CALLID id, const std::string& message)
+ManagerImpl::displayTextMessage(const CallID& id, const std::string& message)
{
if(_gui) {
- _gui->displayTextMessage(id, message);
+ _gui->displayTextMessage(id, message);
}
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
-ManagerImpl::displayErrorText (CALLID id, const std::string& message)
+ManagerImpl::displayErrorText(const CallID& id, const std::string& message)
{
if(_gui) {
_gui->displayErrorText(id, message);
@@ -900,10 +688,7 @@ ManagerImpl::displayErrorText (CALLID id, const std::string& message)
}
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
ManagerImpl::displayError (const std::string& error)
{
@@ -912,22 +697,16 @@ ManagerImpl::displayError (const std::string& error)
}
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
-ManagerImpl::displayStatus (const std::string& status)
+ManagerImpl::displayStatus(const std::string& status)
{
if(_gui) {
_gui->displayStatus(status);
}
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
ManagerImpl::displayConfigError (const std::string& message)
{
@@ -936,42 +715,32 @@ ManagerImpl::displayConfigError (const std::string& message)
}
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
-ManagerImpl::startVoiceMessageNotification (const std::string& nb_msg)
+ManagerImpl::startVoiceMessageNotification(const AccountID& accountId, const std::string& nb_msg)
{
- if (_gui) _gui->sendVoiceNbMessage(nb_msg);
+ if (_gui) _gui->sendVoiceNbMessage(accountId, nb_msg);
}
-/**
- * SipEvent Thread
- * for outgoing call, send by SipEvent
- */
+//THREAD=VoIP
void
-ManagerImpl::stopVoiceMessageNotification (void)
+ManagerImpl::stopVoiceMessageNotification(const AccountID& accountId)
{
- if (_gui) _gui->sendVoiceNbMessage(std::string("0"));
+ if (_gui) _gui->sendVoiceNbMessage(accountId, std::string("0"));
}
-/**
- * SipEvent Thread
- */
+//THREAD=VoIP
void
-ManagerImpl::registrationSucceed()
+ManagerImpl::registrationSucceed(const AccountID& accountid)
{
- if (_gui) _gui->sendRegistrationState(true);
+ if (_gui) _gui->sendRegistrationState(accountid, true);
}
-/**
- * SipEvent Thread
- */
+//THREAD=VoIP
void
-ManagerImpl::registrationFailed()
+ManagerImpl::registrationFailed(const AccountID& accountid)
{
- if (_gui) _gui->sendRegistrationState(false);
+ if (_gui) _gui->sendRegistrationState(accountid, false);
}
/**
@@ -1109,7 +878,7 @@ ManagerImpl::getTelephoneFile()
* By AudioRTP thread
*/
void
-ManagerImpl::notificationIncomingCall (void) {
+ManagerImpl::notificationIncomingCall(void) {
AudioLayer* audiolayer = getAudioDriver();
if (audiolayer != 0) {
@@ -1123,7 +892,6 @@ ManagerImpl::notificationIncomingCall (void) {
tone.getNext(buf, tone.getSize());
audiolayer->putUrgent(buf, sizeof(int16)*nbInt16);
}
-
}
/**
@@ -1158,37 +926,30 @@ ManagerImpl::getStunInfo (StunAddress4& stunSvrAddr, int port)
}
bool
-ManagerImpl::useStun (void)
-{
- if (getConfigInt(SIGNALISATION, USE_STUN)) {
- return true;
- } else {
- return false;
+ManagerImpl::behindNat(int port)
+{
+ StunAddress4 stunSvrAddr;
+ stunSvrAddr.addr = 0;
+
+ // Stun server
+ std::string svr = getConfigString(SIGNALISATION, STUN_SERVER);
+
+ // Convert char* to StunAddress4 structure
+ bool ret = stunParseServerName ((char*)svr.data(), stunSvrAddr);
+ if (!ret) {
+ _debug("SIP: Stun server address (%s) is not valid\n", svr.data());
+ return 0;
}
+
+ // Firewall address
+ //_debug("STUN server: %s\n", svr.data());
+ return getStunInfo(stunSvrAddr, port);
}
+
///////////////////////////////////////////////////////////////////////////////
// Private functions
///////////////////////////////////////////////////////////////////////////////
-
-/**
- * Multi Thread
- */
-CALLID
-ManagerImpl::generateNewCallId (void)
-{
- CALLID random_id = (unsigned)rand();
-
- // Check if already a call with this id exists
- _mutex.enterMutex();
- while (getCall(random_id) != NULL && random_id != 0) {
- random_id = rand();
- }
- _mutex.leaveMutex();
- // If random_id is not attributed, returns it.
- return random_id;
-}
-
/**
* Initialization: Main Thread
* @return 1: ok
@@ -1357,7 +1118,7 @@ ManagerImpl::initZeroconf(void)
void
ManagerImpl::initVolume()
{
- _debugInit("Initiate Volume\n");
+ _debugInit("Initiate Volume");
setSpkrVolume(getConfigInt(AUDIO, VOLUME_SPKR));
setMicVolume(getConfigInt(AUDIO, VOLUME_MICRO));
}
@@ -1453,58 +1214,68 @@ ManagerImpl::getEvents() {
return true;
}
+// TODO: rewrite this
/**
* Main Thread
*/
bool
ManagerImpl::getCallStatus(const std::string& sequenceId)
{
- // TODO: implement account
- std::string accountId = "acc1";
+ if (!_gui) { return false; }
+ ost::MutexLock m(_callAccountMapMutex);
+ CallAccountMap::iterator iter = _callAccountMap.begin();
+ TokenList tk;
std::string code;
std::string status;
- TokenList tk;
- Call* call;
-
- if (_gui!=NULL) {
- ost::MutexLock m(_mutex);
- CallVector::iterator iter = _callVector.begin();
- while(iter!=_callVector.end()){
- call = (*iter);
- switch( call->getState() ) {
- case Call::Progressing: code="110"; status="Trying"; break;
- case Call::Ringing: code="111"; status = "Ringing"; break;
- case Call::Answered: code="112"; status = "Established"; break;
- case Call::Busy: code="113"; status = "Busy"; break;
- case Call::OnHold: code="114"; status = "Held"; break;
- case Call::OffHold: code="115"; status = "Unheld"; break;
- default: code="125"; status="Other";
+ std::string destination;
+ std::string number;
+
+ while (iter != _callAccountMap.begin())
+ {
+ Call* call = getAccountLink(iter->second)->getCall(iter->first);
+ Call::ConnectionState state = call->getConnectionState();
+ if (state != Call::Connected) {
+ switch(state) {
+ case Call::Trying: code="110"; status = "Trying"; break;
+ case Call::Ringing: code="111"; status = "Ringing"; break;
+ case Call::Progressing: code="125"; status = "Progressing"; break;
+ case Call::Disconnected: code="125"; status = "Disconnected"; break;
+ default: code=""; status= "";
}
-
- // No Congestion
- // No Wrong Number
- // 116 <CSeq> <call-id> <acc> <destination> Busy
- std::string destination = call->getCallerIdName();
- std::string number = call->getCallerIdNumber();
- if (number!="") {
- destination.append(" <");
- destination.append(number);
- destination.append(">");
+ } else {
+ switch (call->getState()) {
+ case Call::Active: code="112"; status = "Established"; break;
+ case Call::Hold: code="114"; status = "Held"; break;
+ case Call::Busy: code="113"; status = "Busy"; break;
+ case Call::Refused: code="125"; status = "Refused"; break;
+ case Call::Error: code="125"; status = "Error"; break;
+ case Call::Inactive: code="125"; status = "Inactive"; break;
}
- tk.push_back(accountId);
- tk.push_back(destination);
- tk.push_back(status);
- _gui->sendCallMessage(code, sequenceId, (*iter)->getId(), tk);
- iter++;
- tk.clear();
}
+
+ // No Congestion
+ // No Wrong Number
+ // 116 <CSeq> <call-id> <acc> <destination> Busy
+ destination = call->getPeerName();
+ number = call->getPeerNumber();
+ if (number!="") {
+ destination.append(" <");
+ destination.append(number);
+ destination.append(">");
+ }
+ tk.push_back(iter->second);
+ tk.push_back(destination);
+ tk.push_back(status);
+ _gui->sendCallMessage(code, sequenceId, iter->first, tk);
+ tk.clear();
+
+ iter++;
}
+
return true;
}
-/**
- * Main Thread
- */
+//THREAD=Main
bool
ManagerImpl::getConfigAll(const std::string& sequenceId)
{
@@ -1521,18 +1292,14 @@ ManagerImpl::getConfigAll(const std::string& sequenceId)
return returnValue;
}
-/**
- * Main Thread
- */
+//THREAD=Main
bool
ManagerImpl::getConfig(const std::string& section, const std::string& name, TokenList& arg)
{
return _config.getConfigTreeItemToken(section, name, arg);
}
-/**
- * Main Thread
- */
+//THREAD=Main
// throw an Conf::ConfigTreeItemException if not found
int
ManagerImpl::getConfigInt(const std::string& section, const std::string& name)
@@ -1545,9 +1312,7 @@ ManagerImpl::getConfigInt(const std::string& section, const std::string& name)
return 0;
}
-/**
- * Main Thread
- */
+//THREAD=Main
std::string
ManagerImpl::getConfigString(const std::string& section, const std::string&
name)
@@ -1560,18 +1325,14 @@ name)
return "";
}
-/**
- * Main Thread
- */
+//THREAD=Main
bool
ManagerImpl::setConfig(const std::string& section, const std::string& name, const std::string& value)
{
return _config.setConfigTreeItem(section, name, value);
}
-/**
- * Main Thread
- */
+//THREAD=Main
bool
ManagerImpl::setConfig(const std::string& section, const std::string& name, int value)
{
@@ -1580,9 +1341,7 @@ ManagerImpl::setConfig(const std::string& section, const std::string& name, int
return _config.setConfigTreeItem(section, name, valueStream.str());
}
-/**
- * Main Thread
- */
+//THREAD=Main
bool
ManagerImpl::getConfigList(const std::string& sequenceId, const std::string& name)
{
@@ -1634,9 +1393,7 @@ ManagerImpl::getConfigList(const std::string& sequenceId, const std::string& nam
return returnValue;
}
-/**
- * User request Main Thread (list)
- */
+//THREAD=Main
bool
ManagerImpl::getAudioDeviceList(const std::string& sequenceId, int ioDeviceMask)
{
@@ -1672,6 +1429,7 @@ ManagerImpl::getAudioDeviceList(const std::string& sequenceId, int ioDeviceMask)
return returnValue;
}
+//THREAD=Main
bool
ManagerImpl::getCountryTones(const std::string& sequenceId)
{
@@ -1687,6 +1445,7 @@ ManagerImpl::getCountryTones(const std::string& sequenceId)
return true;
}
+//THREAD=Main
void
ManagerImpl::sendCountryTone(const std::string& sequenceId, int index, const std::string& name) {
TokenList tk;
@@ -1695,9 +1454,7 @@ ManagerImpl::sendCountryTone(const std::string& sequenceId, int index, const std
_gui->sendMessage("100", sequenceId, tk);
}
-/**
- * User action : main thread
- */
+//THREAD=Main
bool
ManagerImpl::getDirListing(const std::string& sequenceId, const std::string& path, int *nbFile) {
TokenList tk;
@@ -1726,21 +1483,8 @@ ManagerImpl::getDirListing(const std::string& sequenceId, const std::string& pat
}
}
-/**
- * Multi Thread
- */
-void
-ManagerImpl::switchCall(CALLID id)
-{
- // we can only switch the current call id if we
- // it's not selected yet..
- if (_currentCallId == 0 ) {
- setCurrentCallId(id);
- }
-}
-
-/**
- * Main Thread - Gui
+//THREAD=Main
+/*
* Experimental...
*/
bool
@@ -1781,7 +1525,7 @@ ManagerImpl::setSwitch(const std::string& switchName) {
// ACCOUNT handling
bool
-ManagerImpl::associateCallToAccount(CALLID callID, const AccountID& accountID)
+ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accountID)
{
if (getAccountFromCall(callID) == AccountNULL) { // nothing with the same ID
if ( accountExists(accountID) ) { // account id exist in AccountMap
@@ -1797,7 +1541,7 @@ ManagerImpl::associateCallToAccount(CALLID callID, const AccountID& accountID)
}
AccountID
-ManagerImpl::getAccountFromCall(const CALLID callID)
+ManagerImpl::getAccountFromCall(const CallID& callID)
{
ost::MutexLock m(_callAccountMapMutex);
CallAccountMap::iterator iter = _callAccountMap.find(callID);
@@ -1809,7 +1553,7 @@ ManagerImpl::getAccountFromCall(const CALLID callID)
}
bool
-ManagerImpl::removeCallAccount(CALLID callID)
+ManagerImpl::removeCallAccount(const CallID& callID)
{
ost::MutexLock m(_callAccountMapMutex);
if ( _callAccountMap.erase(callID) ) {
@@ -1818,15 +1562,19 @@ ManagerImpl::removeCallAccount(CALLID callID)
return false;
}
-CALLID
+CallID
ManagerImpl::getNewCallID()
{
- CALLID random_id = (unsigned)rand();
+ std::ostringstream random_id("s");
+ random_id << (unsigned)rand();
+
// when it's not found, it return ""
- while (getAccountFromCall(random_id) != AccountNULL) {
- random_id = rand();
+ while (getAccountFromCall(random_id.str()) != AccountNULL) {
+ random_id.clear();
+ random_id << "s";
+ random_id << (unsigned)rand();
}
- return random_id;
+ return random_id.str();
}
short
@@ -1837,7 +1585,7 @@ ManagerImpl::loadAccountMap()
_accountMap[ACCOUNT_SIP0] = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, ACCOUNT_SIP0);
nbAccount++;
- _accountMap[ACCOUNT_AIX0] = AccountCreator::createAccount(AccountCreator::AIX_ACCOUNT, ACCOUNT_AIX0);
+ _accountMap[ACCOUNT_IAX0] = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, ACCOUNT_IAX0);
nbAccount++;
return nbAccount;
@@ -1907,7 +1655,7 @@ bool ManagerImpl::testCallAccountMap()
if ( removeCallAccount(1) != false ) {
_debug("TEST: removeCallAccount with empty list failed\n");
}
- CALLID newid = getNewCallID();
+ CallID newid = getNewCallID();
if ( associateCallToAccount(newid, "acc0") == false ) {
_debug("TEST: associateCallToAccount with new CallID empty list failed\n");
}
@@ -1917,7 +1665,7 @@ bool ManagerImpl::testCallAccountMap()
if ( getAccountFromCall( newid ) != "acc0" ) {
_debug("TEST: getAccountFromCall don't return the good account id\n");
}
- CALLID secondnewid = getNewCallID();
+ CallID secondnewid = getNewCallID();
if ( associateCallToAccount(secondnewid, "xxxx") == true ) {
_debug("TEST: associateCallToAccount with unknown account id failed\n");
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index 7e37bc9f56285865a4e6872d3a8c859ce17cee51..e47b2a4cca91c13f6996eb071a5faee744d84bc6 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -25,16 +25,17 @@
#include <string>
#include <vector>
+#include <set>
#include <map>
#include <cc++/thread.h>
#include "stund/stun.h"
-#include "call.h"
#include "observer.h"
#include "config/config.h"
+
#include "account.h"
+#include "call.h"
-//#include "audio/audiodevice.h"
#include "audio/tonelist.h" // for Tone::TONEID declaration
#include "audio/audiofile.h"
#include "audio/dtmf.h"
@@ -42,44 +43,40 @@
class AudioLayer;
class CodecDescriptor;
-
class GuiFramework;
-
class TelephoneTone;
-
class VoIPLink;
#ifdef USE_ZEROCONF
class DNSService;
#endif
-#define NOTIFICATION_LEN 250
// Status
-#define CONNECTED_STATUS "Connected"
-#define LOGGED_IN_STATUS "Logged in"
-#define RINGING_STATUS "Ringing"
-#define TRYING_STATUS "Trying ..."
-#define HANGUP_STATUS "Hang up"
-#define ONHOLD_STATUS "On hold ..."
-#define TRANSFER_STATUS "Transfer to:"
-#define MUTE_ON_STATUS "Mute on"
-#define ENTER_NUMBER_STATUS "Enter Phone Number:"
+//#define CONNECTED_STATUS "Connected"
+//#define LOGGED_IN_STATUS "Logged in"
+//#define RINGING_STATUS "Ringing"
+//#define TRYING_STATUS "Trying ..."
+//#define HANGUP_STATUS "Hang up"
+//#define ONHOLD_STATUS "On hold ..."
+//#define TRANSFER_STATUS "Transfer to:"
+//#define MUTE_ON_STATUS "Mute on"
+//#define ENTER_NUMBER_STATUS "Enter Phone Number:"
-/*
- * Define a type for a list of call
+/**
+ * Define a type for a AccountMap container
*/
-typedef std::vector< Call* > CallVector;
-
+typedef std::map<AccountID, Account*> AccountMap;
+
/**
* Define a type for a CallID to AccountID Map inside ManagerImpl
*/
-typedef std::map<CALLID, AccountID> CallAccountMap;
+typedef std::map<CallID, AccountID> CallAccountMap;
/**
- * Define a type for a AccountMap container
+ * Define a type for CallID vector (waiting list, incoming not answered)
*/
-typedef std::map<AccountID, Account*> AccountMap;
-
+typedef std::set<CallID> CallIDSet;
+
/**
* To send multiple string
*/
@@ -91,87 +88,107 @@ public:
~ManagerImpl (void);
// Init a new VoIPLink, audio codec and audio driver
+ /**
+ * Initialisation of thread (sound) and map
+ */
void init (void);
+
+ /**
+ * Terminate all thread (sound, link) and clear map
+ */
void terminate (void);
- // Set the graphic user interface
- void setGui (GuiFramework* gui);
+ /**
+ * Set the graphic user interface : only GuiServer right now
+ * @param gui A GuiFramework gui implmentation
+ */
+ void setGui (GuiFramework* gui) { _gui = gui; }
// Accessor to audiodriver
- // it's multi-thread and use mutex internally
+ // it's multi-thread and use mutex internally
AudioLayer* getAudioDriver(void) const { return _audiodriverPA ;}
- // Codec Descriptor
+ /**
+ * Get a descriptor map of codec available
+ */
CodecDescriptorMap& getCodecDescriptorMap(void) {return _codecDescriptorMap;}
- /*
- * Attribute a new random id for a new call
- * and check if it's already attributed to existing calls.
- * If not exists, returns 'id' otherwise return 0
- */
- CALLID generateNewCallId (void);
-
- /*
- * Add a new call at the end of the CallVector with identifiant 'id'
- */
- Call* pushBackNewCall (CALLID id, Call::CallType type);
- void callSetInfo(CALLID id, const std::string& name, const std::string& number);
- bool callCanBeAnswered(CALLID id);
- bool callCanBeClosed(CALLID id);
- bool callIsOnHold(CALLID id);
-
- /*
+ /**
* Functions which occur with a user's action
*/
- int outgoingCall (const std::string& to);
- int hangupCall (CALLID id);
- int cancelCall (CALLID id);
- int answerCall (CALLID id);
- int onHoldCall (CALLID id);
- int offHoldCall (CALLID id);
- int transferCall (CALLID id, const std::string& to);
+ bool outgoingCall(const AccountID& accountId, const CallID& id, const std::string& to);
+ bool answerCall(const CallID& id);
+ bool hangupCall(const CallID& id);
+ bool cancelCall(const CallID& id);
+ bool onHoldCall(const CallID& id);
+ bool offHoldCall(const CallID& id);
+ bool transferCall(const CallID& id, const std::string& to);
void mute();
void unmute();
- int refuseCall (CALLID id);
+ bool refuseCall(const CallID& id);
+ /** Save config on file */
bool saveConfig (void);
- bool registerVoIPLink (void);
- bool unregisterVoIPLink (void);
+ /**
+ * Initialize action (main thread)
+ * @param accountId Account to register
+ * @return true if setRegister is call without failure, else return false
+ */
+ bool registerVoIPLink(const AccountID& accountId);
+ /**
+ * Unregister an account
+ * @param accountId Account to unregister
+ * @return true if the unregister method is send correctly
+ */
+ bool unregisterVoIPLink(const AccountID& accountId);
- bool sendTextMessage(const std::string& account, const std::string& to, const std::string& message);
+ bool sendTextMessage(const AccountID& accountId, const std::string& to, const std::string& message);
- /**
+ /*
* Handle choice of the DTMF-send-way
*
* @param id: callid of the line.
* @param code: pressed key.
*/
- bool sendDtmf (CALLID id, char code);
- bool playDtmf (char code);
+ bool sendDtmf(const CallID& id, char code);
+ bool playDtmf(char code);
bool playTone ();
void stopTone();
- CALLID getCurrentCallId() { ost::MutexLock m(_mutex); return _currentCallId; }
-
- int incomingCall (CALLID id, const std::string& name, const std::string& number);
- void peerAnsweredCall (CALLID id);
- int peerRingingCall (CALLID id);
- int peerHungupCall (CALLID id);
- void incomingMessage(const std::string& message);
-
- void displayTextMessage (CALLID id, const std::string& message);
- void displayErrorText (CALLID id, const std::string& message);
- void displayError (const std::string& error);
- void displayStatus (const std::string& status);
+
+ // From links
+ /**
+ * When receiving a new incoming call, add it to the callaccount map
+ * and notify user
+ * @param call A call pointer
+ * @param accountid an account id
+ * @return true if the call was added correctly
+ */
+ bool incomingCall(Call* call, const AccountID& accountId);
+ void peerAnsweredCall(const CallID& id);
+ void peerRingingCall(const CallID& id);
+ void peerHungupCall(const CallID& id);
+ void incomingMessage(const AccountID& accountId, const std::string& message);
+
+ void displayTextMessage (const CallID& id, const std::string& message);
+ void displayErrorText (const CallID& id, const std::string& message);
+ void displayError(const std::string& error);
+ void displayStatus(const std::string& status);
void displayConfigError(const std::string& message);
- void startVoiceMessageNotification (const std::string& nb_msg);
- void stopVoiceMessageNotification (void);
+ void startVoiceMessageNotification(const AccountID& accountId, const std::string& nb_msg);
+ void stopVoiceMessageNotification(const AccountID& accountId);
- void registrationSucceed();
- void registrationFailed();
+ /** Notify the user that registration succeeded */
+ void registrationSucceed(const AccountID& accountId);
+ /** Notify the user that registration succeeded */
+ void registrationFailed(const AccountID& accountId);
// configuration function requests
+ /** Start events thread*/
+ // TODO: receive account name
bool getEvents();
+
+ //
bool getZeroconf(const std::string& sequenceId);
bool attachZeroconfEvents(const std::string& sequenceId, Pattern::Observer& observer);
bool detachZeroconfEvents(Pattern::Observer& observer);
@@ -182,30 +199,33 @@ public:
bool setConfig(const std::string& section, const std::string& name, int value);
bool getConfigList(const std::string& sequenceId, const std::string& name);
void selectAudioDriver(void);
+ /** Set Audio Driver with switchName == audiodriver */
bool setSwitch(const std::string& switchName);
// configuration function for extern
// throw an Conf::ConfigTreeItemException if not found
+ /** Get a int from the config tree */
int getConfigInt(const std::string& section, const std::string& name);
+ /** Get a string from the config tree */
std::string getConfigString(const std::string& section, const std::string& name);
- /*
+ /**
* Handle audio sounds heard by a caller while they wait for their
* connection to a called party to be completed.
*/
void ringback ();
- /*
+ /**
* Handle played music when an incoming call occurs
*/
void ringtone ();
void congestion ();
- void callBusy(CALLID id);
- void callFailure(CALLID id);
+ void callBusy(const CallID& id);
+ void callFailure(const CallID& id);
- // return 0 if no tone (init before calling this function)
+ /** @return 0 if no tone (init before calling this function) */
AudioLoop* getTelephoneTone();
- // return 0 if the wav is stopped
+ /** @return 0 if the wav is stopped */
AudioLoop* getTelephoneFile();
/**
@@ -213,20 +233,12 @@ public:
* new call, not anwsered or refused
*/
bool incomingCallWaiting(void);
- /*
+ /**
* Notification of incoming call when you are already busy
*/
- void notificationIncomingCall (void);
+ void notificationIncomingCall(void);
+
- /*
- * Get information about firewall
- * @param stunSvrAddr: stun server
- * @param port port number to open to test the connection
- * @return true if the connection is successful
- */
- bool getStunInfo(StunAddress4& stunSvrAddr, int port);
- bool useStun (void);
-
/*
* Inline functions to manage volume control
* Read by main thread and AudioLayer thread
@@ -237,20 +249,40 @@ public:
unsigned short getMicVolume(void) { return _mic_volume; }
void setMicVolume(unsigned short mic_vol) { _mic_volume = mic_vol; }
+ // Manage information about firewall
/*
- * Manage information about firewall
+ * Get information about firewall
+ * @param stunSvrAddr: stun server
+ * @param port port number to open to test the connection
+ * @return true if the connection is successful
*/
+ bool getStunInfo(StunAddress4& stunSvrAddr, int port);
+
inline int getFirewallPort (void) { return _firewallPort; }
inline void setFirewallPort (int port) { _firewallPort = port; }
inline std::string getFirewallAddress (void) { return _firewallAddr; }
- /*
+ /**
+ * If you are behind a NAT, you have to use STUN server, specified in
+ * STUN configuration(you can change this one by default) to give you an
+ * public IP address and assign a port number.
+ * Note: Set firewall port/address retreive
+ * @param port : on which port we want to listen to
+ *
+ * Return true if we are behind a NAT (without error)
+ */
+ bool behindNat(int port);
+
+ /**
* Init default values for the different fields
*/
void initConfigFile (void);
+
+ /**
+ * Tell if the setup was already loaded
+ */
bool hasLoadedSetup() { return _setupLoaded; }
-
enum REGISTRATION_STATE {
UNREGISTERED,
REGISTERED,
@@ -259,6 +291,24 @@ public:
REGISTRATION_STATE getRegistrationState() { return _registerState; }
+ /** Return a new random callid that is not present in the list
+ * @return a brand new callid
+ */
+ CallID getNewCallID();
+
+ /**
+ * Get the current call id
+ * @return the call id or ""
+ */
+ const CallID& getCurrentCallId();
+
+ /**
+ * Check if a call is the current one
+ * @param id the new callid
+ * @return if the id is the current call
+ */
+ bool isCurrentCall(const CallID& callId);
+
private:
/**
* Create .PROGNAME directory in home user and create
@@ -304,14 +354,24 @@ private:
bool getCountryTones(const std::string& sequenceId);
void sendCountryTone(const std::string& sequenceId, int index, const std::string& name);
- /*
- * Erase the Call(id) from the CallVector
- * Protected by other function by _mutex lock
+ /**
+ * Tell if there is a current call processed
+ * @return true if there is a current call
+ */
+ bool hasCurrentCall();
+
+ /**
+ * Switch of current call id
+ * @param id the new callid
*/
- void deleteCall (CALLID id);
- Call* getCall (CALLID id);
- void setCurrentCallId(CALLID id);
- void removeCallFromCurrent(CALLID id);
+ void switchCall(const CallID& id);
+
+ /** Current Call ID */
+ CallID _currentCallId2;
+
+ /** Protected current call access */
+ ost::Mutex _currentCallMutex;
+
/*
* Play one tone
@@ -347,45 +407,41 @@ private:
short _spkr_volume;
short _mic_volume;
short _mic_volume_before_mute;
-
- //
// End of sound variable
- //
+
// Multithread variable (protected by _mutex)
//
/** Mutex to protect access to code section */
ost::Mutex _mutex;
- /* Vector of calls */
- CallVector _callVector;
- // Current callid : protected implicitely by function using _mutex
- CALLID _currentCallId;
- // functions that set mutex:
- // terminate, pushBackNewCall, generateNewCallId, outgoingCall (after gen/push)
- // hangupCall, cancelCall, answerCall, onHoldCall, offHoldCall, transferCall, refuseCall,
- // callSetInfo, callCanBeClosed, callCanBeAnswered, callIsOnHold, incomingCall,
- // peerAnsweredCall, peerRingingCall, peerHunguCall, callBusy, callFailure
- // getCallStatus, getCurrentCallId
- // functions that are called by those functions
- // getCall, deleteCall, stopTone, switchCall, decWaitingCall, setCurrentCallId, getAudioDriver, ringtone, incWaitingCall
-
- // warning, incomingCallWaiting | incWaitingCall | decWaitingCall are prtected by _incomingCallMutex
-
//
// Multithread variable (non protected)
//
GuiFramework* _gui;
+ /** Waiting Call Vectors */
+ CallIDSet _waitingCall;
+ /** Protect waiting call list, access by many voip/audio threads */
+ ost::Mutex _waitingCallMutex;
+ /** Number of waiting call, synchronize with waitingcall callidvector */
+ unsigned int _nbIncomingWaitingCall;
/**
- * Multithreaded
- * Incomings Call:
+ * Add incoming callid to the waiting list
+ * @param id CallID to add
*/
- ost::Mutex _incomingCallMutex;
- unsigned int _nbIncomingWaitingCall;
- void incWaitingCall(void);
- void decWaitingCall(void);
-
+ void addWaitingCall(const CallID& id);
+ /**
+ * Remove incoming callid to the waiting list
+ * @param id CallID to remove
+ */
+ void removeWaitingCall(const CallID& id);
+ /**
+ * Tell if a call is waiting and should be remove
+ * @param id CallID to test
+ * @return true if the call is waiting
+ */
+ bool isWaitingCall(const CallID& id);
/**
* Path of the ConfigFile
@@ -401,13 +457,10 @@ private:
// return false if exosip or the network checking failed
bool initRegisterVoIPLink();
// true if we tried to register Once
- bool _hasTriedToRegister;
+ bool _hasTriedToRegister;
// Register state
REGISTRATION_STATE _registerState;
-
- void switchCall(CALLID id);
-
// tell if we have zeroconf is enabled
int _hasZeroconf;
@@ -417,8 +470,6 @@ private:
DNSService *_DNSService;
#endif
-// CALLID
-
/** Map to associate a CallID to the good account */
CallAccountMap _callAccountMap;
/** Mutex to lock the call account map (main thread + voiplink thread) */
@@ -430,26 +481,21 @@ private:
* @param accountID the known accountID present in accountMap
* @return true if the new association is create
*/
- bool associateCallToAccount(CALLID callID, const AccountID& accountID);
+ bool associateCallToAccount(const CallID& callID, const AccountID& accountID);
/** Return the AccountID from a CallID
* Protected by mutex
* @param callID the CallID in the list
* @return the accountID associated or "" if the callID is not found
*/
- AccountID getAccountFromCall(const CALLID callID);
+ AccountID getAccountFromCall(const CallID& callID);
/** Remove a CallID/AccountID association
* Protected by mutex
* @param callID the CallID to remove
* @return true if association is removed
*/
- bool removeCallAccount(CALLID callID);
-
- /** Return a new random callid that is not present in the list
- * @return a brand new callid
- */
- CALLID getNewCallID();
+ bool removeCallAccount(const CallID& callID);
/** Contains a list of account (sip, aix, etc) and their respective voiplink/calls */
AccountMap _accountMap;
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index 153d2d172fc9556edff9e1ac16d0666eefb05618..54d9d5cfad2a38e32703ad8e88526500a8f44126 100644
--- a/src/sipaccount.cpp
+++ b/src/sipaccount.cpp
@@ -18,14 +18,14 @@
*/
#include "sipaccount.h"
#include "sipvoiplink.h"
+#include "manager.h"
#define SIP_FULL_NAME "SIP.fullName"
#define SIP_USER_PART "SIP.userPart"
-#define SIP_AUTH_USER_NAME "SIP.username"
+#define SIP_AUTH_NAME "SIP.username"
#define SIP_PASSWORD "SIP.password"
#define SIP_HOST_PART "SIP.hostPart"
#define SIP_PROXY "SIP.proxy"
-#define SIP_AUTO_REGISTER "SIP.autoregister"
#define SIP_STUN_SERVER "STUN.STUNserver"
#define SIP_USE_STUN "STUN.useStun"
@@ -46,7 +46,8 @@ bool
SIPAccount::createVoIPLink()
{
if (!_link) {
- _link = new SipVoIPLink();
+ //_link = new SipVoIPLink();
+ _link = new SIPVoIPLink(_accountID);
}
return (_link != 0 ? true : false);
}
@@ -55,7 +56,14 @@ bool
SIPAccount::registerAccount()
{
if (_link && !_registered) {
- _registered = (_link->setRegister() >= 0) ? true : false;
+ SIPVoIPLink* tmplink = dynamic_cast<SIPVoIPLink*> (_link);
+ if (tmplink) {
+ tmplink->setProxy(Manager::instance().getConfigString(_accountID,SIP_PROXY));
+ tmplink->setUserPart(Manager::instance().getConfigString(_accountID,SIP_USER_PART));
+ tmplink->setAuthName(Manager::instance().getConfigString(_accountID,SIP_AUTH_NAME));
+ tmplink->setPassword(Manager::instance().getConfigString(_accountID,SIP_PASSWORD));
+ }
+ _registered = _link->setRegister();
}
return _registered;
}
@@ -64,7 +72,7 @@ bool
SIPAccount::unregisterAccount()
{
if (_link && _registered) {
- _registered = (_link->setUnregister() == 0) ? false : true;
+ _registered = _link->setUnregister();
}
return !_registered;
}
@@ -73,6 +81,13 @@ bool
SIPAccount::init()
{
if (_link && !_enabled) {
+ _link->setFullName(Manager::instance().getConfigString(_accountID,SIP_FULL_NAME));
+ _link->setHostName(Manager::instance().getConfigString(_accountID,SIP_HOST_PART));
+ int useStun = Manager::instance().getConfigInt(_accountID,SIP_USE_STUN);
+ SIPVoIPLink* tmplink = dynamic_cast<SIPVoIPLink*> (_link);
+ if (tmplink) {
+ tmplink->setUseStun( useStun!=0 ? true : false);
+ }
_link->init();
_enabled = true;
return true;
@@ -102,10 +117,18 @@ SIPAccount::initConfig(Conf::ConfigTree& config)
config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_ENABLE,"1", type_int));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_FULL_NAME, "", type_str));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_USER_PART, "", type_str));
- config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_AUTH_USER_NAME, "", type_str));
+ config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_HOST_PART, "", type_str));
+ config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_AUTH_NAME, "", type_str));
+ config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_PASSWORD, "", type_str));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_PROXY, "", type_str));
- config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_AUTO_REGISTER, "1", type_int));
+ config.addConfigTreeItem(section, Conf::ConfigTreeItem(CONFIG_ACCOUNT_AUTO_REGISTER, "1", type_int));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_STUN_SERVER, "stun.fwdnet.net:3478", type_str));
config.addConfigTreeItem(section, Conf::ConfigTreeItem(SIP_USE_STUN, "0", type_int));
}
+void
+SIPAccount::loadConfig()
+{
+ _shouldInitOnStart = Manager::instance().getConfigInt(_accountID, CONFIG_ACCOUNT_ENABLE) ? true : false;
+ _shouldRegisterOnStart = Manager::instance().getConfigInt(_accountID, CONFIG_ACCOUNT_AUTO_REGISTER) ? true : false;
+}
diff --git a/src/sipaccount.h b/src/sipaccount.h
index 157c0ad51863f5defe0bcca5aad72baba527b493..c5da7b18fe833a28525af96b300d5dfcac390e8c 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -30,10 +30,11 @@ class SIPAccount : public Account
public:
SIPAccount(const AccountID& accountID);
- ~SIPAccount();
+ virtual ~SIPAccount();
/* virtual Account function implementation */
void initConfig(Conf::ConfigTree& config);
+ void loadConfig();
bool registerAccount();
bool unregisterAccount();
bool init();
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 36997919e07add582bba505b6ffb5b81f4be1bbe..f9e56662987407c7579b88668bbf3d9d361da449 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -1,270 +1,107 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+/*
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
- * Portions Copyright (C) 2002,2003 Aymeric Moizard <jack@atosc.org>
+ * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
- * This is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as
- * published by the Free Software Foundation; either version 2,
- * or (at your option) any later version.
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
*
- * This is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
*
- * You should have received a copy of the GNU General Public
- * License along with dpkg; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include <osipparser2/sdp_message.h>
-#include <string.h> // strcpy
-
-// For AF_INET
-#include <sys/socket.h>
-#include <sstream>
-
-#include "global.h"
-#include "audio/audiocodec.h"
-#include "audio/codecDescriptor.h"
#include "sipcall.h"
+#include "global.h" // for _debug
+#include <sstream> // for media buffer
-SipCall::SipCall (CALLID id, CodecDescriptorMap& codecMap) : _localIp("127.0.0.1"), _codecMap(codecMap)
-{
- _id = id; // Same id of Call object
- _cid = 0; // call id, from the sipvoiplink
- _did = 0; // dialog id
- _tid = 0; // transaction id
-
- _standby = false;
- _status_code = 0;
-
- alloc(); // char* allocation
- _audiocodec = 0;
-
- _local_audio_port = 0;
- _remote_sdp_audio_port = 0;
- _local_sendrecv = 0; /* _SENDRECV, _SENDONLY, _RECVONLY */
- _remote_sendrecv = 0;
-
- _reinvite = false;
-}
-
-
-SipCall::~SipCall (void)
-{
- dealloc();
- //delete _audiocodec; don't delete it, the Manager will do it...
- _audiocodec = 0;
-}
-
-void
-SipCall::setLocalAudioPort (int newport)
-{
- _local_audio_port = newport;
-}
-
-int
-SipCall::getLocalAudioPort (void)
-{
- return _local_audio_port;
-}
-
-void
-SipCall::setId (CALLID id)
-{
- _id = id;
-}
-
-CALLID
-SipCall::getId (void)
-{
- return _id;
-}
-
-void
-SipCall::setDid (int did)
-{
- _did = did;
-}
-
-int
-SipCall::getDid (void)
-{
- return _did;
-}
-
-void
-SipCall::setCid (int cid)
-{
- _cid = cid;
-}
-
-int
-SipCall::getCid (void)
-{
- return _cid;
-}
+#define _SENDRECV 0
+#define _SENDONLY 1
+#define _RECVONLY 2
-void
-SipCall::setTid (int tid)
+SIPCall::SIPCall(const CallID& id, Call::CallType type) : Call(id, type),
+ _localIPAddress(""), _remoteIPAddress("")
{
- _tid = tid;
+ _cid = 0;
+ _did = 0;
+ _tid = 0;
+ _audioCodec = 0;
+ _localAudioPort = 0;
+ _localExternalAudioPort = 0;
+ _remoteAudioPort = 0;
}
-int
-SipCall::getTid (void)
+SIPCall::~SIPCall()
{
- return _tid;
}
-int
-SipCall::getRemoteSdpAudioPort (void)
+CodecDescriptorMap&
+SIPCall::getCodecMap()
{
- return _remote_sdp_audio_port;
+ return _codecMap;
}
-char*
-SipCall::getRemoteSdpAudioIp (void)
+/**
+ * Answer incoming call correclty before telling the user
+ * @param event eXosip Event
+ */
+bool
+SIPCall::SIPCallInvite(eXosip_event_t *event)
{
- return _remote_sdp_audio_ip;
-}
-
-// newIncomingCall is called when the IP-Phone user receives a new call.
-int
-SipCall::newIncomingCall (eXosip_event_t *event) {
-
- _cid = event->cid;
- _did = event->did;
- _tid = event->tid;
-
if (_did < 1 && _cid < 1) {
- return -1; /* not enough information for this event?? */
+ _debug("SIP Failure: Invalid cid and did\n");
+ return false;
}
- osip_strncpy (_textinfo, event->textinfo, 255);
- if (event->response != NULL) {
- _status_code = event->response->status_code;
- snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
- _debug(" Status: code:%1$d reason:%2$s\n", _status_code, _reason_phrase);
+ if (event->request == NULL) {
+ _debug("SIP Failure: No request into the event\n");
+ return false;
}
- strcpy(_remote_uri, "");
- _name = "";
- _number = "";
- if (event->request != NULL) {
- char *tmp = NULL;
-
- osip_from_to_str(event->request->from, &tmp);
- if (tmp != NULL) {
- snprintf (_remote_uri, 255, "%s", tmp);
- osip_free (tmp);
-
- // Get the name/number
- osip_from_t *from;
- osip_from_init(&from);
- osip_from_parse(from, _remote_uri);
- char *name = osip_from_get_displayname(from);
- if ( name != NULL ) {
- _name = name;
- }
- osip_uri_t* url = osip_from_get_url(from);
- if ( url != NULL && url->username != NULL) {
- _number = url->username;
- }
- osip_from_free(from);
- }
- }
- _debug(" Name: %s\n", _name.c_str());
- _debug(" Number: %s\n", _number.c_str());
- _debug(" Remote URI: %s\n", _remote_uri);
+ setCid(event->cid);
+ setDid(event->did);
+ setTid(event->tid);
- /* negotiate payloads */
- sdp_message_t *remote_sdp = NULL;
- if (event->request != NULL) {
- eXosip_lock();
- remote_sdp = eXosip_get_sdp_info (event->request);
- eXosip_unlock();
+ setPeerInfoFromRequest(event);
+
+ sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
+ if (remote_sdp == 0) {
+ return false;
}
- if (remote_sdp == NULL) {
- _debug("< Sending 400 Bad Request (no SDP)\n");
- eXosip_lock();
- eXosip_call_send_answer (_tid, 400, NULL);
- eXosip_unlock();
- return -1;
- }
- /* TODO: else build an offer */
- // Remote Media IP
- eXosip_lock();
- sdp_connection_t *conn = eXosip_get_audio_connection (remote_sdp);
- eXosip_unlock();
- if (conn != NULL && conn->c_addr != NULL) {
- snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
+ sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
+ if (remote_med == 0) {
+ sdp_message_free (remote_sdp);
+ return false;
}
- // Remote Media Port
- eXosip_lock();
- sdp_media_t *remote_med = eXosip_get_audio_media (remote_sdp);
- eXosip_unlock();
-
- if (remote_med == NULL || remote_med->m_port == NULL) {
- // no audio media proposed
- _debug("< Sending 415 Unsupported media type\n");
- eXosip_lock();
- eXosip_call_send_answer (_tid, 415, NULL);
- eXosip_unlock();
+ if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
+ _debug("SIP Failure: unable to set IP address and port from SDP\n");
sdp_message_free (remote_sdp);
- return -1;
+ return false;
}
- _remote_sdp_audio_port = atoi(remote_med->m_port);
- _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
- // Remote Payload
- char *tmp = NULL;
- if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
- int pos = 0;
- while (!osip_list_eol (remote_med->m_payloads, pos)) {
- tmp = (char *) osip_list_get (remote_med->m_payloads, pos);
- if (tmp != NULL ) {
- int payload = atoi(tmp);
- // stop if we find a correct codec
- if (0 != _codecMap.getCodec((CodecType)payload)){
- break;
- }
- }
- tmp = NULL;
- pos++;
- }
- }
- setAudioCodec(0);
- if (tmp != NULL) {
- int payload = atoi (tmp);
- _debug(" Payload: %d\n", payload);
- setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
- }
- if (getAudioCodec() == 0) {
- _debug("< Sending 415 Unsupported media type\n");
- eXosip_lock();
- eXosip_call_send_answer (_tid, 415, NULL);
- eXosip_unlock();
+ if (!setAudioCodecFromSDP(remote_med, event->tid)) {
sdp_message_free (remote_sdp);
- return -1;
+ return false;
}
osip_message_t *answer = 0;
eXosip_lock();
_debug("< Building Answer 183\n");
- if (0 == eXosip_call_build_answer (_tid, 183, &answer)) {
+ if (0 == eXosip_call_build_answer (event->tid, 183, &answer)) {
if ( 0 != sdp_complete_message(remote_sdp, answer)) {
osip_message_free(answer);
// Send 415 Unsupported media type
- _debug("< Sending Answer 415\n");
- eXosip_call_send_answer (_tid, 415, NULL);
+ _debug("< Sending Answer 415 : unsupported media type\n");
+ eXosip_call_send_answer (event->tid, 415, NULL);
} else {
sdp_message_t *local_sdp = eXosip_get_sdp_info(answer);
@@ -274,8 +111,8 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
}
if (local_sdp != NULL && local_med != NULL) {
/* search if stream is sendonly or recvonly */
- _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
- _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
+ int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
+ int _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
_debug(" Remote SendRecv: %d\n", _remote_sendrecv);
_debug(" Local SendRecv: %d\n", _local_sendrecv);
if (_local_sendrecv == _SENDRECV) {
@@ -286,149 +123,73 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
sdp_message_free (local_sdp);
}
_debug("< Sending answer 183\n");
- if (0 != eXosip_call_send_answer (_tid, 183, answer)) {
+ if (0 != eXosip_call_send_answer (event->tid, 183, answer)) {
_debug("SipCall::newIncomingCall: cannot send 183 progress?\n");
}
}
}
eXosip_unlock ();
- sdp_message_free (remote_sdp);
- return 0;
+ sdp_message_free (remote_sdp);
+ return true;
}
-// newReinviteCall is called when the IP-Phone user receives a change in the call
-// it's almost an newIncomingCall but we send a 200 OK
-// See: 3.7. Session with re-INVITE (IP Address Change)
-int
-SipCall::newReinviteCall (eXosip_event_t *event) {
-
- _cid = event->cid;
- _did = event->did;
- _tid = event->tid;
-
+/**
+ * newReinviteCall is called when the IP-Phone user receives a change in the call
+ * it's almost an newIncomingCall but we send a 200 OK
+ * See: 3.7. Session with re-INVITE (IP Address Change)
+ * @param event eXosip Event
+ * @return true if ok
+ */
+bool
+SIPCall::SIPCallReinvite(eXosip_event_t *event)
+{
if (_did < 1 && _cid < 1) {
- return -1; /* not enough information for this event?? */
+ _debug("SIP Failure: Invalid cid and did\n");
+ return false;
}
- osip_strncpy (_textinfo, event->textinfo, 255);
- if (event->response != NULL) {
- _status_code = event->response->status_code;
- snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
- _debug(" Status: %d %s\n", _status_code, _reason_phrase);
+ if (event->request == NULL) {
+ _debug("SIP Failure: No request into the event\n");
+ return false;
}
- strcpy(_remote_uri, "");
- _name = "";
- _number = "";
- if (event->request != NULL) {
- char *tmp = NULL;
-
- osip_from_to_str(event->request->from, &tmp);
- if (tmp != NULL) {
- snprintf (_remote_uri, 255, "%s", tmp);
- osip_free (tmp);
-
- // Get the name/number
- osip_from_t *from;
- osip_from_init(&from);
- osip_from_parse(from, _remote_uri);
- char *name = osip_from_get_displayname(from);
- if ( name != NULL ) {
- _name = name;
- }
- osip_uri_t* url = osip_from_get_url(from);
- if ( url != NULL && url->username != NULL ) {
- _number = url->username;
- }
- osip_from_free(from);
- }
- }
- _debug(" Name: %s\n", _name.c_str());
- _debug(" Number: %s\n", _number.c_str());
- _debug(" Remote URI: %s\n", _remote_uri);
+ setCid(event->cid);
+ setDid(event->did);
+ setTid(event->tid);
- /* negotiate payloads */
- sdp_message_t *remote_sdp = NULL;
- if (event->request != NULL) {
- eXosip_lock();
- remote_sdp = eXosip_get_sdp_info (event->request);
- eXosip_unlock();
- }
- if (remote_sdp == NULL) {
- _debug("< Sending 400 Bad Request (no sdp)\n");
- // Send 400 BAD REQUEST
- eXosip_lock();
- eXosip_call_send_answer (_tid, 400, NULL);
- eXosip_unlock();
- return 0;
- }
- /* TODO: else build an offer */
+ setPeerInfoFromRequest(event);
- // Remote Media IP
- eXosip_lock();
- sdp_connection_t *conn = eXosip_get_audio_connection (remote_sdp);
- eXosip_unlock();
- if (conn != NULL && conn->c_addr != NULL) {
- snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
+ sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
+ if (remote_sdp == 0) {
+ return false;
}
- // Remote Media Port
- eXosip_lock();
- sdp_media_t *remote_med = eXosip_get_audio_media (remote_sdp);
- eXosip_unlock();
-
- if (remote_med == NULL || remote_med->m_port == NULL) {
- // no audio media proposed
- _debug("< Sending 415 Unsupported media type\n");
- eXosip_lock();
- eXosip_call_send_answer (_tid, 415, NULL);
- eXosip_unlock();
+ sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
+ if (remote_med == 0) {
sdp_message_free (remote_sdp);
- return 0;
+ return false;
}
- _remote_sdp_audio_port = atoi(remote_med->m_port);
- _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
- // Remote Payload
- char *tmp = NULL;
- if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
- int pos = 0;
- while (!osip_list_eol (remote_med->m_payloads, pos)) {
- tmp = (char *) osip_list_get (remote_med->m_payloads, pos);
- if (tmp != NULL) {
- // we could check if the payload is active here...
- break;
- }
- tmp = NULL;
- pos++;
- }
+ if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
+ _debug("SIP Failure: unable to set IP address and port from SDP\n");
+ sdp_message_free (remote_sdp);
+ return false;
}
- setAudioCodec(0);
- if (tmp != NULL) {
- int payload = atoi (tmp);
- _debug(" Payload: %d\n", payload);
- setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
- }
- if (getAudioCodec() == 0) {
- _debug("< Sending 415 Unsupported media type\n");
- eXosip_lock();
- eXosip_call_send_answer (_tid, 415, NULL);
- eXosip_unlock();
+ if (!setAudioCodecFromSDP(remote_med, event->tid)) {
sdp_message_free (remote_sdp);
- return 0;
+ return false;
}
osip_message_t *answer = 0;
eXosip_lock();
_debug("< Building Answer 200\n");
- if (0 == eXosip_call_build_answer (_tid, 200, &answer)) {
+ if (0 == eXosip_call_build_answer (event->tid, 200, &answer)) {
if ( 0 != sdp_complete_message(remote_sdp, answer)) {
osip_message_free(answer);
// Send 415 Unsupported media type
- eXosip_call_send_answer (_tid, 415, NULL);
+ eXosip_call_send_answer (event->tid, 415, NULL);
_debug("< Sending Answer 415\n");
} else {
@@ -439,8 +200,8 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
}
if (local_sdp != NULL && local_med != NULL) {
/* search if stream is sendonly or recvonly */
- _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
- _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
+ int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
+ int _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
_debug(" Remote SendRecv: %d\n", _remote_sendrecv);
_debug(" Local SendRecv: %d\n", _local_sendrecv);
if (_local_sendrecv == _SENDRECV) {
@@ -451,88 +212,44 @@ SipCall::newReinviteCall (eXosip_event_t *event) {
sdp_message_free (local_sdp);
}
_debug("< Sending answer 200\n");
- if (0 != eXosip_call_send_answer (_tid, 200, answer)) {
+ if (0 != eXosip_call_send_answer (event->tid, 200, answer)) {
_debug("SipCall::newIncomingCall: cannot send 200 OK?\n");
}
}
}
eXosip_unlock ();
sdp_message_free (remote_sdp);
- _reinvite = true;
- return 0;
+ return true;
}
-int
-SipCall::ringingCall (eXosip_event_t *event) {
-
- this->_cid = event->cid;
- this->_did = event->did;
- this->_tid = event->tid;
-
- if (this->_did < 1 && this->_cid < 1) {
- return -1;
- }
-
- osip_strncpy (_textinfo, event->textinfo, 255);
-
- if (event->response != NULL) {
- _status_code = event->response->status_code;
- snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
- }
-
- if (event->request != NULL) {
- char *tmp = NULL;
-
- osip_from_to_str (event->request->from, &tmp);
- if (tmp != NULL) {
- snprintf (_remote_uri, 255, "%s", tmp);
- osip_free (tmp);
- }
- }
- return 0;
-}
-
-int
-SipCall::receivedAck (eXosip_event_t *event)
+ /**
+ * Peer answered to a call (on hold or not)
+ * @param event eXosip Event
+ * @return true if ok
+ */
+bool
+SIPCall::SIPCallAnswered(eXosip_event_t *event)
{
- _cid = event->cid;
- _did = event->did;
- return 0;
-}
-
-
-
-int
-SipCall::answeredCall(eXosip_event_t *event) {
- _cid = event->cid;
- _did = event->did;
-
- if (_did < 1 && _cid < 1) {
- return -1; /* not enough information for this event?? */
+ if (_did < 1 && _cid < 1) {
+ _debug("SIP Failure: Invalid cid and did\n");
+ return false;
}
- osip_strncpy(this->_textinfo, event->textinfo, 255);
- if (event->response != NULL) {
- _status_code = event->response->status_code;
- snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
+ if (event->request == NULL) {
+ _debug("SIP Failure: No request into the event\n");
+ return false;
+ }
- char *tmp = NULL;
+ setCid(event->cid);
+ setDid(event->did);
- osip_from_to_str (event->response->from, &tmp);
- if (tmp != NULL) {
- snprintf (_remote_uri, 255, "%s", tmp);
- osip_free (tmp);
- }
- }
- _debug(" Status: %d %s\n", _status_code, _reason_phrase);
- _debug(" From URI: %s\n", _remote_uri);
+ //setPeerInfoFromResponse()
eXosip_lock ();
{
osip_message_t *ack = NULL;
int i;
-
- i = eXosip_call_build_ack (_did, &ack);
+ i = eXosip_call_build_ack (event->did, &ack);
if (i != 0) {
_debug("SipCall::answeredCall: Cannot build ACK for call!\n");
} else {
@@ -554,114 +271,120 @@ SipCall::answeredCall(eXosip_event_t *event) {
sdp_message_free (remote_sdp);
_debug("< Send ACK\n");
- eXosip_call_send_ack (_did, ack);
+ eXosip_call_send_ack (event->did, ack);
}
}
eXosip_unlock ();
-
- return 0;
+ return true;
}
-
-void
-SipCall::answeredCall_without_hold (eXosip_event_t *event)
+ /**
+ * We retreive final SDP info if they changed
+ * @param event eXosip Event
+ * @return true if ok (change / no change) or false on error
+ */
+bool
+SIPCall::SIPCallAnsweredWithoutHold(eXosip_event_t *event)
{
- if (event->response == NULL ) { return; }
-
- // TODO: understand this code..
- if (_cid!=0) {
- eXosip_lock();
- sdp_message_t *sdp = eXosip_get_sdp_info (event->response);
- eXosip_unlock();
- if (sdp != NULL) {
- /* audio is started and session has just been modified */
- sdp_message_free (sdp);
- }
+ if (event->response == NULL || event->request == NULL) { return false; }
+
+ eXosip_lock();
+ sdp_message_t *remote_sdp = eXosip_get_sdp_info (event->response);
+ eXosip_unlock();
+ if (remote_sdp == NULL) {
+ _debug("SIP Failure: no remote sdp\n");
+ sdp_message_free(remote_sdp);
+ return false;
}
- if (event->request != NULL) { /* audio is started */
-
- eXosip_lock();
- sdp_message_t *local_sdp = eXosip_get_sdp_info (event->request);
- sdp_message_t *remote_sdp = eXosip_get_sdp_info (event->response);
- eXosip_unlock();
+ sdp_media_t *remote_med = getRemoteMedia(event->tid, remote_sdp);
+ if (remote_med==NULL) {
+ sdp_message_free(remote_sdp);
+ return false;
+ }
+ if ( ! setRemoteAudioFromSDP(remote_med, remote_sdp) ) {
+ sdp_message_free(remote_sdp);
+ return false;
+ }
- sdp_media_t *remote_med = NULL;
- char *tmp = NULL;
- if (remote_sdp == NULL) {
- _debug("SipCall::answeredCall_without_hold: No remote SDP body found for call\n");
- /* TODO: remote_sdp = retreive from ack above */
- } else {
- eXosip_lock();
- sdp_connection_t *conn = eXosip_get_audio_connection (remote_sdp);
- if (conn != NULL && conn->c_addr != NULL) {
- snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- }
+ char *tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
- remote_med = eXosip_get_audio_media (remote_sdp);
- if (remote_med != NULL && remote_med->m_port != NULL) {
- _remote_sdp_audio_port = atoi (remote_med->m_port);
- }
- eXosip_unlock();
- _debug(" Remote Audio: %s:%d\n", _remote_sdp_audio_ip, _remote_sdp_audio_port);
-
- if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0' &&
- remote_med != NULL) {
- tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
- }
+ setAudioCodec(0);
+ if (tmp != NULL) {
+ int payload = atoi (tmp);
+ _debug(" Remote Payload: %d\n", payload);
+ setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
+ }
- setAudioCodec(0);
- if (tmp != NULL) {
- int payload = atoi (tmp);
- _debug(" Remote Payload: %d\n", payload);
- setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
- }
+/*
+ // search if stream is sendonly or recvonly
+ _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
+ _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
+ if (_local_sendrecv == _SENDRECV) {
+ if (_remote_sendrecv == _SENDONLY)
+ _local_sendrecv = _RECVONLY;
+ else if (_remote_sendrecv == _RECVONLY)
+ _local_sendrecv = _SENDONLY;
}
+ _debug(" Remote Sendrecv: %d\n", _remote_sendrecv);
+ _debug(" Local Sendrecv: %d\n", _local_sendrecv);
+*/
+ sdp_message_free (remote_sdp);
+ return true;
+}
- if (local_sdp == NULL) {
- _debug("SipCall::answeredCall_without_hold: SDP body was probably in the ACK (TODO)\n");
- }
+const std::string&
+SIPCall::getLocalIp()
+{
+ ost::MutexLock m(_callMutex);
+ return _localIPAddress;
+}
- if (remote_sdp != NULL && local_sdp != NULL) {
- int audio_port = 0;
- eXosip_lock();
- sdp_media_t *local_med = eXosip_get_audio_media (local_sdp);
- eXosip_unlock();
- if (local_med != NULL && local_med->m_port != NULL) {
- audio_port = atoi (local_med->m_port);
- }
- _debug(" Local Audio port: %d\n", audio_port);
+unsigned int
+SIPCall::getLocalAudioPort()
+{
+ ost::MutexLock m(_callMutex);
+ return _localAudioPort;
+}
- if (tmp != NULL && audio_port > 0
- && _remote_sdp_audio_port > 0
- && _remote_sdp_audio_ip[0] != '\0') {
+unsigned int
+SIPCall::getRemoteAudioPort()
+{
+ ost::MutexLock m(_callMutex);
+ return _remoteAudioPort;
+}
- /* search if stream is sendonly or recvonly */
- _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
- _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
- if (_local_sendrecv == _SENDRECV) {
- if (_remote_sendrecv == _SENDONLY)
- _local_sendrecv = _RECVONLY;
- else if (_remote_sendrecv == _RECVONLY)
- _local_sendrecv = _SENDONLY;
- }
- }
- _debug(" Remote Sendrecv: %d\n", _remote_sendrecv);
- _debug(" Local Sendrecv: %d\n", _local_sendrecv);
- }
- sdp_message_free (local_sdp);
- sdp_message_free (remote_sdp);
- }
+const std::string&
+SIPCall::getRemoteIp()
+{
+ ost::MutexLock m(_callMutex);
+ return _remoteIPAddress;
}
-int
-SipCall::sdp_complete_message(sdp_message_t * remote_sdp,
- osip_message_t * msg)
+AudioCodec*
+SIPCall::getAudioCodec()
{
- char *tmp = NULL;
- char buf[4096];
+ ost::MutexLock m(_callMutex);
+ return _audioCodec;
+}
- char localip[128];
+void
+SIPCall::setAudioStart(bool start)
+{
+ ost::MutexLock m(_callMutex);
+ _audioStarted = start;
+}
+bool
+SIPCall::isAudioStarted()
+{
+ ost::MutexLock m(_callMutex);
+ return _audioStarted;
+}
+
+ //TODO: humm?
+int
+SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
+{
// Format port to a char*
if (remote_sdp == NULL) {
_debug("SipCall::sdp_complete_message: No remote SDP body found for call\n");
@@ -672,13 +395,11 @@ SipCall::sdp_complete_message(sdp_message_t * remote_sdp,
return -1;
}
- // this exosip is locked and is protected by other function
- eXosip_guess_localip(AF_INET, localip, 128);
-
std::ostringstream media;
// for each medias
int iMedia = 0;
+ char *tmp = NULL;
while (!osip_list_eol(remote_sdp->m_medias, iMedia)) {
sdp_media_t *remote_med = (sdp_media_t *)osip_list_get(remote_sdp->m_medias, iMedia);
if (remote_med == 0) { continue; }
@@ -712,19 +433,20 @@ SipCall::sdp_complete_message(sdp_message_t * remote_sdp,
media << "m=" << remote_med->m_media << " 0 << " << remote_med->m_proto << " \r\n";
} else {
// we add the media line + a=rtpmap list
- media << "m=" << remote_med->m_media << " " << _local_audio_port << " RTP/AVP " << listCodec.str() << "\r\n";
+ media << "m=" << remote_med->m_media << " " << getLocalExternAudioPort() << " RTP/AVP " << listCodec.str() << "\r\n";
media << listRtpMap.str();
}
}
iMedia++;
}
+ char buf[4096];
snprintf (buf, 4096,
"v=0\r\n"
"o=user 0 0 IN IP4 %s\r\n"
"s=session\r\n"
"c=IN IP4 %s\r\n"
"t=0 0\r\n"
- "%s\n", localip, localip, media.str().c_str());
+ "%s\n", getLocalIp().c_str(), getLocalIp().c_str(), media.str().c_str());
osip_message_set_body (msg, buf, strlen (buf));
osip_message_set_content_type (msg, "application/sdp");
@@ -732,78 +454,200 @@ SipCall::sdp_complete_message(sdp_message_t * remote_sdp,
return 0;
}
-int
-SipCall::sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med)
+
+ // TODO: hum???
+int
+SIPCall::sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med)
{
- int pos;
- int pos_media;
-
- /* test media attributes */
- pos = 0;
- while (!osip_list_eol (med->a_attributes, pos)) {
- sdp_attribute_t *at;
-
- at = (sdp_attribute_t *) osip_list_get (med->a_attributes, pos);
- if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendonly")) {
- return _SENDONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "recvonly")) {
- return _RECVONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendrecv")) {
- return _SENDRECV;
- }
- pos++;
- }
+ int pos;
+ int pos_media;
+
+ /* test media attributes */
+ pos = 0;
+ while (!osip_list_eol (med->a_attributes, pos)) {
+ sdp_attribute_t *at;
+
+ at = (sdp_attribute_t *) osip_list_get (med->a_attributes, pos);
+ if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendonly")) {
+ return _SENDONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "recvonly")) {
+ return _RECVONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendrecv")) {
+ return _SENDRECV;
+ }
+ pos++;
+ }
- /* test global attributes */
- pos_media = -1;
- pos = 0;
- while (!osip_list_eol (sdp->a_attributes, pos)) {
- sdp_attribute_t *at;
-
- at = (sdp_attribute_t *) osip_list_get (sdp->a_attributes, pos);
- if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendonly")) {
- return _SENDONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "recvonly")) {
- return _RECVONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendrecv")) {
- return _SENDRECV;
- }
- pos++;
+ /* test global attributes */
+ pos_media = -1;
+ pos = 0;
+ while (!osip_list_eol (sdp->a_attributes, pos)) {
+ sdp_attribute_t *at;
+
+ at = (sdp_attribute_t *) osip_list_get (sdp->a_attributes, pos);
+ if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendonly")) {
+ return _SENDONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "recvonly")) {
+ return _RECVONLY;
+ } else if (at->a_att_field != NULL &&
+ 0 == strcmp (at->a_att_field, "sendrecv")) {
+ return _SENDRECV;
+ }
+ pos++;
+ }
+
+ return _SENDRECV;
+}
+
+bool
+SIPCall::setPeerInfoFromRequest(eXosip_event_t *event)
+{
+ // event->request should not be NULL!
+ char remote_uri[256] = "";
+ std::string name("");
+ std::string number("");
+
+ char *tmp = NULL;
+ osip_from_to_str(event->request->from, &tmp);
+ if (tmp != NULL) {
+ snprintf (remote_uri, 255, "%s", tmp);
+ remote_uri[255] = '\0';
+ osip_free (tmp);
+
+ // Get the name/number
+ osip_from_t *from;
+ osip_from_init(&from);
+ osip_from_parse(from, remote_uri);
+ char *tmpname = osip_from_get_displayname(from);
+ if ( tmpname != NULL ) {
+ name = tmpname;
}
+ osip_uri_t* url = osip_from_get_url(from);
+ if ( url != NULL && url->username != NULL) {
+ number = url->username;
+ }
+ osip_from_free(from);
+ }
- return _SENDRECV;
+ _debug(" Name: %s\n", name.c_str());
+ _debug(" Number: %s\n", number.c_str());
+ _debug(" Remote URI: %s\n", remote_uri);
+
+ setPeerName(name);
+ setPeerNumber(number);
+ return true;
}
-void
-SipCall::alloc(void) {
- this->_reason_phrase = new char[50];
- this->_textinfo = new char[256];
- this->_remote_uri = new char[256];
- this->_remote_sdp_audio_ip = new char[50];
-
- // initialize the strings...
- this->_reason_phrase[0] = '\0';
- this->_textinfo[0] = '\0';
- this->_remote_uri[0] = '\0';
- strcpy(this->_remote_sdp_audio_ip, "127.0.0.1");
+sdp_message_t*
+SIPCall::getRemoteSDPFromRequest(eXosip_event_t *event)
+{
+ // event->request should not be null!
+ /* negotiate payloads */
+ sdp_message_t *remote_sdp = NULL;
+ if (event->request != NULL) {
+ eXosip_lock();
+ remote_sdp = eXosip_get_sdp_info (event->request);
+ eXosip_unlock();
+ }
+ if (remote_sdp == NULL) {
+ _debug("SIP Failure: No SDP into the request\n");
+ _debug("< Sending 400 Bad Request (no SDP)\n");
+ eXosip_lock();
+ eXosip_call_send_answer (event->tid, 400, NULL);
+ eXosip_unlock();
+ return 0;
+ }
+ return remote_sdp;
}
-void
-SipCall::dealloc(void) {
- delete [] _reason_phrase; _reason_phrase = NULL;
- delete [] _textinfo; _textinfo = NULL;
- delete [] _remote_uri; _remote_uri = NULL;
- delete [] _remote_sdp_audio_ip; _remote_sdp_audio_ip = NULL;
+sdp_media_t*
+SIPCall::getRemoteMedia(int tid, sdp_message_t* remote_sdp)
+{
+ // Remote Media Port
+ eXosip_lock();
+ sdp_media_t *remote_med = eXosip_get_audio_media(remote_sdp);
+ eXosip_unlock();
+
+ if (remote_med == NULL || remote_med->m_port == NULL) {
+ // no audio media proposed
+ _debug("SIP Failure: unsupported media\n");
+ _debug("< Sending 415 Unsupported media type\n");
+ eXosip_lock();
+ eXosip_call_send_answer (tid, 415, NULL);
+ eXosip_unlock();
+ sdp_message_free (remote_sdp);
+ return 0;
+ }
+ return remote_med;
}
-void
-SipCall::noSupportedCodec (void) {
- _debug("SipCall::noSupportedCodec: Codec no supported\n");
+bool
+SIPCall::setRemoteAudioFromSDP(sdp_media_t* remote_med, sdp_message_t* remote_sdp)
+{
+ // Remote Media IP
+ eXosip_lock();
+ sdp_connection_t *conn = eXosip_get_audio_connection(remote_sdp);
+ eXosip_unlock();
+ if (conn != NULL && conn->c_addr != NULL) {
+ char _remote_sdp_audio_ip[50] = "";
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ _remote_sdp_audio_ip[49] = '\0';
+ _debug(" Remote Audio IP: %s\n", _remote_sdp_audio_ip);
+ setRemoteIP(_remote_sdp_audio_ip);
+ if (_remote_sdp_audio_ip[0] == '\0') {
+ setRemoteAudioPort(0);
+ return false;
+ }
+ }
+
+ // Remote port
+ int _remote_sdp_audio_port = atoi(remote_med->m_port);
+ _debug(" Remote Audio Port: %d\n", _remote_sdp_audio_port);
+ setRemoteAudioPort(_remote_sdp_audio_port);
+
+ if (_remote_sdp_audio_port == 0) {
+ return false;
+ }
+ return true;
}
+bool
+SIPCall::setAudioCodecFromSDP(sdp_media_t* remote_med, int tid)
+{
+ // Remote Payload
+ char *tmp = NULL;
+ int pos = 0;
+ while (!osip_list_eol (remote_med->m_payloads, pos)) {
+ tmp = (char *) osip_list_get (remote_med->m_payloads, pos);
+ if (tmp != NULL ) {
+ int payload = atoi(tmp);
+ // stop if we find a correct codec
+ if (0 != _codecMap.getCodec((CodecType)payload)){
+ break;
+ }
+ }
+ tmp = NULL;
+ pos++;
+ }
+
+ setAudioCodec(0);
+ if (tmp != NULL) {
+ int payload = atoi (tmp);
+ _debug(" Payload: %d\n", payload);
+ setAudioCodec(_codecMap.getCodec((CodecType)payload)); // codec builder for the mic
+ }
+ if (getAudioCodec() == 0) {
+ _debug("SIPCall Failure: Unable to set codec\n");
+ _debug("< Sending 415 Unsupported media type\n");
+ eXosip_lock();
+ eXosip_call_send_answer(tid, 415, NULL);
+ eXosip_unlock();
+ return false;
+ }
+ return true;
+}
diff --git a/src/sipcall.h b/src/sipcall.h
index bf9c3ab195abc063853bad1d5e8d9d2dee4ce2a4..cbe640a055f25aea066e14bedd060e2df4adb50b 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -1,155 +1,181 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
- * Portions Copyright (C) 2002,2003 Aymeric Moizard <jack@atosc.org>
+/*
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
+ * Author: Yan Morin <yan.morin@savoirfairelinux.com>
+ * Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
- * This is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as
- * published by the Free Software Foundation; either version 2,
- * or (at your option) any later version.
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
*
- * This is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
*
- * You should have received a copy of the GNU General Public
- * License along with dpkg; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
+#ifndef SIPCALL_H
+#define SIPCALL_H
-#ifndef __SIP_CALL_H__
-#define __SIP_CALL_H__
-
-#include <eXosip2/eXosip.h>
-#include <vector>
-#include <string>
#include "call.h"
+#include "audio/codecDescriptor.h"
+#include <eXosip2/eXosip.h>
-class CodecDescriptor;
class AudioCodec;
-#define _SENDRECV 0
-#define _SENDONLY 1
-#define _RECVONLY 2
-
-// Vector of CodecDescriptor
-class SipCall {
+/**
+ @author Yan Morin <yan.morin@gmail.com>
+ SIPCall are SIP implementation of a normal Call
+*/
+class SIPCall : public Call
+{
public:
- SipCall(CALLID id, CodecDescriptorMap& codecMap);
- ~SipCall (void);
+ SIPCall(const CallID& id, Call::CallType type);
+
+ ~SIPCall();
+
+ /** @return SIP call id : protected by eXosip lock */
+ int getCid() { return _cid; }
+ /** @param cid SIP call id : protected by eXosip lock */
+ void setCid(int cid) { _cid = cid ; }
+ /** @return SIP domain id : protected by eXosip lock */
+ int getDid() { return _did; }
+ /** @param did SIP domain id : protected by eXosip lock */
+ void setDid(int did) { _did = did; }
+ /** @return SIP transaction id : protected by eXosip lock */
+ int getTid() { return _tid; }
+ /** @param did SIP transaction id : protected by eXosip lock */
+ void setTid(int tid) { _tid = tid; }
+
+ // AUDIO
+ /** Set internal codec Map: initialization only, not protected */
+ void setCodecMap(const CodecDescriptorMap& map) { _codecMap = map; }
+ CodecDescriptorMap& getCodecMap();
+
+ /** set internal, not protected */
+ void setLocalIp(const std::string& ip) { _localIPAddress = ip; }
+ void setLocalAudioPort(unsigned int port) { _localAudioPort = port;}
+ void setLocalExternAudioPort(unsigned int port) { _localExternalAudioPort = port; }
+ unsigned int getLocalExternAudioPort() { return _localExternalAudioPort; }
/**
- * Store information about incoming call and negociate payload
+ * Answer incoming call correclty before telling the user
+ * @param event eXosip Event
*/
- int newIncomingCall(eXosip_event_t *);
- int newReinviteCall(eXosip_event_t *);
-
- /*
- * Use to answer to a ONHOLD/OFFHOLD event
- */
- int answeredCall (eXosip_event_t *);
-
- /*
- * Use to answer to an incoming call
- */
- void answeredCall_without_hold (eXosip_event_t *);
-
- int ringingCall (eXosip_event_t *);
- int receivedAck (eXosip_event_t *);
-
- /*
- * Manage local audio port for each sipcall
- */
- void setLocalAudioPort (int);
- int getLocalAudioPort (void);
-
- std::string getLocalIp() { return _localIp; }
- void setLocalIp(const std::string& ip) { _localIp = ip; }
-
- bool isReinvite() { return _reinvite; }
- void endReinvite() { _reinvite = false; };
-
- /*
- * Manage id, did (dialog-id), cid (call-id) and tid (transaction-id)
- * for each sipcall
- */
- void setId (CALLID id);
- CALLID getId (void);
- void setDid (int did);
- int getDid (void);
- void setCid (int cid);
- int getCid (void);
- void setTid (int tid);
- int getTid (void);
-
- /*
- * Manage remote sdp audio port
- */
- int getRemoteSdpAudioPort (void);
- char* getRemoteSdpAudioIp (void);
-
- /*
- * Manage audio codec
+ bool SIPCallInvite(eXosip_event_t *event);
+
+ /**
+ * newReinviteCall is called when the IP-Phone user receives a change in the call
+ * it's almost an newIncomingCall but we send a 200 OK
+ * See: 3.7. Session with re-INVITE (IP Address Change)
+ * @param event eXosip Event
+ * @return true if ok
*/
- AudioCodec* getAudioCodec(void) { return _audiocodec; }
- void setAudioCodec(AudioCodec* ac) { _audiocodec = ac; }
- CodecDescriptorMap& getCodecMap() { return _codecMap; }
+ bool SIPCallReinvite(eXosip_event_t *event);
- /*
- * Accessor to remote-uri
+ /**
+ * Peer answered to a call (on hold or not)
+ * @param event eXosip Event
+ * @return true if ok
+ */
+ bool SIPCallAnswered(eXosip_event_t *event);
+ /**
+ * We retreive final SDP info if they changed
+ * @param event eXosip Event
+ * @return true if ok (change / no change) or false on error
*/
- inline char* getRemoteUri (void) { return _remote_uri; }
+ bool SIPCallAnsweredWithoutHold(eXosip_event_t *event);
- /*
- * To avoid confusion when an incoming call occured in the same time
- * that you make an outgoing call
+ /** protected */
+ const std::string& getLocalIp();
+ unsigned int getLocalAudioPort();
+ unsigned int getRemoteAudioPort();
+ const std::string& getRemoteIp();
+ AudioCodec* getAudioCodec();
+
+ /**
+ * Set the audio start boolean (protected by mutex)
+ * @param start true if we start the audio
+ */
+ void setAudioStart(bool start);
+ /**
+ * Tell if the audio is started (protected by mutex)
+ * @return true if it's already started
*/
- inline void setStandBy (bool standby) { _standby = standby; }
- inline bool getStandBy (void) { return _standby; }
+ bool isAudioStarted();
- std::string getName() const { return _name; }
- std::string getNumber() const { return _number; }
+ //TODO: humm?
int sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg);
+
private:
- void alloc (void);
- void dealloc (void);
- void noSupportedCodec(void);
+ /** set internal, not protected */
+ void setRemoteIP(const std::string& ip) { _remoteIPAddress = ip; }
+ void setRemoteAudioPort(unsigned int port) { _remoteAudioPort = port; }
+ void setAudioCodec(AudioCodec* audioCodec) { _audioCodec = audioCodec; }
+ // TODO: hum???
int sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med);
-
- ///////////////////////////
- // Private member variables
- ///////////////////////////
- CodecDescriptorMap& _codecMap;
- AudioCodec* _audiocodec;
-
- CALLID _id;
- int _cid; // call id
- int _did; // dialog id
- int _tid; // transaction id
- bool _standby; // wait for a cid and did when outgoing call is made
-
- int _status_code;
-
- char* _reason_phrase;
- char* _textinfo;
- char* _remote_uri;
-
- char* _remote_sdp_audio_ip;
- int _local_audio_port;
- int _remote_sdp_audio_port;
- int _local_sendrecv; /* _SENDRECV, _SENDONLY, _RECVONLY */
- int _remote_sendrecv; /* _SENDRECV, _SENDONLY, _RECVONLY */
-
- std::string _localIp;
- std::string _name; // set by incoming call
- std::string _number; // set by incoming call
-
- bool _reinvite;
+ /**
+ * Set peer name and number with event->request->from
+ * @param event eXosip event
+ * @return false the event is invalid
+ */
+ bool setPeerInfoFromRequest(eXosip_event_t *event);
+ /**
+ * Get a valid remote SDP or return a 400 bad request response if invalid
+ * @param event eXosip event
+ * @return valid remote_sdp or 0
+ */
+ sdp_message_t* getRemoteSDPFromRequest(eXosip_event_t *event);
+
+ /**
+ * Get a valid remote media or return a 415 unsupported media type
+ * @param tid transaction id
+ * @param remote_sdp Remote SDP pointer
+ * @return valid sdp_media_t or 0
+ */
+ sdp_media_t* getRemoteMedia(int tid, sdp_message_t* remote_sdp);
+
+ /**
+ * Set Audio Port and Audio IP from Remote SDP Info
+ * @param remote_med Remote Media info
+ * @param remote_sdp Remote SDP pointer
+ * @return true if everything is set correctly
+ */
+ bool setRemoteAudioFromSDP(sdp_media_t* remote_med, sdp_message_t* remote_sdp);
+
+ /**
+ * Set Audio Codec with the remote choice
+ * @param remote_med Remote Media info
+ * @return true if everything is set correctly
+ */
+ bool setAudioCodecFromSDP(sdp_media_t* remote_med, int tid);
+
+
+ /** SIP call id */
+ int _cid;
+ /** SIP domain id */
+ int _did;
+ /** SIP transaction id */
+ int _tid;
+
+ /** Codec Map */
+ CodecDescriptorMap _codecMap;
+ /** codec pointer */
+ AudioCodec* _audioCodec;
+ bool _audioStarted;
+
+ // Informations about call socket / audio
+ std::string _localIPAddress;
+ unsigned int _localAudioPort;
+ unsigned int _localExternalAudioPort; // what peer (NAT) should connect to
+
+ std::string _remoteIPAddress;
+ unsigned int _remoteAudioPort;
};
-#endif // __SIP_CALL_H__
+#endif
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 314f462852e1e8d1f12e9413d643e83a40f623de..9214b097222dac6b357745f4fa36f3e583fcb7f5 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -1,10 +1,8 @@
/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
- * Portions Copyright (C) 2002,2003 Aymeric Moizard <jack@atosc.org>
- *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
@@ -19,564 +17,686 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include <eXosip2/eXosip.h>
-#include <osip2/osip.h>
-#include <osipparser2/osip_const.h>
-
#include "sipvoiplink.h"
-#include "global.h"
-#include "audio/codecDescriptor.h"
-#include "manager.h"
-#include "sipcall.h"
-#include "user_cfg.h"
#include "eventthread.h"
+#include "sipcall.h"
+#include <sstream> // for ostringstream
+
+#include "manager.h"
+#include "user_cfg.h" // SIGNALISATION / PULSE #define
-#define DEFAULT_SIP_PORT 5060
-#define RANDOM_SIP_PORT rand() % 64000 + 1024
-#define DEFAULT_LOCAL_PORT 10500
-#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
+// for listener
+#define DEFAULT_SIP_PORT 5060
+#define RANDOM_SIP_PORT rand() % 64000 + 1024
+#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
#define EXOSIP_ERROR_NO 0
#define EXOSIP_ERROR_STD -1
#define EXOSIP_ERROR_BUILDING -2
-#define VOICE_MSG "Voice-Message"
-#define LENGTH_VOICE_MSG 15
+// for registration
+#define EXPIRES_VALUE 180
+
+// 1XX responses
+#define DIALOG_ESTABLISHED 101
+// see: osip_const.h
-SipVoIPLink::SipVoIPLink() : VoIPLink()
+// FOR VOICE Message handling
+#define VOICE_MSG "Voice-Message"
+#define LENGTH_VOICE_MSG 15
+
+// need for hold/unhold
+#define INVITE_METHOD "INVITE"
+
+
+SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
+ : VoIPLink(accountID), _localExternAddress("")
{
- // default _audioRTP object initialization
_evThread = new EventThread(this);
- _localPort = 0;
+
_nMsgVoicemail = 0;
- _reg_id = EXOSIP_ERROR_STD;
+ _eXosipRegID = EXOSIP_ERROR_STD;
+ _eXosipStarted = false;
+
+ _nbTryListenAddr = 2; // number of times to try to start SIP listener
+ _localExternPort = 0;
+
+ // to get random number for RANDOM_PORT
+ srand (time(NULL));
+}
- _registrationSend = false;
- _started = false;
- _localIpAddress = "127.0.0.1"; // sipvoip require this value to check network
+SIPVoIPLink::~SIPVoIPLink()
+{
+ delete _evThread; _evThread = 0;
+ terminate();
+}
+
+bool
+SIPVoIPLink::init()
+{
+ if (!_eXosipStarted) {
+ if (0 != eXosip_init()) {
+ _debug("Could not initialize eXosip\n");
+ return false;
+ }
+ _eXosipStarted = true;
+
+ // check networking capabilities
+ if ( !checkNetwork() ) {
+ _debug("SIP FAILURE: Unable to determine network capabilities\n");
+ return false;
+ }
+
+ // if we useStun and we failed to receive something on port 5060, we try a random port
+ // If use STUN server, firewall address setup
+ int errExosip = 0;
+ int port = DEFAULT_SIP_PORT;
+
+ int iTry = 1; // try number..
+
+ do {
+ if (_useStun && !Manager::instance().behindNat(port)) {
+ port = RANDOM_SIP_PORT;
+ if (!Manager::instance().behindNat(port)) {
+ _debug("SIP FAILURE: Unable to check NAT setting\n");
+ return false; // hoho we can't use the random sip port too...
+ }
+ }
+
+ // second parameter, NULL is "::" for ipv6 and "0.0.0.0" for ipv4, we can put INADDR_ANY
+ errExosip = eXosip_listen_addr(IPPROTO_UDP, INADDR_ANY, port, AF_INET, 0);
+ if (errExosip != 0) {
+ _debug("SIP ERROR: [%d/%d] could not initialize SIP listener on port %d\n", iTry, _nbTryListenAddr, port);
+ port = RANDOM_SIP_PORT;
+ }
+ } while ( errExosip != 0 && iTry < _nbTryListenAddr );
+
+ if ( errExosip != 0 ) { // we didn't succeeded
+ _debug("SIP FAILURE: SIP failed to listen on port %d\n", port);
+ return false;
+ }
+ _localPort = port;
+ _debug("SIP Init: listening on port %d\n", port);
+
+ if (_useStun) {
+ // This method is used to replace contact address with the public address of your NAT
+ // it should be call after eXosip_listen_addr
+ // set by last behindNat() call (ish)...
+ _localExternAddress = Manager::instance().getFirewallAddress();
+ _localExternPort = Manager::instance().getFirewallPort();
+ eXosip_masquerade_contact(_localExternAddress.data(), _localExternPort);
+ } else {
+ _localExternAddress = _localIPAddress;
+ _localExternPort = _localPort;
+ }
+
+ // Set user agent
+ std::string tmp = std::string(PROGNAME_GLOBAL) + "/" + std::string(SFLPHONED_VERSION);
+ eXosip_set_user_agent(tmp.data());
+
+ _debug("SIP Init: starting loop thread (SIP events)\n");
+ _evThread->start();
+ }
+ return _eXosipStarted;
}
-SipVoIPLink::~SipVoIPLink(void) {
- endSipCalls();
- delete _evThread; _evThread = NULL;
- if (_started) {
+
+
+void
+SIPVoIPLink::terminate()
+{
+ terminateSIPCall();
+ if (_eXosipStarted) {
eXosip_quit();
+ _eXosipStarted = false;
}
}
-// for voIPLink interface
void
-SipVoIPLink::terminate(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) {
+ // Release SIP stack.
+ eXosip_lock();
+ eXosip_call_terminate(call->getCid(), call->getDid() );
+ eXosip_unlock();
+ delete call; call = 0;
+ }
+ iter++;
+ }
+ _callMap.clear();
}
bool
-SipVoIPLink::checkNetwork (void)
+SIPVoIPLink::checkNetwork()
{
// Set IP address
- return getSipLocalIp();
+ return loadSIPLocalIP();
}
-
-/**
- * Steps:
- * 1. Init eXosip
- * 2. Try to listen two times on a port
- * if we use stun, we check if we can use 5060, before connecting to eXosip...
- * if we can't we check on a random port
- */
bool
-SipVoIPLink::init(void)
+SIPVoIPLink::loadSIPLocalIP()
{
- // TODO: should be inside the account
- _fullname = Manager::instance().getConfigString(SIGNALISATION,FULL_NAME) ;
- _hostname = Manager::instance().getConfigString(SIGNALISATION,HOST_PART);
-
- if (0 != eXosip_init()) {
- _debug("Could not initialize eXosip\n");
- return false;
- }
- _started = true;
-
- srand (time(NULL));
- int i;
-
- // check networking capabilities
- if ( !checkNetwork() ) {
- return false;
- }
-
- // if we useStun and we failed to receive something on port 5060, we try a random port
- // If use STUN server, firewall address setup
- bool useStun = Manager::instance().useStun();
- int port = DEFAULT_SIP_PORT;
-
- int nbTry = 2; // number of times to try to start SIP listener
- int iTry = 1; // try number..
-
- do {
- if (useStun && !behindNat(port)) {
- port = RANDOM_SIP_PORT;
- if (!behindNat(port)) {
- return false; // hoho we can't use the random sip port too...
- }
- }
-
- // second parameter, NULL is "::" for ipv6 and "0.0.0.0" for ipv4, we can put INADDR_ANY
- i = eXosip_listen_addr(IPPROTO_UDP, INADDR_ANY, port, AF_INET, 0);
- if (i != 0) {
- _debug("SIP ERROR: [%d/%d] could not initialize SIP listener on port %d\n", iTry, nbTry, port);
- port = RANDOM_SIP_PORT;
+ bool returnValue = true;
+ if (_localIPAddress == "127.0.0.1") {
+ char* myIPAddress = new char[65];
+ if (eXosip_guess_localip(AF_INET, myIPAddress, 64) == EXOSIP_ERROR_STD) {
+ returnValue = false;
+ } else {
+ _localIPAddress = std::string(myIPAddress);
+ _debug("Checking network, setting local IP address to: %s\n", myIPAddress);
}
- } while ( i != 0 && iTry < nbTry );
-
- if ( i != 0 ) { // we didn't succeeded
- _debug("SIP FAILURE: SIP failed to listen on port %d\n", port);
- return false;
- }
- _debug("SIP Init: listening on port %d\n", port);
-
- if (useStun) {
- // This method is used to replace contact address with the public address of your NAT
- // it should be call after eXosip_listen_addr
- eXosip_masquerade_contact((Manager::instance().getFirewallAddress()).data(), Manager::instance().getFirewallPort());
+ delete [] myIPAddress; myIPAddress = NULL;
}
-
- // Set user agent
- std::string tmp = std::string(PROGNAME_GLOBAL) + "/" + std::string(SFLPHONED_VERSION);
- eXosip_set_user_agent(tmp.data());
-
- _debug("SIP Init: starting loop thread (SIP events)\n");
- _evThread->start();
- return true;
+ return returnValue;
}
-/**
- * Subscibe to message-summary notify
- * It allows eXosip to not send ' 481 Subcription Does Not Exist ' response
- */
void
-SipVoIPLink::subscribeMessageSummary()
+SIPVoIPLink::getEvent()
{
- osip_message_t *subscribe;
- const char *route= NULL;
-
- // from/to
- ManagerImpl& manager = Manager::instance();
- std::string from = fromHeader(manager.getConfigString(SIGNALISATION, USER_PART), manager.getConfigString(SIGNALISATION, HOST_PART));
-
- // to
- std::string to;
- to = manager.getConfigString(SIGNALISATION, PROXY);
- if (!to.empty()) {
- to = toHeader(manager.getConfigString(SIGNALISATION, USER_PART)) + "@" + to;
- } else {
- to = from;
- }
-
-
- // like in http://www.faqs.org/rfcs/rfc3842.html
- const char *event="message-summary";
- int expires = 86400;
-
- // return 0 if no error
- // the first from is the to... but we send the same
+ eXosip_event_t* event = eXosip_event_wait(0, 50);
eXosip_lock();
- int error = eXosip_subscribe_build_initial_request(&subscribe, to.c_str(), from.c_str(), route, event, expires);
+ eXosip_automatic_action();
eXosip_unlock();
- if (error == 0) {
- // Accept: application/simple-message-summary
- osip_message_set_header (subscribe, "Accept", "application/simple-message-summary");
-
- _debug("Sending Message-summary subscription");
- // return 0 if ok
- eXosip_lock();
- error = eXosip_subscribe_send_initial_request (subscribe);
- eXosip_unlock();
- _debug(" and return %d\n", error);
+ if (event == NULL) {
+ return;
}
+
+ _debug("SIP Event: [cdt=%4d:%4d:%4d] type=#%03d %s \n", event->cid, event->did, event->tid, event->type, event->textinfo);
+ switch (event->type) {
+ /* REGISTER related events */
+ case EXOSIP_REGISTRATION_NEW: /** 00 < announce new registration. */
+ _debug("EXOSIP_REGISTRATION_NEW event is not implemented\n");
+ break;
+ case EXOSIP_REGISTRATION_SUCCESS: /** 01 < user is successfully registred. */
+ Manager::instance().registrationSucceed(getAccountID());
+ break;
+ case EXOSIP_REGISTRATION_FAILURE: /** 02 < user is not registred. */
+ Manager::instance().registrationFailed(getAccountID());
+ break;
+ case EXOSIP_REGISTRATION_REFRESHED: /** 03 < registration has been refreshed. */
+ _debug("EXOSIP_REGISTRATION_REFRESHED event is not implemented\n");
+ break;
+ case EXOSIP_REGISTRATION_TERMINATED: /** 04 < UA is not registred any more. */
+ _debug("EXOSIP_REGISTRATION_TERMINATED event is not implmeneted\n");
+ break;
+
+ /* INVITE related events within calls */
+ case EXOSIP_CALL_INVITE: /** 05 < announce a new call */
+ SIPCallInvite(event);
+ break;
+ case EXOSIP_CALL_REINVITE: /** 06 < announce a new INVITE within call */
+ SIPCallReinvite(event);
+ break;
+
+ case EXOSIP_CALL_NOANSWER: /** 07 < announce no answer within the timeout */
+ _debug("EXOSIP_CALL_NOANSWER event is not implemented\n");
+ break;
+ case EXOSIP_CALL_PROCEEDING: /** 08 < announce processing by a remote app */
+ _debug("EXOSIP_CALL_NOANSWER event is not implemented\n");
+ break;
+ case EXOSIP_CALL_RINGING: /** 09 < announce ringback */
+ SIPCallRinging(event);
+ break;
+ case EXOSIP_CALL_ANSWERED: /** 10 < announce start of call */
+ SIPCallAnswered(event);
+ break;
+ case EXOSIP_CALL_REDIRECTED: /** 11 < announce a redirection */
+ _debug("EXOSIP_CALL_REDIRECTED event is not implemented\n");
+ break;
+ case EXOSIP_CALL_REQUESTFAILURE: /** 12 < announce a request failure */
+ SIPCallRequestFailure(event);
+ break;
+ case EXOSIP_CALL_SERVERFAILURE: /** 13 < announce a server failure */
+ SIPCallServerFailure(event);
+ break;
+ case EXOSIP_CALL_GLOBALFAILURE: /** 14 < announce a global failure */
+ SIPCallServerFailure(event);
+ break;
+ case EXOSIP_CALL_ACK: /** 15 < ACK received for 200ok to INVITE */
+ SIPCallAck(event);
+ break;
+
+ case EXOSIP_CALL_CANCELLED: /** 16 < announce that call has been cancelled */
+ case EXOSIP_CALL_TIMEOUT: /** 17 < announce that call has failed */
+ Manager::instance().displayError("EXOSIP Call Error not implemented yet");
+ break;
+
+ /* request related events within calls (except INVITE) */
+ case EXOSIP_CALL_MESSAGE_NEW: /** 18 < announce new incoming MESSAGE. */
+ SIPCallMessageNew(event);
+ break;
+ case EXOSIP_CALL_MESSAGE_PROCEEDING: /** 19 < announce a 1xx for MESSAGE. */
+ case EXOSIP_CALL_MESSAGE_ANSWERED: /** 20 < announce a 200ok */
+ // 200 OK
+ case EXOSIP_CALL_MESSAGE_REDIRECTED: /** 21 < announce a failure. */
+ case EXOSIP_CALL_MESSAGE_REQUESTFAILURE: /** 22 < announce a failure. */
+ case EXOSIP_CALL_MESSAGE_SERVERFAILURE: /** 23 < announce a failure. */
+ case EXOSIP_CALL_MESSAGE_GLOBALFAILURE: /** 24 < announce a failure. */
+ Manager::instance().displayError("EXOSIP Call Message not implemented yet");
+ break;
+
+ case EXOSIP_CALL_CLOSED: /** 25 < a BYE was received for this call */
+ SIPCallClosed(event);
+ break;
+
+ /* for both UAS & UAC events */
+ case EXOSIP_CALL_RELEASED: /** 26 < call context is cleared. */
+ SIPCallReleased(event);
+ break;
+
+ /* response received for request outside calls */
+ case EXOSIP_MESSAGE_NEW: /** 27 < announce new incoming MESSAGE. */
+ if (event->request == NULL) { break; }
+ SIPMessageNew(event);
+ break;
+ case EXOSIP_MESSAGE_PROCEEDING: /** 28 < announce a 1xx for MESSAGE. */
+ case EXOSIP_MESSAGE_ANSWERED: /** 29 < announce a 200ok */
+ case EXOSIP_MESSAGE_REDIRECTED: /** 30 < announce a failure. */
+ Manager::instance().displayError("EXOSIP Message not implemented yet");
+ break;
+
+ case EXOSIP_MESSAGE_REQUESTFAILURE: /** 31 < announce a failure. */
+ if (event->response !=0 && event->response->status_code == SIP_METHOD_NOT_ALLOWED) {
+ Manager::instance().incomingMessage(getAccountID(), "Message are not allowed");
+ } else {
+ Manager::instance().displayError("EXOSIP_MESSAGE_REQUESTFAILURE not implemented yet");
+ }
+ break;
+ case EXOSIP_MESSAGE_SERVERFAILURE: /** 32 < announce a failure. */
+ case EXOSIP_MESSAGE_GLOBALFAILURE: /** 33 < announce a failure. */
+ Manager::instance().displayError("EXOSIP Message not implemented yet");
+ break;
+
+ /* Presence and Instant Messaging */
+ case EXOSIP_SUBSCRIPTION_UPDATE: /** 34 < announce incoming SUBSCRIBE. */
+ case EXOSIP_SUBSCRIPTION_CLOSED: /** 35 < announce end of subscription. */
+ Manager::instance().displayError("EXOSIP Subscription not implemented yet");
+ break;
+
+ case EXOSIP_SUBSCRIPTION_NOANSWER: /** 37 < announce no answer */
+ case EXOSIP_SUBSCRIPTION_PROCEEDING: /** 38 < announce a 1xx */
+ Manager::instance().displayError("EXOSIP Subscription resposne not implemented yet");
+ break;
+ case EXOSIP_SUBSCRIPTION_ANSWERED: /** 39 < announce a 200ok */
+ eXosip_lock();
+ eXosip_automatic_action();
+ eXosip_unlock();
+ break;
+
+ case EXOSIP_SUBSCRIPTION_REDIRECTED: /** 40 < announce a redirection */
+ case EXOSIP_SUBSCRIPTION_REQUESTFAILURE: /** 41 < announce a request failure */
+ case EXOSIP_SUBSCRIPTION_SERVERFAILURE: /** 42 < announce a server failure */
+ case EXOSIP_SUBSCRIPTION_GLOBALFAILURE: /** 43 < announce a global failure */
+ case EXOSIP_SUBSCRIPTION_NOTIFY: /** 44 < announce new NOTIFY request */
+ case EXOSIP_SUBSCRIPTION_RELEASED: /** 45 < call context is cleared. */
+ Manager::instance().displayError("EXOSIP Subscription resposne not implemented yet");
+ break;
+
+ case EXOSIP_IN_SUBSCRIPTION_NEW: /** 46 < announce new incoming SUBSCRIBE.*/
+ case EXOSIP_IN_SUBSCRIPTION_RELEASED: /** 47 < announce end of subscription. */
+ Manager::instance().displayError("EXOSIP Subscription not implemented yet");
+ break;
+
+ case EXOSIP_EVENT_COUNT: /** 48 < MAX number of events */
+ break;
+ }
+ eXosip_event_free(event);
}
-int
-SipVoIPLink::setRegister (void)
+bool
+SIPVoIPLink::setRegister()
{
- ManagerImpl& manager = Manager::instance();
-
- if (_reg_id != EXOSIP_ERROR_STD) {
- manager.displayError("Registration already sent. Try to unregister");
- return EXOSIP_ERROR_STD;
+ if (_eXosipRegID != EXOSIP_ERROR_STD) {
+ Manager::instance().displayError("SIP Error: Registration already sent. Try to unregister");
+ return false;
}
- // all this will be inside the profil associate with the voip link
- std::string proxy = "sip:" + manager.getConfigString(SIGNALISATION, PROXY);
- std::string hostname = "sip:" + manager.getConfigString(SIGNALISATION, HOST_PART);
- std::string from = fromHeader(manager.getConfigString(SIGNALISATION, USER_PART), manager.getConfigString(SIGNALISATION, HOST_PART));
-
- if (manager.getConfigString(SIGNALISATION, HOST_PART).empty()) {
- manager.displayConfigError("Fill host part field");
- return EXOSIP_ERROR_STD;
+ std::string hostname = getHostName();
+ if (hostname.empty()) {
+ Manager::instance().displayConfigError("Fill host part field");
+ return false;
}
- if (manager.getConfigString(SIGNALISATION, USER_PART).empty()) {
- manager.displayConfigError("Fill user part field");
- return EXOSIP_ERROR_STD;
+
+ if (_userpart.empty()) {
+ Manager::instance().displayConfigError("Fill user part field");
+ return false;
}
+
+ std::string proxy = "sip:" + _proxy;
+ hostname = "sip:" + hostname;
+ std::string from = SIPFromHeader(_userpart, getHostName());
+
osip_message_t *reg = NULL;
eXosip_lock();
- if (!manager.getConfigString(SIGNALISATION, PROXY).empty()) {
- _debug("REGISTER From: %s to %s\n", from.data(), proxy.data());
- _reg_id = eXosip_register_build_initial_register ((char*)from.data(),
- (char*)proxy.data(), NULL, EXPIRES_VALUE, ®);
+ if (!_proxy.empty()) {
+ _debug("SIP Register: From: %s to %s\n", from.data(), proxy.data());
+ _eXosipRegID = eXosip_register_build_initial_register(from.data(),
+ proxy.data(), NULL, EXPIRES_VALUE, ®);
} else {
- _debug("REGISTER From: %s to %s\n", from.data(), hostname.data());
- _reg_id = eXosip_register_build_initial_register ((char*)from.data(),
- (char*)hostname.data(), NULL, EXPIRES_VALUE, ®);
+ _debug("SIP Register: From: %s to %s\n", from.data(), hostname.data());
+ _eXosipRegID = eXosip_register_build_initial_register(from.data(),
+ hostname.data(), NULL, EXPIRES_VALUE, ®);
}
eXosip_unlock();
- if (_reg_id < EXOSIP_ERROR_NO ) {
- return EXOSIP_ERROR_STD;
+ if (_eXosipRegID < EXOSIP_ERROR_NO ) {
+ return false;
}
- if (setAuthentication() != EXOSIP_ERROR_NO) {
- _debug("No authentication\n");
- return EXOSIP_ERROR_STD;
+ if (!sendSIPAuthentification()) {
+ _debug("SIP Register: No authentication\n");
+ return false;
}
osip_message_set_header (reg, "Event", "Registration");
osip_message_set_header (reg, "Allow-Events", "presence");
eXosip_lock();
- int i = eXosip_register_send_register (_reg_id, reg);
- if (i == -2) {
- _debug("Cannot build registration, check the setup\n");
+ int eXosipErr = eXosip_register_send_register(_eXosipRegID, reg);
+ if (eXosipErr == EXOSIP_ERROR_BUILDING) {
+ _debug("SIP Failure: Cannot build registration, check the setup\n");
eXosip_unlock();
- return EXOSIP_ERROR_STD;
+ return false;
}
- if (i == -1) {
- _debug("Registration sending failed\n");
+ if (eXosipErr == EXOSIP_ERROR_STD) {
+ _debug("SIP Failure: Registration sending failed\n");
eXosip_unlock();
- return EXOSIP_ERROR_STD;
+ return false;
}
eXosip_unlock();
- // subscribe to message one time?
- // subscribeMessageSummary();
- _registrationSend = true;
- return i;
+ return true;
}
-/**
- * setUnregister
- * unregister if we already send the first registration
- * @return -1 if there is an error
- */
-int
-SipVoIPLink::setUnregister (void)
+std::string
+SIPVoIPLink::SIPFromHeader(const std::string& userpart, const std::string& hostpart)
{
- if ( _registrationSend ) {
- int i = 0;
- osip_message_t *reg = NULL;
-
- eXosip_lock();
-
- if (_reg_id > 0) {
- i = eXosip_register_build_register (_reg_id, 0, ®);
- }
- eXosip_unlock();
- if (i < 0) {
- return EXOSIP_ERROR_STD;
- }
-
- eXosip_lock();
- _debug("< Sending REGISTER (expire=0)\n");
- i = eXosip_register_send_register (_reg_id, reg);
- if (i == EXOSIP_ERROR_BUILDING) {
- _debug(" Cannot build registration (unregister), check the setup\n");
- eXosip_unlock();
- return EXOSIP_ERROR_STD;
- }
- if (i == EXOSIP_ERROR_STD) {
- _debug(" Registration (unregister) Failed\n");
- }
- eXosip_unlock();
- _reg_id = EXOSIP_ERROR_STD;
- return i;
- } else {
- // no registration send before
- return EXOSIP_ERROR_STD;
- }
+ return ("\"" + getFullName() + "\"" + " <sip:" + userpart + "@" + hostpart + ">");
}
-/**
- * Get the Sip FROM url (add sip:, add @host, etc...)
- */
-std::string
-SipVoIPLink::getSipFrom() {
-
- // Form the From header field basis on configuration panel
- std::string user = Manager::instance().getConfigString(SIGNALISATION, USER_PART);
- std::string host = Manager::instance().getConfigString(SIGNALISATION, HOST_PART);
- if ( host.empty() ) {
- host = getLocalIpAddress();
+bool
+SIPVoIPLink::sendSIPAuthentification()
+{
+ std::string login = _authname;
+ if (login.empty()) {
+ login = _userpart;
}
- return fromHeader(user, host);
+ if (login.empty()) {
+ Manager::instance().displayConfigError("Fill authentification name");
+ return false;
+ }
+ if (_password.empty()) {
+ Manager::instance().displayConfigError("Fill password field");
+ return false;
+ }
+ eXosip_lock();
+ int returnValue = eXosip_add_authentication_info(login.data(), login.data(), _password.data(), NULL, NULL);
+ eXosip_unlock();
+
+ return (returnValue != EXOSIP_ERROR_STD ? true : false);
}
-/**
- * Get the Sip TO url (add sip:, add @host, etc...)
- */
-std::string
-SipVoIPLink::getSipTo(const std::string& to_url) {
- // Form the From header field basis on configuration panel
- bool isRegistered = (_reg_id < EXOSIP_ERROR_NO) ? false : true;
+bool
+SIPVoIPLink::setUnregister()
+{
+ if ( _eXosipRegID == EXOSIP_ERROR_STD) return false;
+ int eXosipErr = EXOSIP_ERROR_NO;
+ osip_message_t *reg = NULL;
- // 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 = Manager::instance().getConfigString(SIGNALISATION, HOST_PART);
- if(!host.empty()) {
- return toHeader(to_url + "@" + host);
- }
+ eXosip_lock();
+ eXosipErr = eXosip_register_build_register (_eXosipRegID, 0, ®);
+ eXosip_unlock();
+
+ if (eXosipErr != EXOSIP_ERROR_NO) {
+ _debug("SIP Failure: Unable to build registration for setUnregister");
+ return false;
}
- return toHeader(to_url);
-}
-/**
- * Get the sip proxy (add sip: if there is one)
- * @return empty string or <sip:proxy;lr> url
- */
-std::string
-SipVoIPLink::getSipRoute() {
- std::string proxy = Manager::instance().getConfigString(SIGNALISATION, PROXY);
- if ( !proxy.empty() ) {
- proxy = "<sip:" + proxy + ";lr>";
+ eXosip_lock();
+ _debug("< Sending REGISTER (expire=0)\n");
+ eXosipErr = eXosip_register_send_register (_eXosipRegID, reg);
+ if (eXosipErr == EXOSIP_ERROR_BUILDING) {
+ _debug("SIP Failure: Cannot build registration (unregister), check the setup\n");
+ eXosip_unlock();
+ return false;
}
- return proxy; // return empty
+ if (eXosipErr == EXOSIP_ERROR_STD) {
+ _debug("SIP Failure: Unable to send registration (unregister)\n");
+ }
+ eXosip_unlock();
+ _eXosipRegID = EXOSIP_ERROR_STD;
+
+ return true;
}
-int
-SipVoIPLink::outgoingInvite (CALLID id, const std::string& to_url)
+Call*
+SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
- bool has_ip = checkNetwork();
- std::string from = getSipFrom();
- std::string to = getSipTo(to_url);
- std::string route = getSipRoute();
-
- _debug(" From: %s\n", from.data());
- _debug(" To: %s\n", to.data());
- _debug(" Route: %s\n", route.data());
-
- // If no SIP proxy setting for direct call with only IP address
- if (has_ip) {
- if (startCall(id, from, to, "", route) <= 0) {
- _debug("Warning SipVoIPLink: call not started\n");
- return EXOSIP_ERROR_STD;
+ SIPCall* call = new SIPCall(id, Call::Outgoing);
+ if (call) {
+ call->setPeerNumber(toUrl);
+ // 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;
}
- } else {
- Manager::instance().displayErrorText(id, "No network found\n");
- return EXOSIP_ERROR_STD;
}
- return 0;
+ return call;
}
-/**
- * @return 0 is good, -1 is bad
- */
-int
-SipVoIPLink::answer(CALLID id)
+bool
+SIPVoIPLink::answer(const CallID& id)
{
- int i;
- int port;
- char tmpbuf[64];
- bzero (tmpbuf, 64);
- // Get port
- snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
-
- _debug("%10d: Answer call [cid = %d, did = %d]\n", id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
- port = getSipCall(id)->getLocalAudioPort();
- _debug(" Local audio port: %d\n", port);
+ _debug("SIP Action: start answering\n");
- osip_message_t *answerMessage = NULL;
- SipCall* ca = getSipCall(id);
+ SIPCall* call = getSIPCall(id);
+ if (call==0) {
+ _debug("SIP Failure: SIPCall doesn't exists\n");
+ return false;
+ }
// Send 180 RINGING
_debug("< Send 180 Ringing\n");
eXosip_lock ();
- eXosip_call_send_answer(ca->getTid(), RINGING, NULL);
+ eXosip_call_send_answer(call->getTid(), SIP_RINGING, NULL);
eXosip_unlock ();
+ call->setConnectionState(Call::Ringing);
// Send 200 OK
+ osip_message_t *answerMessage = NULL;
eXosip_lock();
- i = eXosip_call_build_answer(ca->getTid(), OK, &answerMessage);
+ _debug("< Building 200 OK\n");
+ int i = eXosip_call_build_answer(call->getTid(), SIP_OK, &answerMessage);
if (i != 0) {
_debug("< Send 400 Bad Request\n");
- eXosip_call_send_answer (ca->getTid(), BAD_REQ, NULL);
+ eXosip_call_send_answer (call->getTid(), SIP_BAD_REQUEST, NULL);
} else {
// use exosip, bug locked
i = 0;
- sdp_message_t *remote_sdp = eXosip_get_remote_sdp(ca->getDid());
+ sdp_message_t *remote_sdp = eXosip_get_remote_sdp(call->getDid());
if (remote_sdp!=NULL) {
- i = ca->sdp_complete_message(remote_sdp, answerMessage);
+ i = call->sdp_complete_message(remote_sdp, answerMessage);
if (i!=0) {
osip_message_free(answerMessage);
}
- sdp_message_free (remote_sdp);
+ sdp_message_free(remote_sdp);
}
if (i != 0) {
_debug("< Send 415 Unsupported Media Type\n");
- eXosip_call_send_answer (ca->getTid(), UNSUP_MEDIA_TYPE, NULL);
+ eXosip_call_send_answer (call->getTid(), SIP_UNSUPPORTED_MEDIA_TYPE, NULL);
} else {
_debug("< Send 200 OK\n");
- eXosip_call_send_answer (ca->getTid(), OK, answerMessage);
+ eXosip_call_send_answer (call->getTid(), SIP_OK, answerMessage);
}
}
eXosip_unlock();
- // Incoming call is answered, start the sound channel.
- _debug(" Starting AudioRTP\n");
- if (_audiortp.createNewSession (getSipCall(id)) < 0) {
- _debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
- i = EXOSIP_ERROR_STD;
+ if(i==0) {
+ // Incoming call is answered, start the sound channel.
+ _debug(" Starting AudioRTP\n");
+ if (_audiortp.createNewSession(call) >= 0) {
+ call->setAudioStart(true);
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+ return true;
+ } else {
+ _debug("FATAL: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ }
}
- return i;
-}
+ removeCall(call->getCallId());
+ return false;
+/* int i;
+ int port;
+ char tmpbuf[64];
+ bzero (tmpbuf, 64);
+ // Get port
+ snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
-/**
- * @return > 0 is good, -1 is bad
- */
-int
-SipVoIPLink::hangup (CALLID id)
+ _debug("%10d: Answer call [cid = %d, did = %d]\n", id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
+ port = getSipCall(id)->getLocalAudioPort();
+ _debug(" Local audio port: %d\n", port);
+*/
+}
+
+bool
+SIPVoIPLink::hangup(const CallID& id)
{
- int i = 0;
- SipCall* sipcall = getSipCall(id);
- if (sipcall == NULL) { return EXOSIP_ERROR_STD; }
- _debug("%10d: Hang up call [cid = %d, did = %d]\n",
- id, sipcall->getCid(), sipcall->getDid());
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
+
+ _debug("Hang up call %s [cd: %3d %3d]\n", id.data(), call->getCid(), call->getDid());
// Release SIP stack.
eXosip_lock();
- i = eXosip_call_terminate (sipcall->getCid(), sipcall->getDid());
+ eXosip_call_terminate(call->getCid(), call->getDid());
eXosip_unlock();
// Release RTP channels
- if (id == Manager::instance().getCurrentCallId()) {
+ if (Manager::instance().isCurrentCall(id)) {
_audiortp.closeRtpSession();
}
-
- deleteSipCall(id);
- return i;
+ removeCall(id);
+ return true;
}
-int
-SipVoIPLink::cancel (CALLID id)
+bool
+SIPVoIPLink::cancel(const CallID& id)
{
- int i = 0;
- SipCall* sipcall = getSipCall(id);
- _debug("%10d: Cancel call [cid = %d]\n", id, sipcall->getCid());
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
+
+ _debug("Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
// Release SIP stack.
eXosip_lock();
- i = eXosip_call_terminate (sipcall->getCid(), -1);
+ eXosip_call_terminate(call->getCid(), -1);
eXosip_unlock();
- deleteSipCall(id);
- return i;
+ removeCall(id);
+ return true;
}
-/*
- * @return -1 = sipcall not present
- */
-int
-SipVoIPLink::onhold (CALLID id)
+bool
+SIPVoIPLink::onhold(const CallID& id)
{
- osip_message_t *invite;
- int i;
- int did;
-
- sdp_message_t *local_sdp = NULL;
-
- SipCall *sipcall = getSipCall(id);
- if ( sipcall == NULL ) { return EXOSIP_ERROR_STD; }
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
- did = sipcall->getDid();
+ int did = call->getDid();
eXosip_lock ();
- local_sdp = eXosip_get_local_sdp (did);
+ sdp_message_t *local_sdp = eXosip_get_local_sdp(did);
eXosip_unlock ();
if (local_sdp == NULL) {
- return EXOSIP_ERROR_STD;
+ _debug("SIP Failure: unable to find local_sdp\n");
+ return false;
}
// Build INVITE_METHOD for put call on-hold
+ osip_message_t *invite = NULL;
eXosip_lock ();
- i = eXosip_call_build_request (did, INVITE_METHOD, &invite);
+ int exosipErr = eXosip_call_build_request (did, INVITE_METHOD, &invite);
eXosip_unlock ();
- if (i != 0) {
+ if (exosipErr != 0) {
sdp_message_free(local_sdp);
- return EXOSIP_ERROR_STD;
+ _debug("SIP Failure: unable to build invite method to hold call\n");
+ return false;
}
/* add sdp body */
{
char *tmp = NULL;
- i = sdp_hold_call (local_sdp);
+ int i = sdp_hold_call(local_sdp);
if (i != 0) {
sdp_message_free (local_sdp);
osip_message_free (invite);
- return EXOSIP_ERROR_STD;
+ _debug("SIP Failure: Unable to hold call in SDP\n");
+ return false;
}
- i = sdp_message_to_str (local_sdp, &tmp);
- sdp_message_free (local_sdp);
+ i = sdp_message_to_str(local_sdp, &tmp);
+ sdp_message_free(local_sdp);
if (i != 0) {
osip_message_free (invite);
osip_free (tmp);
- return EXOSIP_ERROR_STD;
+ _debug("SIP Failure: Unable to translate sdp message to string\n");
+ return false;
}
osip_message_set_body (invite, tmp, strlen (tmp));
osip_free (tmp);
osip_message_set_content_type (invite, "application/sdp");
}
- // Send request
+ // Stop sound
+ call->setAudioStart(false);
_audiortp.closeRtpSession();
+ // send request
+ _debug("< Send on hold request\n");
eXosip_lock ();
- i = eXosip_call_send_request (did, invite);
+ exosipErr = eXosip_call_send_request (did, invite);
eXosip_unlock ();
- // Disable audio
- return i;
+ return true;
}
-/**
- * @return 0 is good, -1 is bad
- */
-int
-SipVoIPLink::offhold (CALLID id)
+bool
+SIPVoIPLink::offhold(const CallID& id)
{
- osip_message_t *invite;
- int i;
- int did;
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
- sdp_message_t *local_sdp = NULL;
+ int did = call->getDid();
- did = getSipCall(id)->getDid();
eXosip_lock ();
- local_sdp = eXosip_get_local_sdp (did);
+ sdp_message_t *local_sdp = eXosip_get_local_sdp(did);
eXosip_unlock ();
+
if (local_sdp == NULL) {
- return EXOSIP_ERROR_STD;
+ _debug("SIP Failure: unable to find local_sdp\n");
+ return false;
}
// Build INVITE_METHOD for put call off-hold
+ osip_message_t *invite;
eXosip_lock ();
- i = eXosip_call_build_request (did, INVITE_METHOD, &invite);
+ int exosipErr = eXosip_call_build_request (did, INVITE_METHOD, &invite);
eXosip_unlock ();
- if (i != 0) {
+ if (exosipErr != 0) {
sdp_message_free(local_sdp);
return EXOSIP_ERROR_STD;
}
@@ -585,11 +705,11 @@ SipVoIPLink::offhold (CALLID id)
{
char *tmp = NULL;
- i = sdp_off_hold_call (local_sdp);
+ int i = sdp_off_hold_call (local_sdp);
if (i != 0) {
sdp_message_free (local_sdp);
osip_message_free (invite);
- return EXOSIP_ERROR_STD;
+ return false;
}
i = sdp_message_to_str (local_sdp, &tmp);
@@ -597,7 +717,7 @@ SipVoIPLink::offhold (CALLID id)
if (i != 0) {
osip_message_free (invite);
osip_free (tmp);
- return EXOSIP_ERROR_STD;
+ return false;
}
osip_message_set_body (invite, tmp, strlen (tmp));
osip_free (tmp);
@@ -607,516 +727,67 @@ SipVoIPLink::offhold (CALLID id)
// Send request
_debug("< Send off hold request\n");
eXosip_lock ();
- i = eXosip_call_send_request (did, invite);
+ exosipErr = eXosip_call_send_request (did, invite);
eXosip_unlock ();
// Enable audio
_debug(" Starting AudioRTP\n");
- if (_audiortp.createNewSession (getSipCall(id)) < 0) {
- _debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
- i = EXOSIP_ERROR_STD;
+ // it's sure that this is the current call id...
+ if (_audiortp.createNewSession(call) < 0) {
+ _debug("SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
+ return false;
}
- return i;
+ return true;
}
-int
-SipVoIPLink::transfer (CALLID id, const std::string& to)
+bool
+SIPVoIPLink::transfer(const CallID& id, const std::string& to)
{
- osip_message_t *refer;
- int i;
- std::string tmp_to;
- tmp_to = toHeader(to);
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
+
+ std::string tmp_to = SIPToHeader(to);
if (tmp_to.find("@") == std::string::npos) {
- tmp_to = tmp_to + "@" + Manager::instance().getConfigString(SIGNALISATION,
-HOST_PART);
+ tmp_to = tmp_to + "@" + getHostName();
}
+ osip_message_t *refer;
eXosip_lock();
// Build transfer request
- i = eXosip_call_build_refer (getSipCall(id)->getDid(), (char*)tmp_to.data(),
- &refer);
- if (i == 0) {
+ int exosipErr = eXosip_call_build_refer(call->getDid(), (char*)tmp_to.data(), &refer);
+ if (exosipErr == 0) {
// Send transfer request
- i = eXosip_call_send_request (getSipCall(id)->getDid(), refer);
+ exosipErr = eXosip_call_send_request(call->getDid(), refer);
}
eXosip_unlock();
- return i;
+
+ // shall we delete the call?
+ removeCall(id);
+ return true;
}
-int
-SipVoIPLink::refuse (CALLID id)
+bool
+SIPVoIPLink::refuse (const CallID& id)
{
- int i;
- char tmpbuf[64];
- bzero (tmpbuf, 64);
- // Get local port
- snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
-
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
+
osip_message_t *answerMessage = NULL;
eXosip_lock();
// not BUSY.. where decline the invitation!
- i = eXosip_call_build_answer (getSipCall(id)->getTid(), SIP_DECLINE, &answerMessage);
- if (i == 0) {
- i = eXosip_call_send_answer (getSipCall(id)->getTid(), SIP_DECLINE, answerMessage);
+ int exosipErr = eXosip_call_build_answer(call->getTid(), SIP_DECLINE, &answerMessage);
+ if (exosipErr == 0) {
+ exosipErr = eXosip_call_send_answer(call->getTid(), SIP_DECLINE, answerMessage);
}
eXosip_unlock();
- return i;
-}
-
-int
-SipVoIPLink::getEvent (void)
-{
- // wait for 0 s, 50 ms
- eXosip_event_t* event = eXosip_event_wait (0, 50);
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
-
- if (event == NULL) {
- return EXOSIP_ERROR_STD;
- }
-
- SipCall* sipcall = NULL;
- CALLID id = 0;
- int returnValue = EXOSIP_ERROR_NO;
-
- _debug("SIP Event: #%03d %s\n", event->type, event->textinfo);
-
- switch (event->type) {
- // IP-Phone user receives a new call
- case EXOSIP_CALL_INVITE: //
- _debug("> INVITE (receive)\n");
- checkNetwork();
-
- // Set local random port for incoming call
- if (!Manager::instance().useStun()) {
- setLocalPort(RANDOM_LOCAL_PORT);
- } else {
- // If there is a firewall
- if (behindNat(DEFAULT_LOCAL_PORT) != 0) {
- setLocalPort(Manager::instance().getFirewallPort());
- } else {
- returnValue = EXOSIP_ERROR_STD;
- break;
- }
- }
-
- // Generate id
- id = Manager::instance().generateNewCallId();
- Manager::instance().pushBackNewCall(id, Call::Incoming);
- _debug("%10d: [cid = %d, did = %d]\n", id, event->cid, event->did);
-
- // Associate an audio port with a call
- sipcall = getSipCall(id);
- sipcall->setLocalAudioPort(_localPort);
- sipcall->setLocalIp(getLocalIpAddress());
- _debug(" Local listening port: %d\n", _localPort);
- _debug(" Local listening IP: %s\n", getLocalIpAddress().c_str());
-
- if (sipcall->newIncomingCall(event) == 0 ) {
- if (Manager::instance().incomingCall(id, sipcall->getName(), sipcall->getNumber()) == EXOSIP_ERROR_STD) {
- Manager::instance().displayError("Incoming Call Failed");
- deleteSipCall(id);
- }
- } else {
- Manager::instance().peerHungupCall(id);
- deleteSipCall(id);
- Manager::instance().displayError("Incoming Call Failed");
- }
- break;
-
- case EXOSIP_CALL_REINVITE:
- _debug("> INVITE (reinvite)\n");
- //eXosip_call_send_answer(event->tid, 403, NULL);
- //488 as http://www.atosc.org/pipermail/public/osip/2005-June/005385.html
-
- id = findCallId(event);
- if (id != 0) {
- sipcall = getSipCall(id);
- if (sipcall != 0) {
- _debug("%10d: Receive Reinvite [cid = %d, did = %d], localport=%d\n", id, event->cid, event->did,sipcall->getLocalAudioPort());
-
- if ( id == Manager::instance().getCurrentCallId() ) {
- Manager::instance().stopTone();
- _audiortp.closeRtpSession();
- }
- sipcall->newReinviteCall(event);
- // we should receive an ack after that...
- }
- } else {
- _debug("< Send 488 Not Acceptable Here");
- eXosip_lock();
- eXosip_call_send_answer(event->tid, 488, NULL);
- eXosip_unlock();
- }
- break;
-
- case EXOSIP_CALL_PROCEEDING: // 8
- // proceeding call...
- break;
-
- case EXOSIP_CALL_RINGING: // 9 peer call is ringing
- id = findCallIdInitial(event);
- _debug("%10d: Receive Call Ringing [cid = %d, did = %d]\n", id, event->cid, event->did);
- if (id != 0) {
- getSipCall(id)->ringingCall(event);
- Manager::instance().peerRingingCall(id);
- } else {
- returnValue = -1;
- }
- break;
-
- // The peer-user answers
- case EXOSIP_CALL_ANSWERED: // 10
- {
- id = findCallIdInitial(event);
- if ( id != 0) {
- sipcall = getSipCall(id);
- if ( sipcall != 0 ) {
- _debug("%10d: Receive Call Answer [cid = %d, did = %d], localport=%d\n", id, event->cid, event->did, sipcall->getLocalAudioPort());
-
- // Answer
- if (Manager::instance().callCanBeAnswered(id)) {
- sipcall->setStandBy(false);
- if (sipcall->answeredCall(event) != -1) {
- sipcall->answeredCall_without_hold(event);
- Manager::instance().peerAnsweredCall(id);
-
- if(!Manager::instance().callIsOnHold(id) && Manager::instance().getCurrentCallId()==id) {
- // Outgoing call is answered, start the sound channel.
- _debug(" Starting AudioRTP\n");
- if (_audiortp.createNewSession(sipcall) < 0) {
- _debug(" FATAL: Unable to start sound (%s:%d)\n",
- __FILE__, __LINE__);
- returnValue = -1;
- }
- }
- }
- } else {
- // Answer to on/off hold to send ACK
- _debug(" Answering call\n");
- sipcall->answeredCall(event);
- }
- }
- } else {
- returnValue = -1;
- }
- }
- break;
- case EXOSIP_CALL_REDIRECTED: // 11
- break;
-
- case EXOSIP_CALL_ACK: // 15
- id = findCallId(event);
- _debug("%10d: Receive ACK [cid = %d, did = %d]\n", id, event->cid, event->did);
- if (id != 0 ) {
- sipcall = getSipCall(id);
- if(sipcall != 0 ) {
- sipcall->receivedAck(event);
- if (sipcall->isReinvite()) {
- sipcall->endReinvite();
- if(!Manager::instance().callIsOnHold(id) && Manager::instance().getCurrentCallId()==id) {
- _debug(" Starting AudioRTP\n");
- _audiortp.createNewSession(sipcall);
- } else {
- _debug(" Didn't start RTP because it's on hold or it's not the current call id\n");
- }
- }
- }
- } else {
- returnValue = -1;
- }
- break;
-
- // The peer-user closed the phone call(we received BYE).
- case EXOSIP_CALL_CLOSED: // 25
- id = findCallId(event);
- if (id==0) { id = findCallIdInitial(event); }
- _debug("%10d: Receive BYE [cid = %d, did = %d]\n", id, event->cid, event->did);
- if (id != 0) {
- if (Manager::instance().callCanBeClosed(id)) {
- sipcall = getSipCall(id);
- _audiortp.closeRtpSession();
- }
- Manager::instance().peerHungupCall(id);
- deleteSipCall(id);
- } else {
- returnValue = -1;
- }
- break;
- case EXOSIP_CALL_RELEASED:
- if (event) {
- _debug("SIP call released: [cid = %d, did = %d]\n", event->cid, event->did);
- id = findCallId(event);
- if (id!=0) {
- // not supposed to be execute on a current call...
- Manager::instance().callFailure(id);
- deleteSipCall(id);
- }
-
- }
- break;
- case EXOSIP_CALL_REQUESTFAILURE: //12
-
- _debug("Request Failure, receive code %d\n", event->response->status_code);
- // Handle 4XX errors
- switch (event->response->status_code) {
- case AUTH_REQUIRED:
- _debug("SIP Server ask required authentification: logging...\n");
- setAuthentication();
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
- break;
- case UNAUTHORIZED:
- _debug("Request is unauthorized. SIP Server ask authentification: logging...\n");
- setAuthentication();
- break;
-
- case BAD_REQ:
- case FORBIDDEN:
- case NOT_FOUND:
- case NOT_ALLOWED:
- case NOT_ACCEPTABLE:
- case REQ_TIMEOUT:
- case TEMP_UNAVAILABLE:
- case ADDR_INCOMPLETE:
- case NOT_ACCEPTABLE_HERE: // 488
- // Display error on the screen phone
- //Manager::instance().displayError(event->response->reason_phrase);
- id = findCallId(event);
- Manager::instance().displayErrorText(id, event->response->reason_phrase);
- Manager::instance().callFailure(id);
- deleteSipCall(id);
- break;
- case BUSY_HERE:
- id = findCallId(event);
- Manager::instance().displayErrorText(id, event->response->reason_phrase);
- Manager::instance().callBusy(id);
- deleteSipCall(id);
- break;
- case REQ_TERMINATED:
- break;
- default:
- // sipphone.com send 478 Unresolveable destination (478/TM)
- id = findCallId(event);
- Manager::instance().displayErrorText(id, event->response->reason_phrase);
- Manager::instance().callFailure(id);
- deleteSipCall(id);
- break;
- }
-
- break;
-
- case EXOSIP_CALL_SERVERFAILURE:
- // Handle 5XX errors
- switch (event->response->status_code) {
- case SERVICE_UNAVAILABLE:
- id = findCallId(event);
- Manager::instance().callFailure(id);
- break;
- default:
- break;
- }
- break;
-
- case EXOSIP_CALL_GLOBALFAILURE:
- // Handle 6XX errors
- switch (event->response->status_code) {
- case BUSY_EVERYWHERE:
- case DECLINE:
- id = findCallId(event);
- Manager::instance().callFailure(id);
- break;
- default:
- break;
- }
- break;
-
- case EXOSIP_CALL_MESSAGE_NEW: // 18
- if (0 == event->request) break;
- if (MSG_IS_INFO(event->request)) {
- _debug("Receive a call message request info\n");
- osip_content_type_t* c_t = event->request->content_type;
- if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
- _debug(" Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
- // application/dtmf-relay
- if (strcmp(c_t->type, "application") == 0 && strcmp(c_t->subtype, "dtmf-relay") == 0) {
- handleDtmfRelay(event);
- }
- }
- }
-
- osip_message_t *answerOKNewMessage;
- eXosip_lock();
- if ( 0 == eXosip_call_build_answer(event->tid, OK, &answerOKNewMessage)) {
- _debug("< Sending 200 OK\n");
- eXosip_call_send_answer(event->tid, OK, answerOKNewMessage);
- } else {
- _debug("Could not sent an OK message\n");
- }
- eXosip_unlock();
- break;
-
- case EXOSIP_REGISTRATION_SUCCESS: // 1
- // Manager::instance().displayStatus(LOGGED_IN_STATUS);
- Manager::instance().registrationSucceed();
- break;
-
- case EXOSIP_REGISTRATION_FAILURE: // 2
- //Manager::instance().displayError("getEvent : Registration Failure");
- Manager::instance().registrationFailed();
- break;
-
- case EXOSIP_MESSAGE_NEW: //27
-
- if ( event->request == NULL) {
- break; // do nothing
- }
- unsigned int k;
-
- if (MSG_IS_OPTIONS(event->request)) {
- for (k = 0; k < _sipcallVector.size(); k++) {
- if (_sipcallVector.at(k)->getCid() == event->cid) {
- break;
- }
- }
-
- // TODO: Que faire si rien trouve??
- eXosip_lock();
- if (k == _sipcallVector.size()) {
- /* answer 200 ok */
- eXosip_options_send_answer (event->tid, OK, NULL);
-
- } else if (_sipcallVector.at(k)->getCid() == event->cid) {
- /* already answered! */
- } else {
- /* answer 486 ok */
- eXosip_options_send_answer (event->tid, BUSY_HERE, NULL);
- }
- eXosip_unlock();
- }
-
- // Voice message
- else if (MSG_IS_NOTIFY(event->request)){
- _debug("> NOTIFY Voice message\n");
- int ii;
- unsigned int pos;
- unsigned int pos_slash;
-
- osip_body_t *body = NULL;
- // Get the message body
- ii = osip_message_get_body(event->request, 0, &body);
- if (ii != 0) {
- _debug(" Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
- returnValue = EXOSIP_ERROR_STD;
- break;
- }
-
- // Analyse message body
- if (!body || !body->body) {
- returnValue = EXOSIP_ERROR_STD;
- break;
- }
- std::string str(body->body);
- pos = str.find(VOICE_MSG);
-
- if (pos == std::string::npos) {
- // If the string is not found
- returnValue = EXOSIP_ERROR_STD;
- break;
- }
-
- pos_slash = str.find ("/");
- std::string nb_msg = str.substr(pos + LENGTH_VOICE_MSG,
- pos_slash - (pos + LENGTH_VOICE_MSG));
-
- // Set the number of voice-message
- setMsgVoicemail(atoi(nb_msg.data()));
-
- if (getMsgVoicemail() != 0) {
- // If there is at least one voice-message, start notification
- Manager::instance().startVoiceMessageNotification(nb_msg);
- } else {
- // Stop notification when there is 0 voice message
- Manager::instance().stopVoiceMessageNotification();
- }
- // http://www.jdrosen.net/papers/draft-ietf-simple-im-session-00.txt
- } else if (MSG_IS_MESSAGE(event->request)) {
- _debug("> MESSAGE received\n");
- // osip_content_type_t* osip_message::content_type
- osip_content_type_t* c_t = event->request->content_type;
- if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
- _debug(" Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
-
- osip_body_t *body = NULL;
- // Get the message body
- if (0 == osip_message_get_body(event->request, 0, &body)) {
- _debug(" Body length: %d\n", body->length);
- if (body->body!=0 &&
- strcmp(c_t->type,"text") == 0 &&
- strcmp(c_t->subtype,"plain") == 0
- ) {
- _debug(" Text body: %s\n", body->body);
- Manager::instance().incomingMessage(body->body);
- }
- }
- }
- osip_message_t *answerOK;
- eXosip_lock();
- if ( 0 == eXosip_message_build_answer(event->tid, OK, &answerOK)) {
- _debug("< Sending 200 OK\n");
- eXosip_message_send_answer(event->tid, OK, answerOK);
- }
- eXosip_unlock();
- }
- break;
-
- case EXOSIP_MESSAGE_REQUESTFAILURE: // 31
- // 405 Method Not Allowed
- if (event->response != 0 && event->response->status_code == SIP_METHOD_NOT_ALLOWED) {
-// setAllows(event);
-// AllowList::iterator iter = _allowList.find(std::string("MESSAGE"));
-// if (iter==_allowList.end()) {
- Manager::instance().incomingMessage("Message are not allowed");
-// }
- }
- break;
-
- case EXOSIP_SUBSCRIPTION_ANSWERED: // 38
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
- break;
-
- case EXOSIP_SUBSCRIPTION_REQUESTFAILURE: //40
- break;
-
- default:
- returnValue = EXOSIP_ERROR_STD;
- break;
- }
- eXosip_event_free(event);
-
- return returnValue;
-}
-
-int
-SipVoIPLink::getLocalPort (void)
-{
- return _localPort;
+ return true;
}
-void
-SipVoIPLink::setLocalPort (int port)
+bool
+SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
- _localPort = port;
-}
-
-void
-SipVoIPLink::carryingDTMFdigits (CALLID id, char code) {
- SipCall* sipcall = getSipCall(id);
- if (sipcall == 0) { return; }
+ SIPCall* call = getSIPCall(id);
+ if (call==0) { _debug("Call doesn't exist\n"); return false; }
int duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
osip_message_t *info;
@@ -1127,438 +798,666 @@ SipVoIPLink::carryingDTMFdigits (CALLID id, char code) {
eXosip_lock();
// Build info request
- i = eXosip_call_build_info (getSipCall(id)->getDid(), &info);
+ i = eXosip_call_build_info(call->getDid(), &info);
if (i == 0) {
- snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n",
- code, duration);
+ snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
osip_message_set_content_type (info, "application/dtmf-relay");
osip_message_set_body (info, dtmf_body, strlen (dtmf_body));
// Send info request
- i = eXosip_call_send_request(sipcall->getDid(), info);
+ i = eXosip_call_send_request(call->getDid(), info);
}
eXosip_unlock();
-
+
delete[] dtmf_body; dtmf_body = NULL;
-}
-
-void
-SipVoIPLink::newOutgoingCall (CALLID id)
-{
- SipCall* sipcall = new SipCall(id, Manager::instance().getCodecDescriptorMap());
- if (sipcall != NULL) {
- _sipcallVector.push_back(sipcall);
- sipcall->setStandBy(true);
- }
+ return true;
}
-void
-SipVoIPLink::newIncomingCall (CALLID id)
+bool
+SIPVoIPLink::sendMessage(const std::string& to, const std::string& body)
{
- SipCall* sipcall = new SipCall(id, Manager::instance().getCodecDescriptorMap());
- if (sipcall != NULL) {
- _sipcallVector.push_back(sipcall);
- }
-}
+ bool returnValue = false;
-void
-SipVoIPLink::deleteSipCall (CALLID id)
-{
- std::vector< SipCall* >::iterator iter = _sipcallVector.begin();
+ // fast return
+ if (body.empty()) {return returnValue; }
- while(iter != _sipcallVector.end()) {
- if (*iter && (*iter)->getId() == id) {
- delete *iter; *iter = NULL;
- _sipcallVector.erase(iter);
- return;
- }
- iter++;
+ osip_message_t* message = 0;
+ const char* method = "MESSAGE";
+
+ std::string sipFrom = getSipFrom();
+ std::string sipTo = getSipTo(to);
+ std::string sipRoute = getSipRoute();
+
+ if (!SIPCheckUrl(sipFrom)) {
+ Manager::instance().displayConfigError("Error in source address");
+ return returnValue;
+ }
+ if (!SIPCheckUrl(sipTo)) {
+ Manager::instance().displayError("Error in destination address");
+ return returnValue;
}
-}
-void
-SipVoIPLink::endSipCalls()
-{
- std::vector< SipCall* >::iterator iter = _sipcallVector.begin();
- while(iter != _sipcallVector.end()) {
- if ( *iter ) {
+ int eXosipError = EXOSIP_ERROR_STD;
+ eXosip_lock();
+ if ( sipRoute.empty() ) {
+ eXosipError = eXosip_message_build_request(&message, method, sipTo.c_str(), sipFrom.c_str(), NULL);
+ } else {
+ eXosipError = eXosip_message_build_request(&message, method, sipTo.c_str(), sipFrom.c_str(), sipRoute.c_str());
+ }
- // Release SIP stack.
- eXosip_lock();
- eXosip_call_terminate ((*iter)->getCid(), (*iter)->getDid());
- eXosip_unlock();
+ if (eXosipError == EXOSIP_ERROR_NO) {
+ // add body
+ // add message
+ // src: http://www.atosc.org/pipermail/public/osip/2005-October/006007.html
+ osip_message_set_expires(message, "120");
+ osip_message_set_body(message, body.c_str(), body.length());
+ osip_message_set_content_type(message, "text/plain");
- // Release RTP channels
- _audiortp.closeRtpSession();
- delete *iter; *iter = NULL;
+ eXosipError = eXosip_message_send_request(message);
+ if (eXosipError == EXOSIP_ERROR_NO) {
+ // correctly send the message
+ returnValue = true;
}
- iter++;
}
- _sipcallVector.clear();
+ eXosip_unlock();
+ return returnValue;
}
-SipCall*
-SipVoIPLink::getSipCall (CALLID id)
+bool
+SIPVoIPLink::SIPOutgoingInvite(SIPCall* call)
{
- SipCall* sipcall = NULL;
- for (unsigned int i = 0; i < _sipcallVector.size(); i++) {
- sipcall = _sipcallVector.at(i);
- if (sipcall && sipcall->getId() == id) {
- return sipcall;
- }
+ // 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 NULL;
+ return true;
}
-AudioCodec*
-SipVoIPLink::getAudioCodec (CALLID id)
+bool
+SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
{
- SipCall* sipcall = getSipCall(id);
- if (sipcall != NULL) {
- return sipcall->getAudioCodec();
- } else {
- return NULL;
+ if (!call) return false;
+
+ std::string to = getSipTo(call->getPeerNumber());
+ std::string from = getSipFrom();
+ std::string route = getSipRoute();
+ _debug(" From: %s\n", from.data());
+ _debug(" Route: %s\n", route.data());
+
+ if (!SIPCheckUrl(from)) {
+ _debug("SIP Error: Source address is invalid %s\n", from.data());
+ Manager::instance().displayConfigError("Error in source address");
+ return false;
+ }
+ if (!SIPCheckUrl(to)) {
+ Manager::instance().displayErrorText(call->getCallId(), "Error in destination address");
+ return false;
}
-}
-///////////////////////////////////////////////////////////////////////////////
-// Private functions
-///////////////////////////////////////////////////////////////////////////////
-int
-SipVoIPLink::sdp_hold_call (sdp_message_t * sdp)
-{
- int pos;
- int pos_media = -1;
- char *rcvsnd;
- int recv_send = -1;
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && 0 == strcmp (rcvsnd, "sendonly")) {
- recv_send = 0;
- } else if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "recvonly")
- || 0 == strcmp (rcvsnd, "sendrecv"))) {
- recv_send = 0;
- sprintf (rcvsnd, "sendonly");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ osip_message_t *invite;
+ eXosip_lock();
+ int eXosipError = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
+ (char*)from.data(),
+ (char*)route.data(),
+ (char*)subject.data());
+
+ if (eXosipError != 0) {
+ eXosip_unlock();
+ return false; // error when building the invite
}
- pos_media = 0;
- while (!sdp_message_endof_media (sdp, pos_media)) {
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && 0 == strcmp (rcvsnd, "sendonly")) {
- recv_send = 0;
- } else if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "recvonly")
- || 0 == strcmp (rcvsnd, "sendrecv"))) {
- recv_send = 0;
- sprintf (rcvsnd, "sendonly");
+ setCallAudioLocal(call);
+
+ std::ostringstream media_audio;
+ std::ostringstream rtpmap_attr;
+ int payload;
+ int nbChannel;
+
+ // Set rtpmap according to the supported codec order
+ CodecMap map = call->getCodecMap().getMap();
+ CodecMap::iterator iter = map.begin();
+
+ while(iter != map.end()) {
+ if (iter->second!=0 && iter->second->isActive()) {
+ payload = iter->first;
+ // add each payload in the list of payload
+ media_audio << payload << " ";
+
+ rtpmap_attr << "a=rtpmap: " << payload << " " <<
+ iter->second->getCodecName().data() << "/" << iter->second->getClockRate();
+
+ nbChannel = iter->second->getChannel();
+ if (nbChannel!=1) {
+ rtpmap_attr << "/" << nbChannel;
}
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ rtpmap_attr << "\r\n";
}
- pos_media++;
+ // go to next codec
+ iter++;
}
- if (recv_send == -1) {
- /* we need to add a global attribute with a field set to "sendonly" */
- sdp_message_a_attribute_add (sdp, -1, osip_strdup ("sendonly"), NULL);
+ // http://www.antisip.com/documentation/eXosip2/group__howto1__initialize.html
+ // tell sip if we support SIP extension like 100rel
+ // osip_message_set_supported (invite, "100rel");
+
+ /* add sdp body */
+ {
+ char tmp[4096];
+ snprintf (tmp, 4096,
+ "v=0\r\n"
+ "o=SFLphone 0 0 IN IP4 %s\r\n"
+ "s=call\r\n"
+ "c=IN IP4 %s\r\n"
+ "t=0 0\r\n"
+ "m=audio %d RTP/AVP %s\r\n"
+ "%s",
+ _localExternAddress.c_str(), _localExternAddress.c_str(), call->getLocalExternAudioPort(), media_audio.str().c_str(), rtpmap_attr.str().c_str());
+ // media_audio should be one, two or three numbers?
+ osip_message_set_body (invite, tmp, strlen (tmp));
+ osip_message_set_content_type (invite, "application/sdp");
+ _debug("SDP send: %s", tmp);
}
+
+ _debug("> INVITE To <%s>\n", to.data());
+ int cid = eXosip_call_send_initial_invite(invite);
- return 0;
+ // Keep the cid in case of cancelling
+ call->setCid(cid);
+
+ if (cid <= 0) {
+ eXosip_unlock();
+ return false ;
+ } else {
+ _debug("SIP Info: Outgoing callID is %s, cid=%d\n", call->getCallId().data(), cid);
+ eXosip_call_set_reference (cid, NULL);
+ }
+ eXosip_unlock();
+
+ return true;
}
-int
-SipVoIPLink::sdp_off_hold_call (sdp_message_t * sdp)
-{
- int pos;
- int pos_media = -1;
- char *rcvsnd;
+/**
+ * Get the Sip FROM url (add sip:, add @host, etc...)
+ */
+std::string
+SIPVoIPLink::getSipFrom() {
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "sendonly")
- || 0 == strcmp (rcvsnd, "recvonly"))) {
- sprintf (rcvsnd, "sendrecv");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ // Form the From header field basis on configuration panel
+ std::string host = getHostName();
+ if ( host.empty() ) {
+ host = _localIPAddress;
}
+ return SIPFromHeader(_userpart, host);
+}
- pos_media = 0;
- while (!sdp_message_endof_media (sdp, pos_media)) {
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "sendonly")
- || 0 == strcmp (rcvsnd, "recvonly"))) {
- sprintf (rcvsnd, "sendrecv");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+/**
+ * Get the Sip TO url (add sip:, add @host, etc...)
+ */
+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
+ if (to_url.find("@") == std::string::npos && isRegistered) {
+ std::string host = getHostName();
+ if(!host.empty()) {
+ return SIPToHeader(to_url + "@" + host);
}
- pos_media++;
}
-
- return 0;
+ return SIPToHeader(to_url);
}
-bool
-SipVoIPLink::behindNat(int port)
-{
- StunAddress4 stunSvrAddr;
- stunSvrAddr.addr = 0;
-
- // Stun server
- std::string svr = Manager::instance().getConfigString(SIGNALISATION, STUN_SERVER);
-
- // Convert char* to StunAddress4 structure
- bool ret = stunParseServerName ((char*)svr.data(), stunSvrAddr);
- if (!ret) {
- _debug("SIP: Stun server address (%s) is not valid\n", svr.data());
- return 0;
+/**
+ * Get the sip proxy (add sip: if there is one)
+ * @return empty string or <sip:proxy;lr> url
+ */
+std::string
+SIPVoIPLink::getSipRoute() {
+ std::string proxy = _proxy;
+ if ( !proxy.empty() ) {
+ proxy = "<sip:" + proxy + ";lr>";
}
-
- // Firewall address
- //_debug("STUN server: %s\n", svr.data());
- return Manager::instance().getStunInfo(stunSvrAddr, port);
+ return proxy; // return empty
}
-/**
- * Get the local Ip by eXosip
- * only if the local ip address is to his default value: 127.0.0.1
- * setLocalIpAdress
- * @return false if not found
- */
-bool
-SipVoIPLink::getSipLocalIp (void)
+std::string
+SIPVoIPLink::SIPToHeader(const std::string& to)
{
- bool returnValue = true;
- if (getLocalIpAddress() == "127.0.0.1") {
- char* myIPAddress = new char[65];
- if (eXosip_guess_localip(AF_INET, myIPAddress, 64) == -1) {
- returnValue = false;
- } else {
- setLocalIpAddress(std::string(myIPAddress));
- _debug("Checking network, setting local ip address to: %s\n", myIPAddress);
- }
- delete [] myIPAddress; myIPAddress = NULL;
+ if (to.find("sip:") == std::string::npos) {
+ return ("sip:" + to );
+ } else {
+ return to;
}
- return returnValue;
}
-int
-SipVoIPLink::checkUrl (const std::string& url)
+bool
+SIPVoIPLink::SIPCheckUrl(const std::string& url)
{
int i;
-
+
osip_from_t *to;
i = osip_from_init(&to);
if (i != 0) {
- _debug("Warning: Cannot initialize\n");
- return EXOSIP_ERROR_STD;
+ _debug("SIP Warning: Cannot initialize osip parser\n");
+ return false;
}
i = osip_from_parse(to, url.data());
if (i != 0) {
- _debug("Warning: Cannot parse url\n");
- return EXOSIP_ERROR_STD;
+ _debug("SIP Warning: Cannot parse url %s\n", url.data());
+ return false;
}
// Free memory
osip_from_free (to);
- return 0;
+ return true;
}
-int
-SipVoIPLink::setAuthentication (void)
+bool
+SIPVoIPLink::setCallAudioLocal(SIPCall* call)
{
- ManagerImpl& manager = Manager::instance();
- std::string login = manager.getConfigString(SIGNALISATION, AUTH_USER_NAME);
- if (login.empty()) {
- login = manager.getConfigString(SIGNALISATION, USER_PART);
- }
- std::string pass = manager.getConfigString(SIGNALISATION, PASSWORD);
- if (pass.empty()) {
- manager.displayConfigError("Fill password field");
- return EXOSIP_ERROR_STD;
+ // Setting Audio
+ unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
+ unsigned int callLocalExternAudioPort = callLocalAudioPort;
+ if (_useStun) {
+ // If use Stun server
+ if (Manager::instance().behindNat(callLocalAudioPort)) {
+ callLocalExternAudioPort = Manager::instance().getFirewallPort();
+ }
}
- eXosip_lock();
+ _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);
- int returnValue = eXosip_add_authentication_info(login.data(), login.data(), pass.data(), NULL, NULL);
- eXosip_unlock();
+ return true;
+}
- return returnValue;
+void
+SIPVoIPLink::SIPCallInvite(eXosip_event_t *event)
+{
+ _debug("> INVITE (receive)\n");
+
+ CallID id = Manager::instance().getNewCallID();
+
+ SIPCall* call = new SIPCall(id, Call::Incoming);
+ if (!call) {
+ _debug("SIP Failure: unable to create an incoming call");
+ return;
+ }
+ setCallAudioLocal(call);
+ call->setConnectionState(Call::Progressing);
+ if (call->SIPCallInvite(event)) {
+ if (Manager::instance().incomingCall(call, getAccountID())) {
+ addCall(call);
+ } else {
+ delete call; call = 0;
+ }
+ } else {
+ delete call; call = 0;
+ }
}
-std::string
-SipVoIPLink::fromHeader (const std::string& user, const std::string& host)
+void
+SIPVoIPLink::SIPCallReinvite(eXosip_event_t *event)
{
- std::string displayname = Manager::instance().getConfigString(SIGNALISATION,
-FULL_NAME);
- return ("\"" + displayname + "\"" + " <sip:" + user + "@" + host + ">");
+ _debug("> REINVITE (receive)\n");
+ SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
+ if (call == 0) {
+ _debug("SIP Failure: unknown call\n");
+ _debug("< Send 488 Not Acceptable Here");
+ eXosip_lock();
+ eXosip_call_send_answer(event->tid, 488, NULL);
+ eXosip_unlock();
+ return;
+ }
+ if ( call->getCallId() == Manager::instance().getCurrentCallId()) {
+ // STOP tone
+ Manager::instance().stopTone();
+ // STOP old rtp session
+ _audiortp.closeRtpSession();
+ call->setAudioStart(false);
+ }
+ call->SIPCallReinvite(event);
}
+void
+SIPVoIPLink::SIPCallRinging(eXosip_event_t *event)
+{
+ SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
+ if (!call) {
+ _debug("SIP Failure: unknown call\n");
+ return;
+ }
+ // we could set the cid/did/tid and get the FROM here...
+ // but we found the call with the cid/did already, why setting it again?
+ // call->ringingCall(event);
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+}
-std::string
-SipVoIPLink::toHeader(const std::string& to)
+void
+SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
{
- if (to.find("sip:") == std::string::npos) {
- return ("sip:" + to );
+ SIPCall* call = 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(event);
+
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ Manager::instance().peerAnsweredCall(call->getCallId());
+ if (Manager::instance().isCurrentCall(call->getCallId())) {
+ _debug("SIP: Starting AudioRTP\n");
+ if ( _audiortp.createNewSession(call) < 0) {
+ _debug("RTP Failure: unable to create new session\n");
+ } else {
+ call->setAudioStart(true);
+ }
+ }
} else {
- return to;
+ _debug("Answering call (on/off hold to send ACK)\n");
+ call->SIPCallAnswered(event);
}
}
-int
-SipVoIPLink::startCall(CALLID id, const std::string& from, const std::string& to, const std::string& subject, const std::string& route)
+void
+SIPVoIPLink::SIPCallRequestFailure(eXosip_event_t *event)
{
- SipCall* sipcall = getSipCall(id);
- if ( sipcall == NULL) {
- return -1; // error, we can't find the sipcall
+ if (!event->response) { return; }
+ // 404 error
+ _debug("SIP Event: Request Failure, receive code %d\n", event->response->status_code);
+ // Handle 4XX errors
+ switch (event->response->status_code) {
+ case SIP_PROXY_AUTHENTICATION_REQUIRED:
+ _debug("SIP Server ask required authentification: logging...\n");
+ sendSIPAuthentification();
+ eXosip_lock();
+ eXosip_automatic_action();
+ eXosip_unlock();
+ break;
+ case SIP_UNAUTHORIZED:
+ _debug("Request is unauthorized. SIP Server ask authentification: logging...\n");
+ sendSIPAuthentification();
+ break;
+
+ case SIP_BUSY_HERE: // 486
+ {
+ SIPCall* call = findSIPCallWithCid(event->cid);
+ if (call!=0) {
+ CallID& id = call->getCallId();
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Busy);
+ Manager::instance().displayErrorText(id, event->response->reason_phrase);
+ Manager::instance().callBusy(id);
+ removeCall(id);
+ }
+ }
+ break;
+ case SIP_REQUEST_TERMINATED: // 487
+ break;
+
+ default:
+ /*case SIP_BAD_REQUEST:
+ case SIP_FORBIDDEN:
+ case SIP_NOT_FOUND:
+ case SIP_METHOD_NOT_ALLOWED:
+ case SIP_406_NOT_ACCEPTABLE:
+ case SIP_REQ_TIME_OUT:
+ case SIP_TEMPORARILY_UNAVAILABLE:
+ case SIP_ADDRESS_INCOMPLETE:
+ case SIP_NOT_ACCEPTABLE_HERE: // 488 */
+ // Display error on the screen phone
+ {
+ SIPCall* call = findSIPCallWithCid(event->cid);
+ if (call!=0) {
+ CallID& id = call->getCallId();
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Error);
+ Manager::instance().displayErrorText(id, event->response->reason_phrase);
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ }
+ }
}
- osip_message_t *invite;
+}
- if (checkUrl(from) != 0) {
- Manager::instance().displayConfigError("Error in source address");
- return -1;
+void
+SIPVoIPLink::SIPCallServerFailure(eXosip_event_t *event)
+{
+ 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) {
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ }
+ break;
}
- if (checkUrl(to) != 0) {
- Manager::instance().displayErrorText(id, "Error in destination address");
- return -1;
+}
+
+void
+SIPVoIPLink::SIPCallAck(eXosip_event_t *event)
+{
+ SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
+ if (!call) { return; }
+ if (!call->isAudioStarted()) {
+ if (Manager::instance().isCurrentCall(call->getCallId())) {
+ _debug(" Starting AudioRTP\n");
+ if ( _audiortp.createNewSession(call) ) {
+ call->setAudioStart(true);
+ }
+ }
}
-
- if (!Manager::instance().useStun()) {
- // Set random port for outgoing call if no firewall
- setLocalPort(RANDOM_LOCAL_PORT);
- _debug(" Setting local port to random: %d\n",_localPort);
- } else {
- // If use Stun server
- if (behindNat(_localPort)) {
- setLocalIpAddress(Manager::instance().getFirewallAddress());
- setLocalPort(Manager::instance().getFirewallPort());
- _debug(" Setting local port to firewall port: %d\n", _localPort);
- } else {
- return -1;
+}
+
+void
+SIPVoIPLink::SIPCallMessageNew(eXosip_event_t *event)
+{
+ if (0 == event->request) return;
+
+ _debug("Receive a call message\n");
+
+ if (MSG_IS_INFO(event->request)) {
+ _debug("It's a Request Info\n");
+ osip_content_type_t* c_t = event->request->content_type;
+ if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
+ _debug(" Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
+ // application/dtmf-relay
+ if (strcmp(c_t->type, "application") == 0 && strcmp(c_t->subtype, "dtmf-relay") == 0) {
+ handleDtmfRelay(event);
+ }
}
}
-
- // Set local audio port for sipcall(id)
- sipcall->setLocalIp(getLocalIpAddress());
- sipcall->setLocalAudioPort(_localPort);
+ osip_message_t *answerOKNewMessage;
eXosip_lock();
- int i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
- (char*)from.data(),
- (char*)route.data(),
- (char*)subject.data());
-
- if (i != 0) {
- eXosip_unlock();
- return EXOSIP_ERROR_STD; // error when building the invite
+ if ( 0 == eXosip_call_build_answer(event->tid, SIP_OK, &answerOKNewMessage)) {
+ _debug("< Sending 200 OK\n");
+ eXosip_call_send_answer(event->tid, SIP_OK, answerOKNewMessage);
+ } else {
+ _debug("SIP Failure: Could not sent an OK message\n");
}
+ eXosip_unlock();
- std::ostringstream media_audio;
- std::ostringstream rtpmap_attr;
- int payload;
- int nbChannel;
+}
- // Set rtpmap according to the supported codec order
- CodecMap map = Manager::instance().getCodecDescriptorMap().getMap();
- CodecMap::iterator iter = map.begin();
+void
+SIPVoIPLink::SIPCallClosed(eXosip_event_t *event)
+{
+ // 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);
+ _audiortp.closeRtpSession();
+ }
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+}
- while(iter != map.end()) {
- if (iter->second!=0 && iter->second->isActive()) {
- payload = iter->first;
- // add each payload in the list of payload
- media_audio << payload << " ";
+void
+SIPVoIPLink::SIPCallReleased(eXosip_event_t *event)
+{
+ // 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...
+ CallID id = call->getCallId();
+ Manager::instance().callFailure(id);
+ removeCall(id);
+}
- rtpmap_attr << "a=rtpmap: " << payload << " " <<
- iter->second->getCodecName().data() << "/" << iter->second->getClockRate();
+void
+SIPVoIPLink::SIPMessageNew(eXosip_event_t *event)
+{
+ if (MSG_IS_OPTIONS(event->request)) {
+ // old handling was
+ // - send 200 OK if call id is not found
+ // - send nothing if call id is found
+ eXosip_lock();
+ eXosip_options_send_answer (event->tid, SIP_OK, NULL);
+ eXosip_unlock();
+ }
+ // Voice message
+ else if (MSG_IS_NOTIFY(event->request)){
+ _debug("> NOTIFY Voice message\n");
+ int ii;
+ unsigned int pos;
+ unsigned int pos_slash;
+
+ osip_body_t *body = NULL;
+ // Get the message body
+ ii = osip_message_get_body(event->request, 0, &body);
+ if (ii != 0) {
+ _debug(" Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
+ return;
+ }
- nbChannel = iter->second->getChannel();
- if (nbChannel!=1) {
- rtpmap_attr << "/" << nbChannel;
- }
- rtpmap_attr << "\r\n";
+ // Analyse message body
+ if (!body || !body->body) {
+ return;
}
- // go to next codec
- iter++;
- }
+ std::string str(body->body);
+ pos = str.find(VOICE_MSG);
- // http://www.antisip.com/documentation/eXosip2/group__howto1__initialize.html
- // tell sip if we support SIP extension like 100rel
- // osip_message_set_supported (invite, "100rel");
+ if (pos == std::string::npos) {
+ // If the string is not found
+ return;
+ }
- /* add sdp body */
- {
- char tmp[4096];
- snprintf (tmp, 4096,
- "v=0\r\n"
- "o=SFLphone 0 0 IN IP4 %s\r\n"
- "s=call\r\n"
- "c=IN IP4 %s\r\n"
- "t=0 0\r\n"
- "m=audio %d RTP/AVP %s\r\n"
- "%s",
- getLocalIpAddress().c_str(), getLocalIpAddress().c_str(), getLocalPort(), media_audio.str().c_str(), rtpmap_attr.str().c_str());
- // media_audio should be one, two or three numbers?
- osip_message_set_body (invite, tmp, strlen (tmp));
- osip_message_set_content_type (invite, "application/sdp");
- _debug("SDP send: %s", tmp);
- }
-
- // this is the cid (call id from exosip)
- _debug("%10d: Receive INVITE\n", id);
- int cid = eXosip_call_send_initial_invite (invite);
- _debug(" Local IP:port: %s:%d\n", getLocalIpAddress().c_str(), getLocalPort());
+ pos_slash = str.find ("/");
+ std::string nb_msg = str.substr(pos + LENGTH_VOICE_MSG,
+ pos_slash - (pos + LENGTH_VOICE_MSG));
- // Keep the cid in case of cancelling
- sipcall->setCid(cid);
+ // Set the number of voice-message
+ int msgVoicemail = atoi(nb_msg.data());
- if (cid <= 0) {
+ if (msgVoicemail != 0) {
+ // If there is at least one voice-message, start notification
+ Manager::instance().startVoiceMessageNotification(getAccountID(), nb_msg);
+ } else {
+ // Stop notification when there is 0 voice message
+ Manager::instance().stopVoiceMessageNotification(getAccountID());
+ }
+
+ // http://www.jdrosen.net/papers/draft-ietf-simple-im-session-00.txt
+ } else if (MSG_IS_MESSAGE(event->request)) {
+ _debug("> MESSAGE received\n");
+ // osip_content_type_t* osip_message::content_type
+ osip_content_type_t* c_t = event->request->content_type;
+ if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
+ _debug(" Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
+
+ osip_body_t *body = NULL;
+ // Get the message body
+ if (0 == osip_message_get_body(event->request, 0, &body)) {
+ _debug(" Body length: %d\n", body->length);
+ if (body->body!=0 &&
+ strcmp(c_t->type,"text") == 0 &&
+ strcmp(c_t->subtype,"plain") == 0
+ ) {
+ _debug(" Text body: %s\n", body->body);
+ Manager::instance().incomingMessage(getAccountID(), body->body);
+ }
+ }
+ }
+ osip_message_t *answerOK;
+ eXosip_lock();
+ if ( 0 == eXosip_message_build_answer(event->tid, SIP_OK, &answerOK)) {
+ _debug("< Sending 200 OK\n");
+ eXosip_message_send_answer(event->tid, SIP_OK, answerOK);
+ }
eXosip_unlock();
- return EXOSIP_ERROR_STD;
- } else {
- eXosip_call_set_reference (cid, NULL);
}
- eXosip_unlock();
- return cid; // this is the Cid
}
-CALLID
-SipVoIPLink::findCallId (eXosip_event_t *e)
+SIPCall*
+SIPVoIPLink::findSIPCallWithCid(int cid)
{
- for (unsigned int k = 0; k < _sipcallVector.size(); k++) {
- SipCall* sipcall = _sipcallVector.at(k);
- if (sipcall && sipcall->getCid() == e->cid &&
- sipcall->getDid() == e->did) {
- return sipcall->getId();
+ if (cid < 1) {
+ _debug("Not enough information for this event\n");
+ return 0;
+ }
+ ost::MutexLock m(_callMapMutex);
+ SIPCall* call = 0;
+ CallMap::iterator iter = _callMap.begin();
+ while(iter != _callMap.end()) {
+ call = dynamic_cast<SIPCall*>(iter->second);
+ if (call && call->getCid() == cid) {
+ return call;
}
+ iter++;
}
return 0;
}
-/**
- * This function is used when findCallId failed (return 0)
- * ie: the dialog id change
- * can be use when anwsering a new call or
- * when cancelling a call
- */
-CALLID
-SipVoIPLink::findCallIdInitial (eXosip_event_t *e)
+SIPCall*
+SIPVoIPLink::findSIPCallWithCidDid(int cid, int did)
{
- for (unsigned int k = 0; k < _sipcallVector.size(); k++) {
- SipCall* sipcall = _sipcallVector.at(k);
- // the dialog id is not set when you do a new call
- // so you can't check it when you want to retreive it
- // for the first call anwser
- if (sipcall && sipcall->getCid() == e->cid) {
- return sipcall->getId();
+ if (cid < 1 && did < -1) {
+ _debug("Not enough information for this event\n");
+ return 0;
+ }
+ ost::MutexLock m(_callMapMutex);
+ SIPCall* call = 0;
+ CallMap::iterator iter = _callMap.begin();
+ while(iter != _callMap.end()) {
+ call = dynamic_cast<SIPCall*>(iter->second);
+ if (call && call->getCid() == cid && call->getDid() == did) {
+ return call;
}
+ iter++;
+ }
+ return 0;
+}
+
+SIPCall*
+SIPVoIPLink::getSIPCall(const CallID& id)
+{
+ Call* call = getCall(id);
+ if (call) {
+ return dynamic_cast<SIPCall*>(call);
}
return 0;
}
@@ -1568,7 +1467,12 @@ SipVoIPLink::findCallIdInitial (eXosip_event_t *e)
* @param event eXosip Event
*/
bool
-SipVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
+SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
+
+ SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
+ if (call==0) { return false; }
+
+
bool returnValue = false;
osip_body_t *body = NULL;
// Get the message body
@@ -1591,9 +1495,9 @@ SipVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
_debug("Signal value: %s\n", signal.c_str());
if (!signal.empty()) {
- unsigned int id = findCallId(event);
- if (id !=0 && id == Manager::instance().getCurrentCallId()) {
+ if (Manager::instance().isCurrentCall(call->getCallId())) {
Manager::instance().playDtmf(signal[0]);
+ returnValue = true;
}
}
/*
@@ -1616,77 +1520,89 @@ SipVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
return returnValue;
}
-// http://www.ietf.org/rfc/rfc3428.txt
-// The size of MESSAGE requests outside of a media session MUST NOT
-// exceed 1300 bytes, unless the UAC has positive knowledge that the
-// message will not traverse a congestion-unsafe link at any hop, or
-// that the message size is at least 200 bytes less than the lowest MTU
-// value found en route to the UAS.
-bool
-SipVoIPLink::sendMessage(const std::string& to, const std::string& body) {
- bool returnValue = false;
-
- // fast return
- if (body.empty()) {return returnValue; }
-
- osip_message_t* message = 0;
- const char* method = "MESSAGE";
-
- std::string sipFrom = getSipFrom();
- std::string sipTo = getSipTo(to);
- std::string sipRoute = getSipRoute();
+///////////////////////////////////////////////////////////////////////////////
+// Private functions
+///////////////////////////////////////////////////////////////////////////////
+int
+SIPVoIPLink::sdp_hold_call (sdp_message_t * sdp)
+{
+ int pos;
+ int pos_media = -1;
+ char *rcvsnd;
+ int recv_send = -1;
- if (checkUrl(sipFrom) != 0) {
- Manager::instance().displayConfigError("Error in source address");
- return returnValue;
- }
- if (checkUrl(sipTo) != 0) {
- Manager::instance().displayError("Error in destination address");
- return returnValue;
+ pos = 0;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ while (rcvsnd != NULL) {
+ if (rcvsnd != NULL && 0 == strcmp (rcvsnd, "sendonly")) {
+ recv_send = 0;
+ } else if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "recvonly")
+ || 0 == strcmp (rcvsnd, "sendrecv"))) {
+ recv_send = 0;
+ sprintf (rcvsnd, "sendonly");
+ }
+ pos++;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
}
- int eXosipError = EXOSIP_ERROR_STD;
- eXosip_lock();
- if ( sipRoute.empty() ) {
- eXosipError = eXosip_message_build_request(&message, method, sipTo.c_str(), sipFrom.c_str(), NULL);
- } else {
- eXosipError = eXosip_message_build_request(&message, method, sipTo.c_str(), sipFrom.c_str(), sipRoute.c_str());
+ pos_media = 0;
+ while (!sdp_message_endof_media (sdp, pos_media)) {
+ pos = 0;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ while (rcvsnd != NULL) {
+ if (rcvsnd != NULL && 0 == strcmp (rcvsnd, "sendonly")) {
+ recv_send = 0;
+ } else if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "recvonly")
+ || 0 == strcmp (rcvsnd, "sendrecv"))) {
+ recv_send = 0;
+ sprintf (rcvsnd, "sendonly");
+ }
+ pos++;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ }
+ pos_media++;
}
- if (eXosipError == EXOSIP_ERROR_NO) {
- // add body
- // add message
- // src: http://www.atosc.org/pipermail/public/osip/2005-October/006007.html
- osip_message_set_expires(message, "120");
- osip_message_set_body(message, body.c_str(), body.length());
- osip_message_set_content_type(message, "text/plain");
-
- eXosipError = eXosip_message_send_request(message);
- if (eXosipError == EXOSIP_ERROR_NO) {
- // correctly send the message
- returnValue = true;
- }
+ if (recv_send == -1) {
+ /* we need to add a global attribute with a field set to "sendonly" */
+ sdp_message_a_attribute_add (sdp, -1, osip_strdup ("sendonly"), NULL);
}
- eXosip_unlock();
- return returnValue;
+
+ return 0;
}
-bool
-SipVoIPLink::setAllows(eXosip_event_t *event) {
- bool returnValue = false;
+int
+SIPVoIPLink::sdp_off_hold_call (sdp_message_t * sdp)
+{
+ int pos;
+ int pos_media = -1;
+ char *rcvsnd;
- _allowList.clear();
+ pos = 0;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ while (rcvsnd != NULL) {
+ if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "sendonly")
+ || 0 == strcmp (rcvsnd, "recvonly"))) {
+ sprintf (rcvsnd, "sendrecv");
+ }
+ pos++;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ }
- if (event->response!=0 && event->response->allows!=0) {
- int allowsSize = osip_list_size(event->response->allows);
- if (allowsSize != 0) { returnValue = true;}
- osip_allow_t *allows = 0;
- for(int pos=0; pos<allowsSize; pos++) {
- allows = (osip_allow_t*)osip_list_get(event->response->allows, pos);
- if (allows!=0 && allows->value!=0) {
- _allowList.push_back(std::string(allows->value));
+ pos_media = 0;
+ while (!sdp_message_endof_media (sdp, pos_media)) {
+ pos = 0;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
+ while (rcvsnd != NULL) {
+ if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "sendonly")
+ || 0 == strcmp (rcvsnd, "recvonly"))) {
+ sprintf (rcvsnd, "sendrecv");
}
+ pos++;
+ rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
}
+ pos_media++;
}
- return returnValue;
+
+ return 0;
}
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 392209dbab16f6552aae536acd485296a30a9a18..b9f87bab7582ae0722eae87e6116bac04ccfca12 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -1,10 +1,8 @@
/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
- * Portions Copyright (C) 2002,2003 Aymeric Moizard <jack@atosc.org>
- *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
@@ -19,216 +17,251 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-
-#ifndef __SIP_VOIP_LINK_H__
-#define __SIP_VOIP_LINK_H__
-
-#include <vector>
-#include <map> // for allowed
-#include <eXosip2/eXosip.h>
+#ifndef SIPVOIPLINK_H
+#define SIPVOIPLINK_H
#include "voIPLink.h"
+#include <string>
+#include <eXosip2/eXosip.h>
#include "audio/audiortp.h"
-#include "call.h" // for CALLID
-
-#define EXPIRES_VALUE 180
-// To build request
-#define INVITE_METHOD "INVITE"
-// 1XX responses
-#define DIALOG_ESTABLISHED 101
-#define RINGING 180
-// 2XX
-#define OK 200
-// 4XX Errors
-#define BAD_REQ 400
-#define UNAUTHORIZED 401
-#define FORBIDDEN 403
-#define NOT_FOUND 404
-#define NOT_ALLOWED 405
-#define NOT_ACCEPTABLE 406
-#define AUTH_REQUIRED 407
-#define REQ_TIMEOUT 408
-#define UNSUP_MEDIA_TYPE 415
-#define TEMP_UNAVAILABLE 480
-#define ADDR_INCOMPLETE 484
-#define BUSY_HERE 486
-#define REQ_TERMINATED 487
-#define NOT_ACCEPTABLE_HERE 488 // Not Acceptable Here
-// 5XX errors
-#define SERVICE_UNAVAILABLE 503
-// 6XX errors
-#define BUSY_EVERYWHERE 600
-#define DECLINE 603
-
-class AudioCodec;
-class CodecDescriptor;
-class SipCall;
-class EventThread;
-typedef std::vector< CodecDescriptor* > CodecDescriptorVector;
-typedef std::list<std::string> AllowList;
+class EventThread;
+class SIPCall;
-class SipVoIPLink : public VoIPLink {
+/**
+ @author Yan Morin <yan.morin@gmail.com>
+*/
+class SIPVoIPLink : public VoIPLink
+{
public:
- SipVoIPLink();
- virtual ~SipVoIPLink();
-
- virtual bool init (void);
- virtual bool checkNetwork (void);
- virtual void terminate (void);
- virtual int setRegister (void);
- virtual int setUnregister (void);
- virtual int outgoingInvite (CALLID id, const std::string& to_url);
- virtual int answer (CALLID id);
- virtual int hangup (CALLID id);
- virtual int cancel (CALLID id);
- virtual int onhold (CALLID id);
- virtual int offhold (CALLID id);
- virtual int transfer (CALLID id, const std::string& to);
- virtual int refuse (CALLID id);
- virtual int getEvent (void);
- virtual void carryingDTMFdigits (CALLID id, char code);
-
- /*
- * To handle the local port
- */
- int getLocalPort (void);
- void setLocalPort (int);
- bool getSipLocalIp (void);
-
- /*
- * Add a new SipCall at the end of the SipCallVector with identifiant 'id'
- */
- void newOutgoingCall(CALLID callid);
- void newIncomingCall(CALLID callid);
-
- /*
- * Erase the SipCall(id) from the SipCallVector
- */
- void deleteSipCall(CALLID callid);
-
- /*
- * Return a pointer to the SipCall with identifiant 'id'
- */
- SipCall* getSipCall(CALLID callid);
-
- /*
- * Accessor to the audio codec of SipCall with identifiant 'id'
- */
- AudioCodec* getAudioCodec(CALLID callid);
-
-// Handle voice-message
- inline void setMsgVoicemail (int nMsg) { _nMsgVoicemail = nMsg; }
- inline int getMsgVoicemail (void) { return _nMsgVoicemail; }
-
- /**
- * send text message
- * the size of message should not exceed 1300 bytes
- */
- virtual bool sendMessage(const std::string& to, const std::string& body);
-
+ SIPVoIPLink(const AccountID& accountID);
+
+ ~SIPVoIPLink();
+
+ /** try to initiate the eXosip engine/thread and set config */
+ bool init(void);
+ void terminate(void);
+ bool checkNetwork(void);
+ void getEvent(void);
+
+ bool setRegister(void);
+ bool setUnregister(void);
+
+ Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
+ bool answer(const CallID& id);
+
+ bool hangup(const CallID& id);
+ bool cancel(const CallID& id);
+ bool onhold(const CallID& id);
+ bool offhold(const CallID& id);
+ bool transfer(const CallID& id, const std::string& to);
+ bool refuse (const CallID& id);
+ bool carryingDTMFdigits(const CallID& id, char code);
+ bool sendMessage(const std::string& to, const std::string& body);
+
+
+
+ // SIP Specific
+ /** If set to true, we check for a firewall
+ * @param use true if we use STUN
+ */
+ void setUseStun(bool use) { _useStun = use; }
+ void setProxy(const std::string& proxy) { _proxy = proxy; }
+ void setUserPart(const std::string& userpart) { _userpart = userpart; }
+ void setAuthName(const std::string& authname) { _authname = authname; }
+ void setPassword(const std::string& password) { _password = password; }
+
+
private:
- /*
- * If you are behind a NAT, you have to use STUN server, specified in
- * STUN configuration(you can change this one by default) to give you an
- * public IP address and assign a port number.
- * @param port : on which port we want to listen to
- *
- * Return false if an error occured and true if no error.
- */
- bool behindNat(int port);
-
- /*
- * Return -1 if an error occured and 0 if no error
- */
- int checkUrl(const std::string& url);
-
- /*
- * Allow the authentication when you want register
- * Return -1 if an error occured and 0 if no error
- */
- int setAuthentication (void);
-
- /*
- * Build a sip address from the user configuration
- * Example: "Display user name" <sip:user@host.com>
- * Return the result in a string
- */
- std::string fromHeader (const std::string& user, const std::string& host);
-
- /*
- * Build a sip address with the number that you want to call
- * Example: sip:124@domain.com
- * Return the result in a string
- */
- std::string toHeader(const std::string& to);
-
- /*
- * Beginning point to make outgoing call.
- * Check the 'from' and 'to' url.
- * Allocate local audio port.
- * Build SDP body.
- * Return -1 if an error occured and 0 if no error
- */
- int startCall (CALLID id, const std::string& from, const std::string& to, const std::string& subject, const std::string& route);
+ /** Terminate every call not hangup | brutal | Protected by mutex */
+ void terminateSIPCall();
+
+ /**
+ * Get the local Ip by eXosip
+ * only if the local ip address is to his default value: 127.0.0.1
+ * setLocalIpAdress
+ * @return false if not found
+ */
+ bool loadSIPLocalIP();
+
+ /**
+ * send SIP authentification
+ * @return true if sending succeed
+ */
+ bool sendSIPAuthentification();
+
+ /**
+ * Get a SIP From header ("fullname" <sip:userpart@hostpart>)
+ * @param userpart
+ * @param hostpart
+ * @return SIP URI for from Header
+ */
+ std::string SIPFromHeader(const std::string& userpart, const std::string& hostpart);
+
+ /**
+ * Build a sip address with the number that you want to call
+ * Example: sip:124@domain.com
+ * @return result as a string
+ */
+ std::string SIPToHeader(const std::string& to);
+
+ /**
+ * Check if an url is sip-valid
+ * @return true if osip tell that is valid
+ */
+ bool SIPCheckUrl(const std::string& url);
+
+
+ /**
+ * SIPOutgoingInvite do SIPStartCall
+ * @return true if all is correct
+ */
+ bool SIPOutgoingInvite(SIPCall* call);
+ /**
+ * Start a SIP Call
+ * @return true if all is correct
+ */
+ bool SIPStartCall(SIPCall* call, const std::string& subject);
std::string getSipFrom();
std::string getSipRoute();
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 true
+ */
+ bool setCallAudioLocal(SIPCall* call);
+ /**
+ * Create a new call and send a incoming call notification to the user
+ * @param event eXosip Event
+ */
+ void SIPCallInvite(eXosip_event_t *event);
- /*
- * Look for call with same cid/did
- * Return the id of the found call
- */
- CALLID findCallId (eXosip_event_t *e);
- CALLID findCallIdInitial (eXosip_event_t *e);
+ /**
+ * Use a exisiting call to restart the audio
+ * @param event eXosip Event
+ */
+ void SIPCallReinvite(eXosip_event_t *event);
- /*
- * To build sdp when call is on-hold
- */
- int sdp_hold_call (sdp_message_t * sdp);
-
- /*
- * To build sdp when call is off-hold
- */
- int sdp_off_hold_call (sdp_message_t * sdp);
+ /**
+ * Tell the user that the call is ringing
+ * @param event eXosip Event
+ */
+ void SIPCallRinging(eXosip_event_t *event);
+
+ /**
+ * Tell the user that the call was answered
+ * @param event eXosip Event
+ */
+ void SIPCallAnswered(eXosip_event_t *event);
+
+ /**
+ * Handling 4XX error
+ * @param event eXosip Event
+ */
+ void SIPCallRequestFailure(eXosip_event_t *event);
+
+ /**
+ * Handling 5XX/6XX error
+ * @param event eXosip Event
+ */
+ void SIPCallServerFailure(eXosip_event_t *event);
/**
- * Subscribe to message summary
+ * Handling ack (restart audio if reinvite)
+ * @param event eXosip Event
*/
- void subscribeMessageSummary();
+ void SIPCallAck(eXosip_event_t *event);
/**
- * End all sip call not deleted
+ * Handling message inside a call (like dtmf)
+ * @param event eXosip Event
*/
- void endSipCalls();
+ void SIPCallMessageNew(eXosip_event_t *event);
+ bool handleDtmfRelay(eXosip_event_t *event);
/**
- * Handle DTMF Relay INFO Request
+ * Peer close the connection
+ * @param event eXosip Event
*/
- bool handleDtmfRelay(eXosip_event_t* event);
+ void SIPCallClosed(eXosip_event_t *event);
/**
- * Get from a response event, all allowed request
+ * The call pointer was released
+ * If the call was not cleared before, report an error
+ * @param event eXosip Event
*/
- bool setAllows(eXosip_event_t *event);
+ void SIPCallReleased(eXosip_event_t *event);
+
+ /**
+ * Receive a new Message request
+ * Option/Notify/Message
+ * @param event eXosip Event
+ */
+ void SIPMessageNew(eXosip_event_t *event);
+
+ /**
+ * Find a SIPCall with cid from eXosip Event
+ * Explication there is no DID when the dialog is not establish...
+ * @param cid call ID
+ * @return 0 or SIPCall pointer
+ */
+ SIPCall* findSIPCallWithCid(int cid);
+
+ /**
+ * Find a SIPCall with cid and did from eXosip Event
+ * @param cid call ID
+ * @param did domain ID
+ * @return 0 or SIPCall pointer
+ */
+ SIPCall* findSIPCallWithCidDid(int cid, int did);
+ SIPCall* getSIPCall(const CallID& id);
+
+ /** To build sdp when call is on-hold */
+ int sdp_hold_call (sdp_message_t * sdp);
+ /** To build sdp when call is off-hold */
+ int sdp_off_hold_call (sdp_message_t * sdp);
+
+
+
+ /** EventThread get every incoming events */
+ EventThread* _evThread;
+ /** Tell if eXosip was stared (eXosip_init) */
+ bool _eXosipStarted;
+ /** Registration identifier, needed by unregister to build message */
+ int _eXosipRegID;
+ /** Number of voicemail */
+ int _nMsgVoicemail;
+
+ /** when we init the listener, how many times we try to bind a port? */
+ int _nbTryListenAddr;
+
+ /** Do we use stun? */
+ bool _useStun;
+
+ /** Local Extern Address is the IP address seens by peers for SIP listener */
+ std::string _localExternAddress;
+
+ /** Local Extern Port is the port seens by peers for SIP listener */
+ unsigned int _localExternPort;
+
+ /** SIP Proxy URL */
+ std::string _proxy;
+ /** SIP UserPart */
+ std::string _userpart;
- ///////////////////////////
- // Private member variables
- ///////////////////////////
- EventThread *_evThread;
- std::vector< SipCall * > _sipcallVector;
- AllowList _allowList;
+ /** SIP Authenfication name */
+ std::string _authname;
- AudioRtp _audiortp;
- int _localPort;
- int _reg_id;
- int _nMsgVoicemail;
+ /** SIP Authenfication password */
+ std::string _password;
- bool _registrationSend; // unregistered
- bool _started; // eXosip_init and eXosip_start
+ /** Starting sound */
+ AudioRtp _audiortp;
};
-#endif // __SIP_VOIP_LINK_H__
+#endif
diff --git a/src/user_cfg.h b/src/user_cfg.h
index 9d55dfe455ad6c6c305a7c96a589793e3091d5b3..1237e55a32f80c4493360fafab843f0abf0862d1 100644
--- a/src/user_cfg.h
+++ b/src/user_cfg.h
@@ -38,7 +38,7 @@
#define PREFERENCES "Preferences"
#define ACCOUNT_SIP0 "SIP0"
-#define ACCOUNT_AIX0 "AIX0"
+#define ACCOUNT_IAX0 "IAX0"
// Fields to fill
diff --git a/src/voIPLink.cpp b/src/voIPLink.cpp
index 6ff6b561721a50bc8905e0506537e413da953069..4612ae53fc3132d698317df9e9c6900e4b3bf108 100644
--- a/src/voIPLink.cpp
+++ b/src/voIPLink.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2005-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
@@ -22,48 +22,61 @@
#include "user_cfg.h"
#include "voIPLink.h"
-#include "manager.h"
-VoIPLink::VoIPLink ()
+VoIPLink::VoIPLink(const AccountID& accountID) : _accountID(accountID), _localIPAddress("127.0.0.1"), _localPort(0)
{
+
}
VoIPLink::~VoIPLink (void)
{
+ clearCallMap();
}
-void
-VoIPLink::setFullName (const std::string& fullname)
+bool
+VoIPLink::addCall(Call* call)
{
- _fullname = fullname;
+ if (call) {
+ if (getCall(call->getCallId()) == 0) {
+ ost::MutexLock m(_callMapMutex);
+ _callMap[call->getCallId()] = call;
+ }
+ }
+ return false;
}
-std::string
-VoIPLink::getFullName (void)
+bool
+VoIPLink::removeCall(const CallID& id)
{
- return _fullname;
+ ost::MutexLock m(_callMapMutex);
+ if (_callMap.erase(id)) {
+ return true;
+ }
+ return false;
}
-void
-VoIPLink::setHostName (const std::string& hostname)
+Call*
+VoIPLink::getCall(const CallID& id)
{
- _hostname = hostname;
+ ost::MutexLock m(_callMapMutex);
+ CallMap::iterator iter = _callMap.find(id);
+ if ( iter != 0 && iter != _callMap.end() ) {
+ return iter->second;
+ }
+ return 0;
}
-std::string
-VoIPLink::getHostName (void)
+bool
+VoIPLink::clearCallMap()
{
- return _hostname;
-}
-
-void
-VoIPLink::setLocalIpAddress (const std::string& ipAdress)
-{
- _localIpAddress = ipAdress;
+ ost::MutexLock m(_callMapMutex);
+ CallMap::iterator iter = _callMap.begin();
+ while( iter != _callMap.end() ) {
+ // if (iter) ?
+ delete iter->second; iter->second = 0;
+ iter++;
+ }
+ _callMap.clear();
+ return true;
}
-std::string
-VoIPLink::getLocalIpAddress (void)
-{
- return _localIpAddress;
-}
diff --git a/src/voIPLink.h b/src/voIPLink.h
index bfd7f957eea9d7174ad6593f957aec70dad31342..e0644bb2209303eb9d6eafd034da63fed2c62624 100644
--- a/src/voIPLink.h
+++ b/src/voIPLink.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2005-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
@@ -22,49 +22,87 @@
#define __VOIP_LINK_H__
#include <string>
+#include "account.h" // for AccountID
#include "call.h"
+#include <map>
+#include <cc++/thread.h> // for mutex
class AudioCodec;
-class Call;
+
+typedef std::map<CallID, Call*> CallMap;
class VoIPLink {
public:
- VoIPLink ();
+ VoIPLink(const AccountID& accountID);
virtual ~VoIPLink (void);
// Pure virtual functions
- virtual int getEvent (void) = 0;
+ virtual void getEvent (void) = 0;
virtual bool init (void) = 0;
virtual bool checkNetwork (void) = 0;
virtual void terminate (void) = 0;
- virtual void newOutgoingCall (CALLID id) = 0;
- virtual void newIncomingCall (CALLID id) = 0;
- virtual int outgoingInvite (CALLID id, const std::string& to_url) = 0;
- virtual int answer (CALLID id) = 0;
- virtual int hangup (CALLID id) = 0;
- virtual int cancel (CALLID id) = 0;
- virtual int onhold (CALLID id) = 0;
- virtual int offhold (CALLID id) = 0;
- virtual int transfer (CALLID id, const std::string& to) = 0;
- virtual int refuse (CALLID id) = 0;
- virtual int setRegister (void) = 0;
- virtual int setUnregister (void) = 0;
- virtual void carryingDTMFdigits(CALLID id, char code) = 0;
- virtual AudioCodec* getAudioCodec (CALLID id) = 0;
+ virtual bool setRegister (void) = 0;
+ virtual bool setUnregister (void) = 0;
+
+ /** Add a new outgoing call and return the call pointer or 0 if and error occurs */
+ virtual Call* newOutgoingCall(const CallID& id, const std::string& toUrl) = 0;
+ virtual bool answer(const CallID& id) = 0;
+
+ virtual bool hangup(const CallID& id) = 0;
+ virtual bool cancel(const CallID& id) = 0;
+ virtual bool onhold(const CallID& id) = 0;
+ virtual bool offhold(const CallID& id) = 0;
+ virtual bool transfer(const CallID& id, const std::string& to) = 0;
+ virtual bool refuse(const CallID& id) = 0;
+ virtual bool carryingDTMFdigits(const CallID& id, char code) = 0;
virtual bool sendMessage(const std::string& to, const std::string& body) = 0;
- void setFullName (const std::string& fullname);
- std::string getFullName (void);
- void setHostName (const std::string& hostname);
- std::string getHostName (void);
- void setLocalIpAddress (const std::string& ipAdress);
- std::string getLocalIpAddress (void);
+ // these method are set only with 'Account init' and can be get by everyone
+ void setFullName (const std::string& fullname) { _fullname = fullname; }
+ std::string& getFullName (void) { return _fullname; }
+ void setHostName (const std::string& hostname) { _hostname = hostname; }
+ std::string& getHostName (void) { return _hostname; }
+ AccountID& getAccountID(void) { return _accountID; }
+
+ /** Get the call pointer from the call map (protected by mutex)
+ * @param id A Call ID
+ * @return call pointer or 0
+ */
+ Call* getCall(const CallID& id);
+
+
+private:
+ std::string _fullname;
+ std::string _hostname;
+ AccountID _accountID;
protected:
-
- std::string _fullname;
- std::string _hostname;
- std::string _localIpAddress;
+ /** Add a call to the call map (protected by mutex)
+ * @param call A call pointer with a unique pointer
+ * @return true if the call was unique and added
+ */
+ bool addCall(Call* call);
+
+ /** remove a call from the call map (protected by mutex)
+ * @param id A Call ID
+ * @return true if the call was correctly removed
+ */
+ bool removeCall(const CallID& id);
+
+ /**
+ * Remove all the call from the map
+ */
+ bool clearCallMap();
+
+ /** Contains all the calls for this Link, protected by mutex */
+ CallMap _callMap;
+ /** Mutex to protect call map */
+ ost::Mutex _callMapMutex;
+
+ /** Get Local IP Address (ie: 127.0.0.1, 192.168.0.1, ...) */
+ std::string _localIPAddress;
+ /** Get local listening port (5060 for SIP, ...) */
+ unsigned int _localPort;
};
#endif // __VOIP_LINK_H__