diff --git a/sflphone-gtk/src/dbus.c b/sflphone-gtk/src/dbus.c
index 4332d09df0d291e09bc2c579e9996e4943892c86..4debf578405a73919aa8ddc0b0966d53e34252ea 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -86,7 +86,7 @@ incoming_message_cb (DBusGProxy *proxy,
const gchar* msg,
void * foo )
{
- g_print ("Messge %s! \n",msg);
+ g_print ("Message %s! \n",msg);
}
diff --git a/src/account.h b/src/account.h
index 2f2c5123268e2e35ae0b8422dba592871dba2019..af0a20d7e425d344be641ec49c3b837c03845c73 100644
--- a/src/account.h
+++ b/src/account.h
@@ -27,38 +27,56 @@
class VoIPLink;
+/**
+ * @file account.h
+ * @brief Interface to protocol account (SIPAccount, IAXAccount)
+ * It can be enable on loading or activate after.
+ * It contains account, configuration, VoIP Link and Calls (inside the VoIPLink)
+ */
+
typedef std::string AccountID;
+
#define AccountNULL ""
+/** Account type: SIP / IAX2 are supported */
#define CONFIG_ACCOUNT_TYPE "Account.type"
+/** Tells if account is enable or not */
#define CONFIG_ACCOUNT_ENABLE "Account.enable"
-//#define CONFIG_ACCOUNT_AUTO_REGISTER "Account.autoregister"
+/** Account alias */
#define CONFIG_ACCOUNT_ALIAS "Account.alias"
-
+/** IAX paramater : full name */
#define IAX_FULL_NAME "IAX.fullName"
+/** IAX paramater : host name */
#define IAX_HOST "IAX.host"
+/** IAX paramater : user name */
#define IAX_USER "IAX.user"
+/** IAX paramater : password */
#define IAX_PASS "IAX.pass"
-
+/** SIP parameter : full name */
#define SIP_FULL_NAME "SIP.fullName"
+/** SIP parameter : user name */
#define SIP_USER_PART "SIP.userPart"
+/** SIP parameter : authorization name */
#define SIP_AUTH_NAME "SIP.username"
+/** SIP parameter : password */
#define SIP_PASSWORD "SIP.password"
+/** SIP parameter : host name */
#define SIP_HOST_PART "SIP.hostPart"
+/** SIP parameter : proxy address */
#define SIP_PROXY "SIP.proxy"
+/** SIP parameter : stun server address */
#define SIP_STUN_SERVER "STUN.server"
+/** SIP parameter : tells if stun is used or not */
#define SIP_USE_STUN "STUN.enable"
+/** SIP parameter : stun port */
#define SIP_STUN_PORT "STUN.port"
-/**
- * Class account is an interface to protocol account (SIPAccount, IAXAccount)
- * It can be enable on loading or activate after.
- * It contains account, configuration, VoIP Link and Calls (inside the VoIPLink)
- * @author Yan Morin
- */
class Account{
public:
Account(const AccountID& accountID);
-
+
+ /**
+ * Virtual destructor
+ */
virtual ~Account();
/**
@@ -74,35 +92,32 @@ class Account{
/**
* Get the voiplink pointer
- * @return the pointer or 0
+ * @return VoIPLink* the pointer or 0
*/
inline VoIPLink* getVoIPLink() { return _link; }
/**
* Register the underlying VoIPLink. Launch the event listener.
- *
* This should update the getRegistrationState() return value.
- *
- * @return false is an error occurs
*/
virtual void registerVoIPLink() = 0;
/**
* Unregister the underlying VoIPLink. Stop the event listener.
- *
* This should update the getRegistrationState() return value.
- *
- * @return false is an error occurs
*/
virtual void unregisterVoIPLink() = 0;
/**
- * Tell if the account is enable or not. See doc for _enabled.
+ * Tell if the account is enable or not.
+ * @return true if enabled
+ * false otherwise
*/
bool isEnabled() { return _enabled; }
/**
- * Return registration state of underlying VoIPLink
+ * Get the registration state of the specified link
+ * @return RegistrationState The registration state of underlying VoIPLink
*/
VoIPLink::RegistrationState getRegistrationState() { return _link->getRegistrationState(); }
@@ -136,9 +151,7 @@ protected:
/**
* Tells if the link is enabled, active.
- *
* This implies the link will be initialized on startup.
- *
* Modified by the configuration (key: ENABLED)
*/
bool _enabled;
diff --git a/src/accountcreator.h b/src/accountcreator.h
index 01215077c2ec20f9576bcbc42262b6e1f876509a..c91b3f7220567779cffa54505ce38209a19bbce6 100644
--- a/src/accountcreator.h
+++ b/src/accountcreator.h
@@ -25,8 +25,8 @@
class Account;
/**
- * AccountCreator create Protocol-specific Account
- * @author Yan Morin <yan.morin@gmail.com>
+ * @file accountcreator.h
+ * @brief Create protocol-specific account
*/
class AccountCreator{
public:
diff --git a/src/audio/audiodevice.h b/src/audio/audiodevice.h
index fb29ebf276f4304556991ffcb21ef384c42c9328..acdfee46368c8f9d187a6d82556600ad4ef8dabb 100644
--- a/src/audio/audiodevice.h
+++ b/src/audio/audiodevice.h
@@ -26,7 +26,8 @@
#define AUDIODEVICERATE 8000
/**
- * Container device for attribute storage
+ * @file audiodevice.c
+ * @brief Container device for attribute storage
* Have almost only get/set method
*/
class AudioDevice {
diff --git a/src/call.h b/src/call.h
index fbac5779970d39c6347c85f38f0b639ab4c4fe84..63b2de4443d93d6a89350e1e7768f597dc1c6380 100644
--- a/src/call.h
+++ b/src/call.h
@@ -20,37 +20,39 @@
#ifndef CALL_H
#define CALL_H
-/* @file call.h
- * @brief A call is the base class for protocol-based calls
- */
#include <string>
#include <cc++/thread.h> // for mutex
#include "audio/codecDescriptor.h"
+/*
+ * @file call.h
+ * @brief A call is the base class for protocol-based calls
+ */
+
typedef std::string CallID;
class Call{
-public:
- /**
- * This determines if the call originated from the local user (Outgoing)
- * or from some remote peer (Incoming).
- */
- enum CallType {Incoming, Outgoing};
-
- /**
- * Tell where we're at with the call. The call gets Connected when we know
- * from the other end what happened with out call. A call can be 'Connected'
- * even if the call state is Busy, Refused, or Error.
- *
- * Audio should be transmitted when ConnectionState = Connected AND
- * CallState = Active.
- */
- enum ConnectionState {Disconnected, Trying, Progressing, Ringing, Connected};
-
- /**
- * The Call State.
- */
- enum CallState {Inactive, Active, Hold, Busy, Refused, Error};
+ public:
+ /**
+ * This determines if the call originated from the local user (Outgoing)
+ * or from some remote peer (Incoming).
+ */
+ enum CallType {Incoming, Outgoing};
+
+ /**
+ * Tell where we're at with the call. The call gets Connected when we know
+ * from the other end what happened with out call. A call can be 'Connected'
+ * even if the call state is Busy, Refused, or Error.
+ *
+ * Audio should be transmitted when ConnectionState = Connected AND
+ * CallState = Active.
+ */
+ enum ConnectionState {Disconnected, Trying, Progressing, Ringing, Connected};
+
+ /**
+ * The Call State.
+ */
+ enum CallState {Inactive, Active, Hold, Busy, Refused, Error};
/**
* Constructor of a call
@@ -72,96 +74,160 @@ public:
* @param number peer number
*/
void setPeerNumber(const std::string& number) { _peerNumber = number; }
+
+ /**
+ * Get the peer number (destination on outgoing)
+ * not protected by mutex (when created)
+ * @return std::string The peer number
+ */
const std::string& getPeerNumber() { return _peerNumber; }
/**
* Set the peer name (caller in ingoing)
* not protected by mutex (when created)
- * @param number peer number
+ * @param name The peer name
*/
void setPeerName(const std::string& name) { _peerName = name; }
+
+ /**
+ * Get the peer name (caller in ingoing)
+ * not protected by mutex (when created)
+ * @return std::string The peer name
+ */
const std::string& getPeerName() { return _peerName; }
/**
* Tell if the call is incoming
+ * @return true if yes
+ * false otherwise
*/
bool isIncoming() { return (_type == Incoming) ? true : false; }
/**
* Set the connection state of the call (protected by mutex)
+ * @param state The connection state
*/
void setConnectionState(ConnectionState state);
+
/**
- * get the connection state of the call (protected by mutex)
+ * Get the connection state of the call (protected by mutex)
+ * @return ConnectionState The connection state
*/
ConnectionState getConnectionState();
/**
* Set the state of the call (protected by mutex)
+ * @param state The call state
*/
void setState(CallState state);
+
/**
- * get the call state of the call (protected by mutex)
+ * Get the call state of the call (protected by mutex)
+ * @return CallState The call state
*/
CallState getState();
/**
* Set the audio start boolean (protected by mutex)
* @param start true if we start the audio
+ * false otherwise
*/
void setAudioStart(bool start);
/**
* Tell if the audio is started (protected by mutex)
* @return true if it's already started
+ * false otherwise
*/
bool isAudioStarted();
// AUDIO
- /** Set internal codec Map: initialization only, not protected */
+ /**
+ * Set internal codec Map: initialization only, not protected
+ * @param map The codec map
+ */
void setCodecMap(const CodecDescriptor& map) { _codecMap = map; }
+
+ /**
+ * Get internal codec Map: initialization only, not protected
+ * @return CodecDescriptor The codec map
+ */
CodecDescriptor& getCodecMap();
- /** Set my IP [not protected] */
+ /**
+ * Set my IP [not protected]
+ * @param ip The local IP address
+ */
void setLocalIp(const std::string& ip) { _localIPAddress = ip; }
- /** Set local audio port, as seen by me [not protected] */
+ /**
+ * Set local audio port, as seen by me [not protected]
+ * @param port The local audio port
+ */
void setLocalAudioPort(unsigned int port) { _localAudioPort = port;}
- /** Set the audio port that remote will see. */
+ /**
+ * Set the audio port that remote will see.
+ * @param port The external audio port
+ */
void setLocalExternAudioPort(unsigned int port) { _localExternalAudioPort = port; }
- /** Return the audio port seen by the remote side. */
+ /**
+ * Return the audio port seen by the remote side.
+ * @return unsigned int The external audio port
+ */
unsigned int getLocalExternAudioPort() { return _localExternalAudioPort; }
- /** Return my IP [mutex protected] */
+ /**
+ * Return my IP [mutex protected]
+ * @return std::string The local IP
+ */
const std::string& getLocalIp();
-
- /** Return port used locally (for my machine) [mutex protected] */
+
+ /**
+ * Return port used locally (for my machine) [mutex protected]
+ * @return unsigned int The local audio port
+ */
unsigned int getLocalAudioPort();
-
- /** Return audio port at destination [mutex protected] */
+
+ /**
+ * Return audio port at destination [mutex protected]
+ * @return unsigned int The remote audio port
+ */
unsigned int getRemoteAudioPort();
- /** Return IP of destination [mutex protected] */
+ /**
+ * Return IP of destination [mutex protected]
+ * @return const std:string The remote IP address
+ */
const std::string& getRemoteIp();
- /** Return audio codec [mutex protected] */
+ /**
+ * Return audio codec [mutex protected]
+ * @return AudioCodecType The payload of the codec
+ */
AudioCodecType getAudioCodec();
-
-
-protected:
+ protected:
/** Protect every attribute that can be changed by two threads */
ost::Mutex _callMutex;
- /** Set remote's IP addr. [not protected] */
+ /**
+ * Set remote's IP addr. [not protected]
+ * @param ip The remote IP address
+ */
void setRemoteIP(const std::string& ip) { _remoteIPAddress = ip; }
- /** Set remote's audio port. [not protected] */
+ /**
+ * Set remote's audio port. [not protected]
+ * @param port The remote audio port
+ */
void setRemoteAudioPort(unsigned int port) { _remoteAudioPort = port; }
- /** Set the audio codec used. [not protected] */
+ /**
+ * Set the audio codec used. [not protected]
+ * @param audioCodec The payload of the codec
+ */
void setAudioCodec(AudioCodecType audioCodec) { _audioCodec = audioCodec; }
/** Codec Map */
@@ -190,7 +256,7 @@ protected:
unsigned int _remoteAudioPort;
-private:
+ private:
/** Unique ID of the call */
CallID _id;
diff --git a/src/dbus/configurationmanager-glue.h b/src/dbus/configurationmanager-glue.h
index 0bda986c4ba852695a61f79f14037a561defdd39..a038f85c4a9702ccfe444ec35e497fffb35bdd53 100644
--- a/src/dbus/configurationmanager-glue.h
+++ b/src/dbus/configurationmanager-glue.h
@@ -25,8 +25,6 @@ public:
register_method(ConfigurationManager, addAccount, _addAccount_stub);
register_method(ConfigurationManager, removeAccount, _removeAccount_stub);
register_method(ConfigurationManager, getAccountList, _getAccountList_stub);
- register_method(ConfigurationManager, getDefaultAccount, _getDefaultAccount_stub);
- register_method(ConfigurationManager, setDefaultAccount, _setDefaultAccount_stub);
register_method(ConfigurationManager, sendRegister, _sendRegister_stub);
register_method(ConfigurationManager, getToneLocaleList, _getToneLocaleList_stub);
register_method(ConfigurationManager, getVersion, _getVersion_stub);
@@ -90,16 +88,6 @@ public:
{ "list", "as", false },
{ 0, 0, 0 }
};
- static ::DBus::IntrospectedArgument getDefaultAccount_args[] =
- {
- { "accountID", "s", false },
- { 0, 0, 0 }
- };
- static ::DBus::IntrospectedArgument setDefaultAccount_args[] =
- {
- { "accountID", "s", true },
- { 0, 0, 0 }
- };
static ::DBus::IntrospectedArgument sendRegister_args[] =
{
{ "accountID", "s", true },
@@ -280,8 +268,6 @@ public:
{ "addAccount", addAccount_args },
{ "removeAccount", removeAccount_args },
{ "getAccountList", getAccountList_args },
- { "getDefaultAccount", getDefaultAccount_args },
- { "setDefaultAccount", setDefaultAccount_args },
{ "sendRegister", sendRegister_args },
{ "getToneLocaleList", getToneLocaleList_args },
{ "getVersion", getVersion_args },
@@ -353,8 +339,6 @@ public:
virtual void addAccount( const std::map< ::DBus::String, ::DBus::String >& details ) = 0;
virtual void removeAccount( const ::DBus::String& accoundID ) = 0;
virtual std::vector< ::DBus::String > getAccountList( ) = 0;
- virtual ::DBus::String getDefaultAccount( ) = 0;
- virtual void setDefaultAccount( const ::DBus::String& accountID ) = 0;
virtual void sendRegister( const ::DBus::String& accountID, const ::DBus::Int32& expire ) = 0;
virtual std::vector< ::DBus::String > getToneLocaleList( ) = 0;
virtual ::DBus::String getVersion( ) = 0;
@@ -465,25 +449,6 @@ private:
wi << argout1;
return reply;
}
- ::DBus::Message _getDefaultAccount_stub( const ::DBus::CallMessage& call )
- {
- ::DBus::MessageIter ri = call.reader();
-
- ::DBus::String argout1 = getDefaultAccount();
- ::DBus::ReturnMessage reply(call);
- ::DBus::MessageIter wi = reply.writer();
- wi << argout1;
- return reply;
- }
- ::DBus::Message _setDefaultAccount_stub( const ::DBus::CallMessage& call )
- {
- ::DBus::MessageIter ri = call.reader();
-
- ::DBus::String argin1; ri >> argin1;
- setDefaultAccount(argin1);
- ::DBus::ReturnMessage reply(call);
- return reply;
- }
::DBus::Message _sendRegister_stub( const ::DBus::CallMessage& call )
{
::DBus::MessageIter ri = call.reader();
diff --git a/src/dbus/configurationmanager-introspec.xml b/src/dbus/configurationmanager-introspec.xml
index 93b17644bf6b1ca65cf33014ed78c0280bbb4c76..6ebaef5a3494ac99c2803d66dfa1ad9135b6a75c 100644
--- a/src/dbus/configurationmanager-introspec.xml
+++ b/src/dbus/configurationmanager-introspec.xml
@@ -25,14 +25,6 @@
<arg type="as" name="list" direction="out"/>
</method>
- <method name="getDefaultAccount">
- <arg type="s" name="accountID" direction="out"/>
- </method>
-
- <method name="setDefaultAccount">
- <arg type="s" name="accountID" direction="in"/>
- </method>
-
<method name="sendRegister">
<arg type="s" name="accountID" direction="in"/>
<arg type="i" name="expire" direction="in"/>
diff --git a/src/dbus/configurationmanager.cpp b/src/dbus/configurationmanager.cpp
index c09f62858bfaad80885e50377406d90b64aeeb42..d5bb6c5ef2fce4f7bd4b287870b4d35a7e0b34d4 100644
--- a/src/dbus/configurationmanager.cpp
+++ b/src/dbus/configurationmanager.cpp
@@ -217,24 +217,6 @@ ConfigurationManager::getRecordDeviceList( )
{
_debug("ConfigurationManager::getRecordDeviceList received\n");
-}
-
- ::DBus::String
-ConfigurationManager::getDefaultAccount( )
-{
- _debug("ConfigurationManager::getDefaultAccount received\n");
- return Manager::instance().getDefaultAccount();
-}
-
-/*
- * used to set a default account
- */
- void
-ConfigurationManager::setDefaultAccount( const ::DBus::String& accountID )
-{
- _debug("ConfigurationManager::setDefaultAccount received\n");
- Manager::instance().setDefaultAccount(accountID);
-
}
::DBus::Int32
diff --git a/src/dbus/configurationmanager.h b/src/dbus/configurationmanager.h
index f5979b1b86705d517c350594e03ae2be1e4e676d..212374bfc50c6fa84dd7c48e52ce45622d44e1c9 100644
--- a/src/dbus/configurationmanager.h
+++ b/src/dbus/configurationmanager.h
@@ -44,8 +44,6 @@ public:
void addAccount( const std::map< ::DBus::String, ::DBus::String >& details );
void removeAccount( const ::DBus::String& accoundID );
std::vector< ::DBus::String > getAccountList( );
- ::DBus::String getDefaultAccount( );
- void setDefaultAccount( const ::DBus::String& accountID );
void sendRegister( const ::DBus::String& accoundID , const ::DBus::Int32& expire );
std::vector< ::DBus::String > getCodecList( );
diff --git a/src/eventthread.h b/src/eventthread.h
index 65daaf9003e2e95556463dbbf992faad206c57c7..21f43a897ac87fc9c639d7c9089acd98bff49a93 100644
--- a/src/eventthread.h
+++ b/src/eventthread.h
@@ -24,9 +24,12 @@
#include <cc++/thread.h>
class VoIPLink;
+
/**
- * General thread to listen events continuously
+ * @file eventthread.h
+ * @brief General thread to listen events continuously
*/
+
class EventThread : public ost::Thread {
public:
/**
diff --git a/src/global.h b/src/global.h
index 0252b420c0de019ed38b8834a3d2aa86c49f31a0..4fe6ded002f067f7a10b8ca91888c56b412b0a58 100644
--- a/src/global.h
+++ b/src/global.h
@@ -61,62 +61,61 @@ typedef short int16;
#define _debugAlsa(...)
#endif
-#define SFLPHONED_VERSION "0.8.2"
+#define SFLPHONED_VERSION "0.8.2" /** Version number */
#define SFLPHONED_VERSIONNUM 0x000802
-#define PROGNAME "sflphoned"
-#define PROGNAME_GLOBAL "sflphone"
-#define PROGDIR "sflphone"
-#define RINGDIR "ringtones"
-#define CODECDIR "codecs"
+#define PROGNAME "sflphoned" /** Binary name */
+#define PROGNAME_GLOBAL "sflphone" /** Program name */
+#define PROGDIR "sflphone" /** Program directory */
+#define RINGDIR "ringtones" /** Ringtones directory */
+#define CODECDIR "codecs" /** Codecs directory */
#define _(arg) arg
#define MONO 1
#define CHANNELS 2
#define SIZEBUF 1024*1024
-#define ALSA_DFT_CARD_ID 0
-
-#define PCM_HW "hw"
-#define PCM_PLUGHW "plughw"
-#define PCM_PULSE "pulse"
-#define PCM_FRONT "plug:front"
-#define PCM_DEFAULT "default"
-#define PCM_DMIX "plug:dmix"
-#define PCM_SURROUND40 "plug:surround40"
-#define PCM_SURROUND41 "plug:surround41"
-#define PCM_SURROUND50 "plug:surround50"
-#define PCM_SURROUND51 "plug:surround51"
-#define PCM_SURROUND71 "plug:surround71"
-#define PCM_IEC958 "plug:iec958"
-
-#define SFL_CODEC_VALID_PREFIX "libcodec_"
-#define SFL_CODEC_VALID_EXTEN ".so"
-#define CURRENT_DIR "."
-#define PARENT_DIR ".."
-
-#define SFL_PCM_BOTH 0x0021
-#define SFL_PCM_PLAYBACK 0x0022
-#define SFL_PCM_CAPTURE 0x0023
+#define ALSA_DFT_CARD_ID 0 /** Index of the default soundcard */
+
+#define PCM_HW "hw" /** Alsa plugin hardware */
+#define PCM_PLUGHW "plughw" /** Alsa plugin */
+#define PCM_PULSE "pulse" /** Alsa plugin for pulse audio */
+#define PCM_FRONT "plug:front" /** Alsa plugin: front PCM */
+#define PCM_DEFAULT "default" /** Default ALSA plugin */
+#define PCM_DMIX "plug:dmix" /** Alsa plugin for software mixing */
+#define PCM_SURROUND40 "plug:surround40" /** Alsa plugin: surround40 */
+#define PCM_SURROUND41 "plug:surround41" /** Alsa plugin: surround41 */
+#define PCM_SURROUND50 "plug:surround50" /** Alsa plugin: surround50 */
+#define PCM_SURROUND51 "plug:surround51" /** Alsa plugin: surround51 */
+#define PCM_SURROUND71 "plug:surround71" /** Alsa plugin: surround71 */
+
+#define SFL_CODEC_VALID_PREFIX "libcodec_" /** Valid prefix for codecs shared library */
+#define SFL_CODEC_VALID_EXTEN ".so" /** Valid extension for codecs shared library */
+#define CURRENT_DIR "." /** Current directory */
+#define PARENT_DIR ".." /** Parent directory */
+
+#define SFL_PCM_BOTH 0x0021 /** To open both playback and capture devices */
+#define SFL_PCM_PLAYBACK 0x0022 /** To open playback device only */
+#define SFL_PCM_CAPTURE 0x0023 /** To open capture device only */
#ifdef USE_IAX
-#define IAX2_ENABLED true
+#define IAX2_ENABLED true /** IAX2 support */
#else
-#define IAX2_ENABLED false
+#define IAX2_ENABLED false /** IAX2 support */
#endif
-#define GSM_STRING_DESCRIPTION "gsm"
-#define SPEEX_STRING_DESCRIPTION "speex"
-#define ILBC_STRING_DESCRIPTION "ilbc"
-#define RINGTONE_ENABLED 1
-#define DISPLAY_DIALPAD 1
-#define START_HIDDEN 1
-#define WINDOW_POPUP 1
+#define GSM_STRING_DESCRIPTION "gsm" /** GSM codec string description */
+#define SPEEX_STRING_DESCRIPTION "speex" /** SPEEX codec string description */
+#define ILBC_STRING_DESCRIPTION "ilbc" /** Ilbc codec string description */
+#define RINGTONE_ENABLED 1 /** Custom ringtone enable or not */
+#define DISPLAY_DIALPAD 1 /** Display dialpad or not */
+#define START_HIDDEN 1 /** SFlphone starts hidden at start-up or not */
+#define WINDOW_POPUP 1 /** Popup mode */
// Error codes for error handling
-#define NO_ERROR 0x0000
-#define ALSA_CAPTURE_DEVICE 0x0001
-#define ALSA_PLAYBACK_DEVICE 0x0010
-#define NETWORK_UNREACHABLE 0x0011
+#define NO_ERROR 0x0000 /** No error - Everything alright */
+#define ALSA_CAPTURE_DEVICE 0x0001 /** Error while opening capture device */
+#define ALSA_PLAYBACK_DEVICE 0x0010 /** Error while opening playback device */
+#define NETWORK_UNREACHABLE 0x0011 /** Network unreachable */
#endif // __GLOBAL_H__
diff --git a/src/iaxaccount.h b/src/iaxaccount.h
index 6e6472395ebb8e49942f9556fd569a3fe8b7699f..2cdfb6232ad0dc338cdb708d69bfdbb7ed34b7f2 100644
--- a/src/iaxaccount.h
+++ b/src/iaxaccount.h
@@ -24,8 +24,8 @@
/**
- * An IAX Account specify IAX specific functions and objects (IAXCall/IAXVoIPLink)
- * @author Yan Morin <yan.morin@gmail.com>
+ * @file: iaxaccount.h
+ * @brief: an IAX Account specify IAX specific functions and objects (IAXCall/IAXVoIPLink)
*/
class IAXAccount : public Account
{
@@ -34,9 +34,19 @@ public:
~IAXAccount();
- /** Actually unuseful, since config loading is done in init() */
+ /**
+ * Actually unuseful, since config loading is done in init()
+ */
void loadConfig();
+
+ /**
+ * Register an account
+ */
void registerVoIPLink();
+
+ /**
+ * Unregister an account
+ */
void unregisterVoIPLink();
private:
diff --git a/src/iaxcall.h b/src/iaxcall.h
index 2c6d612d867d8bfc6409e4a4b6108d90197a9089..ccfe8c41d5dd3e69b9ba9ef1089cf52c0b225a8f 100644
--- a/src/iaxcall.h
+++ b/src/iaxcall.h
@@ -25,42 +25,53 @@
#include <iax/frame.h>
/**
- * IAXCall are IAX implementation of a normal Call
- * @author Yan Morin <yan.morin@gmail.com>
+ * @file: iaxcall.h
+ * @brief: IAXCall are IAX implementation of a normal Call
*/
+
class IAXCall : public Call
{
public:
+ /**
+ * Constructor
+ * @param id The unique ID of the call
+ * @param type The type of the call
+ */
IAXCall(const CallID& id, Call::CallType type);
+ /**
+ * Destructor
+ */
~IAXCall();
- /** Get the session pointer or NULL */
+ /**
+ * @return iax_session* The session pointer or NULL
+ */
struct iax_session* getSession() { return _session; }
- /** Set the session pointer
+ /**
+ * Set the session pointer
* @param session the session pointer to assign
*/
void setSession(struct iax_session* session) { _session = session; }
/**
- * Set format (one single bit
- *
+ * Set format (one single bit)
* This function sets the _audioCodec variable with the correct
* codec.
+ * @param format The format representing the codec
*/
void setFormat(int format);
/**
* Get format for the voice codec used
- *
- * Bitmask for codecs defined in iax/frame.h
+ * @return int Bitmask for codecs defined in iax/frame.h
*/
int getFormat() { return _format; }
/**
- * Get the bitwise list of supported formats
+ * @return int The bitwise list of supported formats
*/
int getSupportedFormat();
@@ -75,7 +86,7 @@ public:
* in this call context.
*
* @param needles The format(s) (bitwise) you are looking for to match
- * @return The matching format, thus 0 if none matches
+ * @return int The matching format, thus 0 if none matches
*/
int getFirstMatchingFormat(int needles);
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 17607f4889c74e8652987ebf42e7589d7b43e930..510ba7642f75e2b19daadb9a2376078d5ef5fefe 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -363,11 +363,9 @@ IAXVoIPLink::sendRegister()
bool result = false;
if (_host.empty()) {
- Manager::instance().displayConfigError("Fill host field for IAX Account");
return false;
}
if (_user.empty()) {
- Manager::instance().displayConfigError("Fill user field for IAX Account");
return false;
}
@@ -656,7 +654,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
}
call->setConnectionState(Call::Connected);
call->setState(Call::Error);
- Manager::instance().displayErrorText(id, "Failure");
Manager::instance().callFailure(id);
removeCall(id);
break;
@@ -690,7 +687,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
case IAX_EVENT_BUSY:
call->setConnectionState(Call::Connected);
call->setState(Call::Busy);
- Manager::instance().displayErrorText(id, "Busy");
Manager::instance().callBusy(id);
removeCall(id);
break;
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 05f1a08735d3fe9350bd0a426f63f98db690ede2..9e2acd6c8d4948ee1533ef217ecfe22c3b3a736c 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -2,6 +2,7 @@
* Copyright (C) 2006-2007 Savoir-Faire Linux inc.
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -27,189 +28,285 @@
#include "audio/codecDescriptor.h"
-/** @todo Remove this fstream/iostream stuff */
-#include <fstream> // fstream + iostream for _fstream debugging...
-#include <iostream>
-
-
class EventThread;
class IAXCall;
class AudioCodec;
class AudioLayer;
-
/**
- * VoIPLink contains a thread that listen to external events
+ * @file iaxvoiplink.h
+ * @brief VoIPLink contains a thread that listen to external events
* and contains IAX Call related functions
- * @author Yan Morin <yan.morin@gmail.com>
*/
+
class IAXVoIPLink : public VoIPLink
{
-public:
+ public:
+
+ /**
+ * Constructor
+ * @param accountID The account containing the voip link
+ */
IAXVoIPLink(const AccountID& accountID);
+ /**
+ * Destructor
+ */
~IAXVoIPLink();
- void getEvent(void);
- bool init (void);
- bool checkNetwork (void) { return false; }
- void terminate (void);
-
- /**
- * Send out registration
- *
- * @return The new registration state (are we registered ?)
- */
- bool sendRegister (void);
-
- /**
- * Destroy registration session
- *
- * @todo Send an IAX_COMMAND_REGREL to force unregistration upstream.
- * Urgency: low
- *
- * @return bool If we're registered upstream
- */
- bool sendUnregister (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) { return false; }
- 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) { return false; }
- bool isContactPresenceSupported() { return false; }
-
-public: // iaxvoiplink only
- void setHost(const std::string& host) { _host = host; }
- void setUser(const std::string& user) { _user = user; }
- void setPass(const std::string& pass) { _pass = pass; }
-
-private:
- /**
- * Get IAX Call from an id
- * @param id CallId
- * @return IAXCall pointer or 0
- */
- IAXCall* getIAXCall(const CallID& id);
-
- /**
- * Delete every call
- */
- void terminateIAXCall();
-
- /**
- * Find a iaxcall by iax session number
- * @param session an iax_session valid pointer
- * @return iaxcall or 0 if not found
- */
- IAXCall* iaxFindCallBySession(struct iax_session* session);
-
- /**
- * Handle IAX Event for a call
- * @param event An iax_event pointer
- * @param call An IAXCall pointer
- */
- void iaxHandleCallEvent(iax_event* event, IAXCall* call);
-
- /**
- * Handle the VOICE events specifically
- * @param event The iax_event containing the IAX_EVENT_VOICE
- * @param call The associated IAXCall
- */
- void iaxHandleVoiceEvent(iax_event* event, IAXCall* call);
-
- /**
- * Handle IAX Registration Reply event
- * @param event An iax_event pointer
- */
- void iaxHandleRegReply(iax_event* event);
-
- /**
- * Handle IAX pre-call setup-related events
- * @param event An iax_event pointer
- */
- void iaxHandlePrecallEvent(iax_event* event);
-
- /**
- * Work out the audio data from Microphone to IAX2 channel
- */
- void sendAudioFromMic(void);
-
- /**
- * Send an outgoing call invite to iax
- * @param call An IAXCall pointer
- */
- bool iaxOutgoingInvite(IAXCall* call);
-
-
- /**
- * Convert CodecMap to IAX format using IAX constants
- * @return `format` ready to go into iax_* calls
- */
- int iaxCodecMapToFormat(IAXCall* call);
-
- /** Threading object */
- EventThread* _evThread;
-
- /** registration session : 0 if not register */
- struct iax_session* _regSession;
-
- /** IAX Host */
- std::string _host;
-
- /** IAX User */
- std::string _user;
-
- /** IAX Password */
- std::string _pass;
-
- /** IAX full name */
- std::string _fullName;
-
- /** Timestamp of when we should refresh the registration up with
- * the registrar. Values can be: EPOCH timestamp, 0 if we want no registration, 1
- * to force a registration. */
- int _nextRefreshStamp;
-
- /** Mutex for iax_ calls, since we're the only one dealing with the incorporated
- * iax_stuff inside this class. */
- ost::Mutex _mutexIAX;
-
- /** Connection to audio card/device */
- AudioLayer* audiolayer;
-
- /** When we receive data, we decode it inside this buffer */
- int16* _receiveDataDecoded;
- /** When we send data, we encode it inside this buffer*/
- unsigned char* _sendDataEncoded;
-
- /** After that we send the data inside this buffer if there is a format conversion or rate conversion. */
- /* Also use for getting mic-ringbuffer data */
- SFLDataFormat* _dataAudioLayer;
-
- /** Buffer for 8000hz samples in conversion */
- float32* _floatBuffer8000;
- /** Buffer for 48000hz samples in conversion */
- float32* _floatBuffer48000;
-
- /** Buffer for 8000hz samples for mic conversion */
- int16* _intBuffer8000;
-
- /** libsamplerate converter for incoming voice */
- SRC_STATE* _src_state_spkr;
-
- /** libsamplerate converter for outgoing voice */
- SRC_STATE* _src_state_mic;
-
- /** libsamplerate error */
- int _src_err;
+ /**
+ * Listen to events sent by the call manager ( asterisk, etc .. )
+ */
+ void getEvent(void);
+
+ /**
+ * Init the voip link
+ * @return true if successful
+ * false otherwise
+ */
+ bool init (void);
+
+ /**
+ * Check if a local IP can be found
+ * @return true if pingable
+ * false otherwise
+ */
+ bool checkNetwork (void) { return false; }
+
+ /**
+ * Terminate a voip link by clearing the call list
+ */
+ void terminate (void);
+
+ /**
+ * Send out registration
+ * @return bool The new registration state (are we registered ?)
+ */
+ bool sendRegister (void);
+
+ /**
+ * Destroy registration session
+ * @todo Send an IAX_COMMAND_REGREL to force unregistration upstream.
+ * Urgency: low
+ * @return bool true if we're registered upstream
+ * false otherwise
+ */
+ bool sendUnregister (void);
+
+ /**
+ * Create a new outgoing call
+ * @param id The ID of the call
+ * @param toUrl The address to call
+ * @return Call* A pointer on the call
+ */
+ Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
+
+ /**
+ * Answer a call
+ * @param id The ID of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool answer(const CallID& id);
+
+ /**
+ * Hangup a call
+ * @param id The ID of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool hangup(const CallID& id);
+
+ /**
+ * Cancel a call
+ * @param id The ID of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool cancel(const CallID& id) { return false; }
+
+ /**
+ * Put a call on hold
+ * @param id The ID of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool onhold(const CallID& id);
+
+ /**
+ * Put a call off hold
+ * @param id The ID of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool offhold(const CallID& id);
+
+ /**
+ * Transfer a call
+ * @param id The ID of the call
+ * @param to The recipient of the transfer
+ * @return bool true on success
+ * false otherwise
+ */
+ bool transfer(const CallID& id, const std::string& to);
+
+ /**
+ * Refuse a call
+ * @param id The ID of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool refuse (const CallID& id);
+
+ /**
+ * Send DTMF
+ * @param id The ID of the call
+ * @param code The code of the DTMF
+ * @return bool true on success
+ * false otherwise
+ */
+ bool carryingDTMFdigits(const CallID& id, char code);
+
+ bool sendMessage(const std::string& to, const std::string& body) { return false; }
+
+ bool isContactPresenceSupported() { return false; }
+
+ public: // iaxvoiplink only
+ /**
+ * @param host Set the host name
+ */
+ void setHost(const std::string& host) { _host = host; }
+
+ /**
+ * @param user Set the user name
+ */
+ void setUser(const std::string& user) { _user = user; }
+
+ /**
+ * @param pass Set the password
+ */
+ void setPass(const std::string& pass) { _pass = pass; }
+
+ private:
+ /**
+ * Get IAX Call from an id
+ * @param id CallId
+ * @return IAXCall pointer or 0
+ */
+ IAXCall* getIAXCall(const CallID& id);
+
+ /**
+ * Delete every call
+ */
+ void terminateIAXCall();
+
+ /**
+ * Find a iaxcall by iax session number
+ * @param session an iax_session valid pointer
+ * @return iaxcall or 0 if not found
+ */
+ IAXCall* iaxFindCallBySession(struct iax_session* session);
+
+ /**
+ * Handle IAX Event for a call
+ * @param event An iax_event pointer
+ * @param call An IAXCall pointer
+ */
+ void iaxHandleCallEvent(iax_event* event, IAXCall* call);
+
+ /**
+ * Handle the VOICE events specifically
+ * @param event The iax_event containing the IAX_EVENT_VOICE
+ * @param call The associated IAXCall
+ */
+ void iaxHandleVoiceEvent(iax_event* event, IAXCall* call);
+
+ /**
+ * Handle IAX Registration Reply event
+ * @param event An iax_event pointer
+ */
+ void iaxHandleRegReply(iax_event* event);
+
+ /**
+ * Handle IAX pre-call setup-related events
+ * @param event An iax_event pointer
+ */
+ void iaxHandlePrecallEvent(iax_event* event);
+
+ /**
+ * Work out the audio data from Microphone to IAX2 channel
+ */
+ void sendAudioFromMic(void);
+
+ /**
+ * Send an outgoing call invite to iax
+ * @param call An IAXCall pointer
+ */
+ bool iaxOutgoingInvite(IAXCall* call);
+
+
+ /**
+ * Convert CodecMap to IAX format using IAX constants
+ * @return `format` ready to go into iax_* calls
+ */
+ int iaxCodecMapToFormat(IAXCall* call);
+
+ /** Threading object */
+ EventThread* _evThread;
+
+ /** registration session : 0 if not register */
+ struct iax_session* _regSession;
+
+ /** IAX Host */
+ std::string _host;
+
+ /** IAX User */
+ std::string _user;
+
+ /** IAX Password */
+ std::string _pass;
+
+ /** IAX full name */
+ std::string _fullName;
+
+ /** Timestamp of when we should refresh the registration up with
+ * the registrar. Values can be: EPOCH timestamp, 0 if we want no registration, 1
+ * to force a registration. */
+ int _nextRefreshStamp;
+
+ /** Mutex for iax_ calls, since we're the only one dealing with the incorporated
+ * iax_stuff inside this class. */
+ ost::Mutex _mutexIAX;
+
+ /** Connection to audio card/device */
+ AudioLayer* audiolayer;
+
+ /** When we receive data, we decode it inside this buffer */
+ int16* _receiveDataDecoded;
+ /** When we send data, we encode it inside this buffer*/
+ unsigned char* _sendDataEncoded;
+
+ /** After that we send the data inside this buffer if there is a format conversion or rate conversion. */
+ /* Also use for getting mic-ringbuffer data */
+ SFLDataFormat* _dataAudioLayer;
+
+ /** Buffer for 8000hz samples in conversion */
+ float32* _floatBuffer8000;
+ /** Buffer for 48000hz samples in conversion */
+ float32* _floatBuffer48000;
+
+ /** Buffer for 8000hz samples for mic conversion */
+ int16* _intBuffer8000;
+
+ /** libsamplerate converter for incoming voice */
+ SRC_STATE* _src_state_spkr;
+
+ /** libsamplerate converter for outgoing voice */
+ SRC_STATE* _src_state_mic;
+
+ /** libsamplerate error */
+ int _src_err;
};
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 725e38b3b827016469e8ad98447fe879708aafd6..a112f240c90061c638cd23a32ee1b02d2824d779 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -77,7 +77,6 @@ ManagerImpl::ManagerImpl (void)
_dtmfKey = 0;
_spkr_volume = 0; // Initialize after by init() -> initVolume()
_mic_volume = 0; // Initialize after by init() -> initVolume()
- _mic_volume_before_mute = 0;
// Call
_nbIncomingWaitingCall=0;
@@ -229,10 +228,6 @@ ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const
ManagerImpl::answerCall(const CallID& id)
{
stopTone(false);
- /*if (hasCurrentCall())
- {
- onHoldCall(getCurrentCallId());
- }*/
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
_debug("Answering Call: Call doesn't exists\n");
@@ -245,9 +240,6 @@ ManagerImpl::answerCall(const CallID& id)
return false;
}
- //Place current call on hold if it isn't
-
-
// if it was waiting, it's waiting no more
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "CURRENT");
removeWaitingCall(id);
@@ -283,7 +275,6 @@ ManagerImpl::hangupCall(const CallID& id)
removeCallAccount(id);
switchCall("");
-
return returnValue;
}
@@ -351,13 +342,7 @@ ManagerImpl::offHoldCall(const CallID& id)
bool returnValue = getAccountLink(accountid)->offhold(id);
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "UNHOLD");
switchCall(id);
- if (returnValue) {
- try {
- //getAudioDriver()->startStream();
- } catch(...) {
- _debugException("! Manager Off hold could not start audio stream");
- }
- }
+
return returnValue;
}
@@ -376,26 +361,9 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
removeCallAccount(id);
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "HUNGUP");
switchCall("");
-
-
return returnValue;
}
-//THREAD=Main
-void
-ManagerImpl::mute() {
- _mic_volume_before_mute = _mic_volume;
- setMicVolume(0);
-}
-
-//THREAD=Main
-void
-ManagerImpl::unmute() {
- if ( _mic_volume == 0 ) {
- setMicVolume(_mic_volume_before_mute);
- }
-}
-
//THREAD=Main : Call:Incoming
bool
ManagerImpl::refuseCall (const CallID& id)
@@ -455,46 +423,6 @@ ManagerImpl::initRegisterAccounts()
return true;
}
-//THREAD=Main
-// Currently unused
- bool
-ManagerImpl::registerAccount(const AccountID& accountId)
-{
- _debug("Register one VoIP Link\n");
-
- // right now, we don't support two SIP account
- // so we close everything before registring a new account
- Account* account = getAccount(accountId);
- if (account != 0) {
- AccountMap::iterator iter = _accountMap.begin();
- while ( iter != _accountMap.end() ) {
- if ( iter->second ) {
- iter->second->unregisterVoIPLink();
- }
- iter++;
- }
- // NOW
- account->registerVoIPLink();
- account->loadContacts();
- account->publishPresence(PRESENCE_ONLINE);
- account->subscribeContactsPresence();
- }
- return true;
-}
-
-//THREAD=Main
-// Currently unused
- bool
-ManagerImpl::unregisterAccount(const AccountID& accountId)
-{
- _debug("Unregister one VoIP Link\n");
-
- if (accountExists( accountId ) ) {
- getAccount(accountId)->unregisterVoIPLink();
- }
- return true;
-}
-
//THREAD=Main
bool
ManagerImpl::sendDtmf(const CallID& id, char code)
@@ -567,15 +495,6 @@ ManagerImpl::playDtmf(char code, bool isTalking)
// put the size in bytes...
// so size * 1 channel (mono) * sizeof (bytes for the data)
audiolayer->playSamples(_buf, size * sizeof(SFLDataFormat), isTalking);
- //audiolayer->putUrgent(_buf, size * sizeof(SFLDataFormat));
-
- // We activate the stream if it's not active yet.
- //if (!audiolayer->isStreamActive()) {
- //audiolayer->startStream();
- //} else {
- //_debugAlsa("send dtmf - sleep\n");
- //audiolayer->sleep(pulselen); // in milliseconds
- //}
}
returnValue = true;
@@ -617,8 +536,6 @@ ManagerImpl::isWaitingCall(const CallID& id) {
return true;
}
-
-
///////////////////////////////////////////////////////////////////////////////
// Management of event peer IP-phone
////////////////////////////////////////////////////////////////////////////////
@@ -730,53 +647,6 @@ ManagerImpl::callFailure(const CallID& id)
}
-//THREAD=VoIP
- void
-ManagerImpl::displayTextMessage(const CallID& id, const std::string& message)
-{
- /*if(_gui) {
- _gui->displayTextMessage(id, message);
- }*/
-}
-
-//THREAD=VoIP
- void
-ManagerImpl::displayErrorText(const CallID& id, const std::string& message)
-{
- /*if(_gui) {
- _gui->displayErrorText(id, message);
- } else {
- std::cerr << message << std::endl;
- }*/
-}
-
-//THREAD=VoIP
- void
-ManagerImpl::displayError (const std::string& error)
-{
- /*if(_gui) {
- _gui->displayError(error);
- }*/
-}
-
-//THREAD=VoIP
- void
-ManagerImpl::displayStatus(const std::string& status)
-{
- /*if(_gui) {
- _gui->displayStatus(status);
- }*/
-}
-
-//THREAD=VoIP
- void
-ManagerImpl::displayConfigError (const std::string& message)
-{
- /*if(_gui) {
- _gui->displayConfigError(message);
- }*/
-}
-
//THREAD=VoIP
void
ManagerImpl::startVoiceMessageNotification(const AccountID& accountId, const std::string& nb_msg)
@@ -784,14 +654,6 @@ ManagerImpl::startVoiceMessageNotification(const AccountID& accountId, const std
if (_dbus) _dbus->getCallManager()->voiceMailNotify(accountId, atoi(nb_msg.c_str()) );
}
-//THREAD=VoIP
- void
-ManagerImpl::stopVoiceMessageNotification(const AccountID& accountId)
-{
- // TODO : do not notify when no messages
- //if (_dbus) _dbus->getCallManager()->voiceMailNotify(accountId, 0 );
-}
-
//THREAD=VoIP
void
ManagerImpl::registrationSucceed(const AccountID& accountid)
@@ -924,7 +786,6 @@ ManagerImpl::ringback () {
void
ManagerImpl::ringtone()
{
- // int hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
if( isRingtoneEnabled() )
{
std::string ringchoice = getConfigString(AUDIO, RING_CHOICE);
@@ -983,11 +844,6 @@ ManagerImpl::getTelephoneFile()
}
}
-
-/**
- * Use Urgent Buffer
- * By AudioRTP thread
- */
void
ManagerImpl::notificationIncomingCall(void) {
@@ -1169,7 +1025,7 @@ ManagerImpl::retrieveActiveCodecs()
}
void
-ManagerImpl::setActiveCodecList(const std::vector<std::string>& list)
+ManagerImpl::setActiveCodecList(const std::vector< ::DBus::String >& list)
{
_debug("Set active codecs list\n");
_codecDescriptorMap.saveActiveCodecs(list);
@@ -1491,7 +1347,7 @@ ManagerImpl::popupMode( void )
}
void
-ManagerImpl::notifyErrClient( const int& errCode )
+ManagerImpl::notifyErrClient( const ::DBus::Int32& errCode )
{
if( _dbus ){
_debug("NOTIFY ERR NUMBER %i\n" , errCode);
@@ -1680,20 +1536,6 @@ ManagerImpl::detachZeroconfEvents(Pattern::Observer& observer)
return returnValue;
}
-/**
- * Main Thread
- *
- * @todo When is this called ? Why this name 'getEvents' ?
- */
-/**
- * DEPRECATED
- bool
- ManagerImpl::getEvents() {
- initRegisterAccounts();
- return true;
- }
- */
-
// TODO: rewrite this
/**
* Main Thread
@@ -1746,7 +1588,6 @@ ManagerImpl::getCallStatus(const std::string& sequenceId)
tk.push_back(iter->second);
tk.push_back(destination);
tk.push_back(status);
- //_gui->sendCallMessage(code, sequenceId, iter->first, tk);
tk.clear();
iter++;
@@ -1755,23 +1596,6 @@ ManagerImpl::getCallStatus(const std::string& sequenceId)
return true;
}
-//THREAD=Main
- bool
-ManagerImpl::getConfigAll(const std::string& sequenceId)
-{
- bool returnValue = false;
- Conf::ConfigTreeIterator iter = _config.createIterator();
- TokenList tk = iter.begin();
- if (tk.size()) {
- returnValue = true;
- }
- while (tk.size()) {
- //_gui->sendMessage("100", sequenceId, tk);
- tk = iter.next();
- }
- return returnValue;
-}
-
//THREAD=Main
bool
ManagerImpl::getConfig(const std::string& section, const std::string& name, TokenList& arg)
@@ -1821,99 +1645,6 @@ ManagerImpl::setConfig(const std::string& section, const std::string& name, int
return _config.setConfigTreeItem(section, name, valueStream.str());
}
-//THREAD=Main
- bool
-ManagerImpl::getAudioDeviceList(const std::string& sequenceId, int ioDeviceMask)
-{
- AudioLayer* audiolayer = getAudioDriver();
- if (audiolayer == 0) { return false; }
-
- bool returnValue = false;
-
- // TODO: test when there is an error on initializing...
- TokenList tk;
- AudioDevice* device = 0;
- int nbDevice = audiolayer->getDeviceCount();
- /*
- for (int index = 0; index < nbDevice; index++ ) {
- device = audiolayer->getAudioDeviceInfo(index, ioDeviceMask);
- if (device != 0) {
- tk.clear();
- std::ostringstream str; str << index; tk.push_back(str.str());
- tk.push_back(device->getName());
- tk.push_back(device->getApiName());
- std::ostringstream rate; rate << (int)(device->getRate()); tk.push_back(rate.str());
- //_gui->sendMessage("100", sequenceId, tk);
-
- // don't forget to delete it after
- delete device; device = 0;
- }
- }*/
- returnValue = true;
-
- std::ostringstream rate;
- rate << "VARIABLE";
- tk.clear();
- tk.push_back(rate.str());
- //_gui->sendMessage("101", sequenceId, tk);
-
- return returnValue;
-}
-
-//THREAD=Main
- bool
-ManagerImpl::getCountryTones(const std::string& sequenceId)
-{
- // see ToneGenerator for the list...
- sendCountryTone(sequenceId, 1, "North America");
- sendCountryTone(sequenceId, 2, "France");
- sendCountryTone(sequenceId, 3, "Australia");
- sendCountryTone(sequenceId, 4, "United Kingdom");
- sendCountryTone(sequenceId, 5, "Spain");
- sendCountryTone(sequenceId, 6, "Italy");
- sendCountryTone(sequenceId, 7, "Japan");
-
- return true;
-}
-
-//THREAD=Main
-void
-ManagerImpl::sendCountryTone(const std::string& sequenceId, int index, const std::string& name) {
- TokenList tk;
- std::ostringstream str; str << index; tk.push_back(str.str());
- tk.push_back(name);
- //_gui->sendMessage("100", sequenceId, tk);
-}
-
-//THREAD=Main
-bool
-ManagerImpl::getDirListing(const std::string& sequenceId, const std::string& path, int *nbFile) {
- TokenList tk;
- try {
- ost::Dir dir(path.c_str());
- const char *cFileName = 0;
- std::string fileName;
- std::string filePathName;
- while ( (cFileName=dir++) != 0 ) {
- fileName = cFileName;
- filePathName = path + DIR_SEPARATOR_STR + cFileName;
- if (fileName.length() && fileName[0]!='.' && !ost::isDir(filePathName.c_str())) {
- tk.clear();
- std::ostringstream str;
- str << (*nbFile);
- tk.push_back(str.str());
- tk.push_back(filePathName);
- //_gui->sendMessage("100", sequenceId, tk);
- (*nbFile)++;
- }
- }
- return true;
- } catch (...) {
- // error to open file dir
- return false;
- }
-}
-
std::vector< std::string >
ManagerImpl::getAccountList()
{
@@ -1932,8 +1663,6 @@ ManagerImpl::getAccountList()
return v;
}
-
-
std::map< std::string, std::string >
ManagerImpl::getAccountDetails(const AccountID& accountID)
{
@@ -1949,13 +1678,6 @@ ManagerImpl::getAccountDetails(const AccountID& accountID)
getConfigString(accountID, CONFIG_ACCOUNT_ALIAS)
)
);
- /*a.insert(
- std::pair<std::string, std::string>(
- CONFIG_ACCOUNT_AUTO_REGISTER,
- getConfigString(accountID, CONFIG_ACCOUNT_AUTO_REGISTER)== "1" ? "TRUE": "FALSE"
- )
- );*/
- _debug("Get account details: %s\n" , getConfigString(accountID, CONFIG_ACCOUNT_ENABLE).c_str());
a.insert(
std::pair<std::string, std::string>(
CONFIG_ACCOUNT_ENABLE,
@@ -2112,18 +1834,11 @@ ManagerImpl::setAccountDetails( const ::DBus::String& accountID,
if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
}
-/*
- * register if it was just enabled, and we hadn't registered
- * unregister if it was enabled/registered, and we want it closed
- * unregister <=> expire = 0 ( FALSE )
- * register <=> expire = 1 ( TRUE )
- */
void
ManagerImpl::sendRegister( const ::DBus::String& accountID , bool expire )
{
// Update the active field
setConfig( accountID, CONFIG_ACCOUNT_ENABLE, expire );
-
Account* acc = getAccount(accountID);
acc->loadConfig();
@@ -2189,90 +1904,6 @@ ManagerImpl::removeAccount(const AccountID& accountID)
if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
}
- std::string
-ManagerImpl::getDefaultAccount()
-{
-
- std::string id;
- id = getConfigString(PREFERENCES, "DefaultAccount");
- _debug("Default Account = %s\n",id.c_str());
- return id;
-}
-
- void
-ManagerImpl::setDefaultAccount(const AccountID& accountID)
-{
- // we write into the Preferences section the field Default
- setConfig("Preferences", "DefaultAccount", accountID);
-}
-
-
-
-
-//THREAD=Main
-/*
- * Experimental...
- */
-bool
-ManagerImpl::setSwitch(const std::string& switchName, std::string& message) {
- AudioLayer* audiolayer = 0;
- if (switchName == "audiodriver" ) {
- // hangup all call here
- audiolayer = getAudioDriver();
-
- int oldSampleRate = 0;
- if (audiolayer) { oldSampleRate = audiolayer->getSampleRate(); }
-
- selectAudioDriver();
- audiolayer = getAudioDriver();
-
- if (audiolayer) {
- int error = audiolayer->getErrorMessage();
- int newSampleRate = audiolayer->getSampleRate();
-
- if (error == -1) {
- message = error;
- return false;
- }
-
- if (newSampleRate != oldSampleRate) {
- _toneMutex.enterMutex();
-
- _debug("Unload Telephone Tone\n");
- delete _telephoneTone; _telephoneTone = NULL;
- _debug("Unload DTMF Key\n");
- delete _dtmfKey; _dtmfKey = NULL;
-
- _debug("Load Telephone Tone\n");
- std::string country = getConfigString(PREFERENCES, ZONE_TONE);
- _telephoneTone = new TelephoneTone(country, newSampleRate);
-
- _debugInit("Loading DTMF key");
- _dtmfKey = new DTMF(newSampleRate);
-
- _toneMutex.leaveMutex();
- }
-
- message = _("Change with success");
- playDtmf('9', true);
- //getAudioDriver()->sleep(300); // in milliseconds
- playDtmf('1', true);
- //getAudioDriver()->sleep(300); // in milliseconds
- playDtmf('1', true);
- return true;
- }
- } else if ( switchName == "echo" ) {
- audiolayer = getAudioDriver();
- if (audiolayer) {
- audiolayer->toggleEchoTesting();
- return true;
- }
- }
-
-
- return false;
-}
-
// ACCOUNT handling
bool
ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accountID)
@@ -2290,22 +1921,6 @@ ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accou
}
}
- AccountID
- ManagerImpl::getAccountFromEvent( std::string authname )
-{
- AccountID id;
- AccountMap::iterator iter = _accountMap.begin();
- while( iter != _accountMap.end() ){
- if( iter -> second != NULL ){
- id = iter -> first ;
- if(getConfigString( id , SIP_AUTH_NAME ) == authname )
- return id;
- }
- }
- return NULL;
-}
-
-
AccountID
ManagerImpl::getAccountFromCall(const CallID& callID)
{
@@ -2353,7 +1968,6 @@ ManagerImpl::loadAccountMap()
std::string accountType;
Account* tmpAccount;
- // iter = std::string
TokenList::iterator iter = sections.begin();
while(iter != sections.end()) {
// Check if it starts with "Account:" (SIP and IAX pour le moment)
@@ -2384,37 +1998,6 @@ ManagerImpl::loadAccountMap()
iter++;
}
-
- /*
- // SIP Loading X account...
- short nbAccountSIP = ACCOUNT_SIP_COUNT_DEFAULT;
- for (short iAccountSIP = 0; iAccountSIP<nbAccountSIP; iAccountSIP++) {
- std::ostringstream accountName;
- accountName << "SIP" << iAccountSIP;
-
- tmpAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, accountName.str());
- if (tmpAccount!=0) {
- _debugMid(" %s", accountName.str().data());
- _accountMap[accountName.str()] = tmpAccount;
- nbAccount++;
- }
- }
-
- // IAX Loading X account...
- short nbAccountIAX = ACCOUNT_IAX_COUNT_DEFAULT;
- for (short iAccountIAX = 0; iAccountIAX<nbAccountIAX; iAccountIAX++) {
- std::ostringstream accountName;
- accountName << "IAX" << iAccountIAX;
- tmpAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, accountName.str());
- if (tmpAccount!=0) {
- _debugMid(" %s", accountName.str().data());
- _accountMap[accountName.str()] = tmpAccount;
- nbAccount++;
- }
- }
- _debugEnd("\n");
- */
-
return nbAccount;
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index 378df590872648b926d5f5b91d0294b0d1c371a7..d2c2bb84b451a8bf80501d20e07622c2137f21c5 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -23,8 +23,6 @@
#ifndef __MANAGER_H__
#define __MANAGER_H__
-//#define TEST
-
#include <string>
#include <vector>
#include <set>
@@ -44,8 +42,6 @@
#include "audio/dtmf.h"
#include "audio/codecDescriptor.h"
-
-
class AudioLayer;
class CodecDescriptor;
class GuiFramework;
@@ -56,677 +52,893 @@ class VoIPLink;
class DNSService;
#endif
-/**
- * Define a type for a AccountMap container
- */
+/** Define a type for a AccountMap container */
typedef std::map<AccountID, Account*> AccountMap;
-
-/**
- * Define a type for a CallID to AccountID Map inside ManagerImpl
- */
+
+/** Define a type for a CallID to AccountID Map inside ManagerImpl */
typedef std::map<CallID, AccountID> CallAccountMap;
-/**
- * Define a type for CallID vector (waiting list, incoming not answered)
- */
+/** Define a type for CallID vector (waiting list, incoming not answered) */
typedef std::set<CallID> CallIDSet;
-/**
- * To send multiple string
- */
+/** To send multiple string */
typedef std::list<std::string> TokenList;
-/**
- * Manager (controller) of sflphone daemon
- */
+/** Manager (controller) of sflphone daemon */
class ManagerImpl {
-public:
- ManagerImpl (void);
- ~ManagerImpl (void);
-
- /**
- * Initialisation of thread (sound) and map.
- *
- * Init a new VoIPLink, audio codec and audio driver
- */
- void init (void);
-
- /**
- * Terminate all thread (sound, link) and unload AccountMap
- */
- void terminate (void);
-
- /**
- * Set user interface manaager.
- * @param man The DBUS interface implementation
- */
- void setDBusManager (DBusManagerImpl* man) { _dbus = man; }
-
- /**
- * Accessor to audiodriver.
- *
- * it's multi-thread and use mutex internally
- */
- AudioLayer* getAudioDriver(void) const { return _audiodriver; }
-
- /**
- * Get a descriptor map of codec available
- */
- CodecDescriptor& getCodecDescriptorMap(void) {return _codecDescriptorMap;}
-
- /**
- * Functions which occur with a user's action
- */
- 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();
- bool refuseCall(const CallID& id);
-
- /** Save config to file */
- bool saveConfig (void);
-
- /**
- * Send registration information (shake hands) for a specific AccountID
- *
- * @param accountId Account to register
- * @return true if sendRegister was called without failure, else return false
- */
- bool registerAccount(const AccountID& accountId);
-
- /**
- * Send unregistration for a specific account. If the protocol
- * doesn't need to send anything, then the state of the account
- * will be set to 'Unregistered', and related objects destroyed.
- *
- * @param accountId Account to unregister
- * @return true if the unregister method is send correctly
- */
- bool unregisterAccount(const AccountID& accountId);
-
- /**
- * Send registration to all enabled accounts
- *
- * @return false if exosip or the network checking fails
- */
- bool initRegisterAccounts();
-
- /**
- * True if we tried to register Once
- */
- bool _hasTriedToRegister;
-
-
- /**
- * Undocumented
- */
- 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(const CallID& id, char code);
- bool playDtmf(char code, bool isTalking);
- bool playTone ();
- bool playToneWithMessage ();
- void stopTone(bool stopAudio/*=true*/);
-
- // 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 AccountID& accountId, const std::string& nb_msg);
- void stopVoiceMessageNotification(const AccountID& accountId);
-
- /** Notify the user that registration succeeded */
- void registrationSucceed(const AccountID& accountId);
- /** Notify the user that unregistration succeeded */
- void unregistrationSucceed(const AccountID& accountId);
- /** Notify the user that registration failed */
- void registrationFailed(const AccountID& accountId);
- /** Notify the user that registration is trying */
- void registrationTrying(const AccountID& accountId);
- void sendRegister( const AccountID& accountId , bool expire );
- // configuration function requests
-
- /**
- * Start events thread. This function actually only calls the private
- * initRegisterVoIPLink().
- *
- * This function should definitively be renamed!
- *
- * @todo Receive account name (???)
- *
- * DEPRECATED
- */
- //bool getEvents();
-
- //
- bool getZeroconf(const std::string& sequenceId);
- bool attachZeroconfEvents(const std::string& sequenceId, Pattern::Observer& observer);
- bool detachZeroconfEvents(Pattern::Observer& observer);
- bool getCallStatus(const std::string& sequenceId);
-
- /**
- * Get account list
- * @return A list of accoundIDs
- */
- std::vector< std::string > getAccountList();
-
- /**
- * Retrieve details about a given account
- */
- std::map< std::string, std::string > getAccountDetails(const AccountID& accountID);
-
- /**
- * Save the details of an existing account, given the account ID
- *
- * This will load the configuration map with the given data.
- * It will also register/unregister links where the 'Enabled' switched.
- */
- void setAccountDetails( const ::DBus::String& accountID,
- const std::map< ::DBus::String, ::DBus::String >& details );
-
- /**
- * Add a new account, and give it a new account ID automatically
- */
- void addAccount(const std::map< ::DBus::String, ::DBus::String >& details);
-
- /**
- * Delete an existing account, unregister VoIPLink associated, and
- * purge from configuration.
- */
- void removeAccount(const AccountID& accountID);
-
- /*
- * Get the default account
- * @return The default account
- */
- std::string getDefaultAccount();
-
-
- /**
- * Get the list of codecs we supports, not ordered
- * @return The list of the codecs
- */
- std::vector< ::DBus::String > getCodecList( void );
- /**
- * Get the info about one codec
- * Name / CLock rate / bitrate / bandwidth
- * @param payload The payload of the codec
- * @return The information
- */
- std::vector< ::DBus::String > getCodecDetails( const ::DBus::Int32& payload);
-
- /**
- * Get a list of supported input audio plugin
- * @return List of names
- */
- std::vector< std::string> getInputAudioPluginList(void);
-
- /**
- * Get a list of supported output audio plugin
- * @return List of names
- */
- std::vector< std::string> getOutputAudioPluginList(void);
-
- /**
- * Set input audio plugin
- */
- void setInputAudioPlugin(const std::string& audioPlugin);
-
- /**
- * Set output audio plugin
- */
- void setOutputAudioPlugin(const std::string& audioPlugin);
-
- /**
- * Get list of supported audio output device
- */
- std::vector<std::string> getAudioOutputDeviceList(void);
-
- /**
- * Set audio output device
- */
- void setAudioOutputDevice(const int index);
-
- /**
- * Get list of supported audio input device
- */
- std::vector<std::string> getAudioInputDeviceList(void);
-
- /**
- * Set audio input device
- */
- void setAudioInputDevice(const int index);
-
- /**
- * Get string array representing integer indexes of output and input device
- */
- std::vector<std::string> getCurrentAudioDevicesIndex();
-
- /**
- * Get index of an audio device
- */
- int getAudioDeviceIndex( const std::string name );
-
- /*
- * Get current alsa plugin
- */
- std::string getCurrentAudioOutputPlugin( void );
-
- /**
- * Convert a list of payload in a special format, readable by the server.
- * Required format: payloads separated with one slash.
- * @return std::string The serializabled string
- */
- std::string serialize(std::vector<std::string> v);
-
- int isIax2Enabled( void );
- int isRingtoneEnabled( void );
- void ringtoneEnabled( void );
- std::string getRingtoneChoice( void );
- void setRingtoneChoice( const std::string& );
- int getDialpad( void );
- void setDialpad( void );
- int isStartHidden( void );
- void startHidden( void );
- int popupMode( void );
- void switchPopupMode( void );
-
-
- /**
- * Inverse of serialize
- */
- std::vector<std::string> retrieveActiveCodecs( void );
-
- /**
- * Get and set the list of the active codecs
- */
- std::vector< ::DBus::String > getActiveCodecList( void );
- void setActiveCodecList( const std::vector< ::DBus::String >& list);
-
- /*
- * Set an account as default
- * @param The ID of the account we want to set as default
- */
- void setDefaultAccount(const AccountID& accountID);
-
- /*
- * Notify the client that an error occured
- * @param errMsg The error message that should popup on the client side
- */
- void notifyErrClient( const ::DBus::Int32& errCode );
-
- bool getConfigAll(const std::string& sequenceId);
- bool getConfig(const std::string& section, const std::string& name, TokenList& arg);
- bool setConfig(const std::string& section, const std::string& name, const std::string& value);
- 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
- * @param sflphoned internal parameter to change
- * @param message to return to the user
- * @return true if everything is ok
- */
- bool setSwitch(const std::string& switchName, std::string& message);
-
- // 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(const CallID& id);
- void callFailure(const CallID& id);
-
- /** @return 0 if no tone (init before calling this function) */
- AudioLoop* getTelephoneTone();
- /** @return 0 if the wav is stopped */
- AudioLoop* getTelephoneFile();
-
- /**
- * @return true is there is one or many incoming call waiting
- * new call, not anwsered or refused
- */
- bool incomingCallWaiting(void);
- /**
- * Notification of incoming call when you are already busy
- */
- void notificationIncomingCall(void);
-
-
- /*
- * Inline functions to manage volume control
- * Read by main thread and AudioLayer thread
- * Write by main thread only
- */
- unsigned short getSpkrVolume(void) { return _spkr_volume; }
- void setSpkrVolume(unsigned short spkr_vol) { _spkr_volume = spkr_vol; }
- unsigned short getMicVolume(void) { return _mic_volume; }
- void setMicVolume(unsigned short mic_vol) { _mic_volume = mic_vol; }
-
- // 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 svr : serveur on which to send request
- * @param port : on which port we want to listen to
- *
- * Return true if we are behind a NAT (without error)
- */
- bool behindNat(const std::string& svr, int port);
-
- /**
- * Init default values for the different fields in the config file.
- * Fills the local _config (Conf::ConfigTree) with the default contents.
- *
- * Called in main.cpp, just before Manager::init().
- */
- void initConfigFile (void);
-
- /**
- * Tell if the setup was already loaded
- */
- bool hasLoadedSetup() { return _setupLoaded; }
-
- /** 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);
-
- /**
- * Map accounts parameters ( authname - hostname ) to an account ID
- */
- AccountID getAccountFromEvent( std::string authname );
-private:
- /**
- * Create .PROGNAME directory in home user and create
- * configuration tree from the settings file if this file exists.
- *
- * @return 0 if creating file failed
- * 1 if config-file exists
- * 2 if file doesn't exist yet.
- */
- int createSettingsPath (void);
-
- /*
- * Initialize audiocodec with config setting
- */
- void initAudioCodec(void);
-
- /*
- * Initialize audiodriver
- */
- void initAudioDriver(void);
-
- /*
- * Initialize zeroconf module and scanning
- */
- void initZeroconf(void);
-
- /*
- * Init the Gui interface (after setting it) inside setGui
- */
- void initGui();
-
- /*
- * Init the volume for speakers/micro from 0 to 100 value
- */
- void initVolume();
-
- /**
- * Configuration
- */
- bool getDirListing(const std::string& sequenceId, const std::string& path, int *nbFile);
- bool getAudioDeviceList(const std::string& sequenceId, int ioDeviceMask);
- Conf::ConfigTree _config;
- bool getCountryTones(const std::string& sequenceId);
- void sendCountryTone(const std::string& sequenceId, int index, const std::string& name);
-
-
-
- /**
- * 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 switchCall(const CallID& id);
-
- /** Current Call ID */
- CallID _currentCallId2;
-
- /** Protected current call access */
- ost::Mutex _currentCallMutex;
-
-
- /*
- * Play one tone
- * @return false if the driver is uninitialize
- */
- bool playATone(Tone::TONEID toneId);
-
- //
- // Multithread variable with extern accessor and change only inside the main thread
- //
- /** Vector of CodecDescriptor */
- CodecDescriptor* _codecBuilder;
-
- //
- // Sound variable
- //
- AudioLayer* _audiodriver;
-
- // Main thread
- DTMF* _dtmfKey;
-
- // map of codec (for configlist request)
- CodecDescriptor _codecDescriptorMap;
-
- /////////////////////
- // Protected by Mutex
- /////////////////////
- ost::Mutex _toneMutex;
- TelephoneTone* _telephoneTone;
- AudioFile _audiofile;
-
- // To handle volume control
- 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;
-
- //
- // Multithread variable (non protected)
- //
- DBusManagerImpl * _dbus;
-
- /** 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;
- /**
- * Add incoming callid to the waiting list
- * @param id CallID to add
- */
- 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
- */
- std::string _path;
- int _exist;
- int _setupLoaded;
-
- // To handle firewall
- int _firewallPort;
- std::string _firewallAddr;
-
- // tell if we have zeroconf is enabled
- int _hasZeroconf;
+ public:
+ ManagerImpl (void);
+ ~ManagerImpl (void);
+
+ /**
+ * Initialisation of thread (sound) and map.
+ * Init a new VoIPLink, audio codec and audio driver
+ */
+ void init (void);
+
+ /**
+ * Terminate all thread (sound, link) and unload AccountMap
+ */
+ void terminate (void);
+
+ /**
+ * Set user interface manager.
+ * @param man The DBUS interface implementation
+ */
+ void setDBusManager (DBusManagerImpl* man) { _dbus = man; }
+
+ /**
+ * Accessor to audiodriver.
+ * it's multi-thread and use mutex internally
+ * @return AudioLayer* The audio layer object
+ */
+ AudioLayer* getAudioDriver(void) const { return _audiodriver; }
+
+ /**
+ * Get a descriptor map of codec available
+ * @return CodecDescriptor The internal codec map
+ */
+ CodecDescriptor& getCodecDescriptorMap(void) {return _codecDescriptorMap;}
+
+ /**
+ * Functions which occur with a user's action
+ * Place a new call
+ * @param accountId The account to make tha call with
+ * @param id The call identifier
+ * @param to The recipient of the call
+ * @return bool true on success
+ * false otherwise
+ */
+ bool outgoingCall(const AccountID& accountId, const CallID& id, const std::string& to);
+
+ /**
+ * Functions which occur with a user's action
+ * Answer the call
+ * @param id The call identifier
+ */
+ bool answerCall(const CallID& id);
+
+ /**
+ * Functions which occur with a user's action
+ * Hangup the call
+ * @param id The call identifier
+ */
+ bool hangupCall(const CallID& id);
+
+ /**
+ * Functions which occur with a user's action
+ * Cancel the call
+ * @param id The call identifier
+ */
+ bool cancelCall(const CallID& id);
+
+ /**
+ * Functions which occur with a user's action
+ * Put the call on hold
+ * @param id The call identifier
+ */
+ bool onHoldCall(const CallID& id);
+
+ /**
+ * Functions which occur with a user's action
+ * Put the call off hold
+ * @param id The call identifier
+ */
+ bool offHoldCall(const CallID& id);
+
+ /**
+ * Functions which occur with a user's action
+ * Transfer the call
+ * @param id The call identifier
+ * @param to The recipient of the transfer
+ */
+ bool transferCall(const CallID& id, const std::string& to);
+
+ /**
+ * Functions which occur with a user's action
+ * Refuse the call
+ * @param id The call identifier
+ */
+ bool refuseCall(const CallID& id);
+
+ /**
+ * Save config to file
+ * @return true on success
+ * false otherwise
+ */
+ bool saveConfig (void);
+
+ /**
+ * Send registration to all enabled accounts
+ * @return false if exosip or the network checking fails
+ */
+ bool initRegisterAccounts();
+
+ /**
+ * @return true if we tried to register once
+ */
+ bool _hasTriedToRegister;
+
+ /**
+ * Undocumented
+ */
+ 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(const CallID& id, char code);
+
+ /**
+ * Play the dtmf-associated sound
+ * @param code The pressed key
+ * @param isTalking In conversation or not. Useful to know whether or not the sound streams are started
+ */
+ bool playDtmf(char code, bool isTalking);
+
+ /**
+ * Play a ringtone
+ * @return bool True on success
+ * false otherwise
+ */
+ bool playTone ();
+
+ /**
+ * Play a special ringtone ( BUSY ) if there's at least one message on the voice mail
+ * @return bool True on success
+ * false otherwise
+ */
+ bool playToneWithMessage ();
+
+ /**
+ * Acts on the audio streams and audio files
+ * @param stopAudio Tells whether or not to stop the streams
+ */
+ void stopTone(bool stopAudio);
+
+ /**
+ * 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 bool True if the call was added correctly
+ */
+ bool incomingCall(Call* call, const AccountID& accountId);
+
+ /**
+ * Notify the user that the recipient of the call has answered and the put the
+ * call in Current state
+ * @param id The call identifier
+ */
+ void peerAnsweredCall(const CallID& id);
+
+ /**
+ * Rings back because the outgoing call is ringing and the put the
+ * call in Ringing state
+ * @param id The call identifier
+ */
+ void peerRingingCall(const CallID& id);
+
+ /**
+ * Put the call in Hungup state, remove the call from the list
+ * @param id The call identifier
+ */
+ void peerHungupCall(const CallID& id);
+
+ /**
+ * Notify the client with an incoming message
+ * @param accountId The account identifier
+ * @param message The content of the message
+ */
+ void incomingMessage(const AccountID& accountId, const std::string& message);
+
+ /**
+ * Notify the user he has voice mails
+ * @param accountId The account identifier
+ * @param nb_msg The number of messages
+ */
+ void startVoiceMessageNotification(const AccountID& accountId, const std::string& nb_msg);
+
+ /**
+ * Notify the user that registration succeeded
+ * @param accountId The account identifier
+ */
+ void registrationSucceed(const AccountID& accountId);
+
+ /**
+ * Notify the user that unregistration succeeded
+ * @param accountId The account identifier
+ */
+ void unregistrationSucceed(const AccountID& accountId);
+
+ /**
+ * Notify the user that registration failed
+ * @param accountId The account identifier
+ */
+ void registrationFailed(const AccountID& accountId);
+
+ /**
+ * Notify the user that registration is trying
+ * @param accountId The account identifier
+ */
+ void registrationTrying(const AccountID& accountId);
+
+ /**
+ * ConfigurationManager - Send registration request
+ * @param accountId The account to register/unregister
+ * @param expire The flag for the type of registration
+ * 0 for unregistration request
+ * 1 for registration request
+ */
+ void sendRegister( const ::DBus::String& accountId , bool expire );
+
+ bool getZeroconf(const std::string& sequenceId);
+ bool attachZeroconfEvents(const std::string& sequenceId, Pattern::Observer& observer);
+ bool detachZeroconfEvents(Pattern::Observer& observer);
+ bool getCallStatus(const std::string& sequenceId);
+
+ /**
+ * Get account list
+ * @return std::vector<std::string> A list of accoundIDs
+ */
+ std::vector< std::string > getAccountList();
+
+ /**
+ * Retrieve details about a given account
+ * @param accountID The account identifier
+ * @return std::map< std::string, std::string > The account details
+ */
+ std::map< std::string, std::string > getAccountDetails(const AccountID& accountID);
+
+ /**
+ * Save the details of an existing account, given the account ID
+ * This will load the configuration map with the given data.
+ * It will also register/unregister links where the 'Enabled' switched.
+ * @param accountID The account identifier
+ * @param details The account parameters
+ */
+ void setAccountDetails( const ::DBus::String& accountID,
+ const std::map< ::DBus::String, ::DBus::String >& details );
+
+ /**
+ * Add a new account, and give it a new account ID automatically
+ * @param details The new account parameters
+ */
+ void addAccount(const std::map< ::DBus::String, ::DBus::String >& details);
+
+ /**
+ * Delete an existing account, unregister VoIPLink associated, and
+ * purge from configuration.
+ * @param accountID The account unique ID
+ */
+ void removeAccount(const AccountID& accountID);
+
+ /**
+ * Get the list of codecs we supports, not ordered
+ * @return The list of the codecs
+ */
+ std::vector< ::DBus::String > getCodecList( void );
+
+ /**
+ * Get the info about one codec
+ * Name / Clock rate / bitrate / bandwidth
+ * @param payload The payload of the codec
+ * @return std::vector<::DBus::string> The information
+ */
+ std::vector< ::DBus::String > getCodecDetails( const ::DBus::Int32& payload);
+
+ /**
+ * Get a list of supported input audio plugin
+ * @return std::vector<std::string> List of names
+ */
+ std::vector< std::string> getInputAudioPluginList(void);
+
+ /**
+ * Get a list of supported output audio plugin
+ * @return std::vector<std::string> List of names
+ */
+ std::vector< std::string> getOutputAudioPluginList(void);
+
+ /**
+ * Set input audio plugin
+ * @param audioPlugin The audio plugin
+ */
+ void setInputAudioPlugin(const std::string& audioPlugin);
+
+ /**
+ * Set output audio plugin
+ * @param audioPlugin The audio plugin
+ */
+ void setOutputAudioPlugin(const std::string& audioPlugin);
+
+ /**
+ * Get list of supported audio output device
+ * @return std::vector<std::string> A list of the audio devices supporting playback
+ */
+ std::vector<std::string> getAudioOutputDeviceList(void);
+
+ /**
+ * Set audio output device
+ * @param index The index of the soundcard
+ */
+ void setAudioOutputDevice(const int index);
+
+ /**
+ * Get list of supported audio input device
+ * @return std::vector<std::string> A list of the audio devices supporting capture
+ */
+ std::vector<std::string> getAudioInputDeviceList(void);
+
+ /**
+ * Set audio input device
+ * @param index The index of the soundcard
+ */
+ void setAudioInputDevice(const int index);
+
+ /**
+ * Get string array representing integer indexes of output and input device
+ * @return std::vector<std::string> A list of the current audio devices
+ */
+ std::vector<std::string> getCurrentAudioDevicesIndex();
+
+ /**
+ * Get index of an audio device
+ * @param name The string description of an audio device
+ * @return int His index
+ */
+ int getAudioDeviceIndex( const std::string name );
+
+ /*
+ * Get current alsa plugin
+ * @return std::string The Alsa plugin
+ */
+ std::string getCurrentAudioOutputPlugin( void );
+
+ /**
+ * Convert a list of payload in a special format, readable by the server.
+ * Required format: payloads separated with one slash.
+ * @return std::string The serializabled string
+ */
+ std::string serialize(std::vector<std::string> v);
+
+ /**
+ * Tells if IAX2 support is enabled
+ * @return int 1 if IAX2 is enabled
+ * 0 otherwise
+ */
+ int isIax2Enabled( void );
+
+ /**
+ * Ringtone option.
+ * If ringtone is enabled, ringtone on incoming call use custom choice. If not, only standart tone.
+ * @return int 1 if enabled
+ * 0 otherwise
+ */
+ int isRingtoneEnabled( void );
+
+ /**
+ * Set the ringtone option
+ * Inverse current value
+ */
+ void ringtoneEnabled( void );
+
+ /**
+ * Get the ringtone
+ * @return gchar* The file name selected as a ringtone
+ */
+ std::string getRingtoneChoice( void );
+
+ /**
+ * Set a ringtone
+ * @param tone The file name of the ringtone
+ */
+ void setRingtoneChoice( const std::string& );
+
+ /**
+ * Tells if the user wants to display the dialpad or not
+ * @return int 1 if dialpad has to be displayed
+ * 0 otherwise
+ */
+ int getDialpad( void );
+
+ /**
+ * Set the dialpad visible or not
+ */
+ void setDialpad( void );
+
+ /**
+ * Configure the start-up option
+ * @return int 1 if SFLphone should start in the system tray
+ * 0 otherwise
+ */
+ int isStartHidden( void );
+
+ /**
+ * Configure the start-up option
+ * At startup, SFLphone can be displayed or start hidden in the system tray
+ */
+ void startHidden( void );
+
+ /**
+ * Configure the popup behaviour
+ * @return int 1 if it should popup on incoming calls
+ * 0 if it should never popups
+ */
+ int popupMode( void );
+
+ /**
+ * Configure the popup behaviour
+ * When SFLphone is in the system tray, you can configure when it popups
+ * Never or only on incoming calls
+ */
+ void switchPopupMode( void );
+
+ /**
+ * Retrieve the formatted list of codecs payload in the user config file and
+ * load in the active list of codecs
+ * @return std::vector<std::string> The vector containing the active codecs
+ */
+ std::vector<std::string> retrieveActiveCodecs( void );
+
+ /**
+ * Get the list of the active codecs
+ * @return std::vector< ::DBus::String > The list of active codecs
+ */
+ std::vector< ::DBus::String > getActiveCodecList( void );
+
+ /**
+ * Set the list of the active codecs
+ * @param list The new list of active codecs
+ */
+ void setActiveCodecList( const std::vector< ::DBus::String >& list);
+
+ /*
+ * Notify the client that an error occured
+ * @param errCode The error code. Could be: ALSA_CAPTURE_ERROR
+ * ALSA_PLAYBACK_ERROR
+ */
+ void notifyErrClient( const ::DBus::Int32& errCode );
+
+ /**
+ * Retrieve in the configuration tree the value of a parameter in a specific section
+ * @param section The section to look in
+ * @param name The name of the parameter you want to get
+ * @param arg Undocumented
+ * @return bool true on success
+ * false otherwise
+ */
+ bool getConfig(const std::string& section, const std::string& name, TokenList& arg);
+
+ /**
+ * Change a specific value in the configuration tree.
+ * This value will then be saved in the user config file sflphonedrc
+ * @param section The section name
+ * @param name The parameter name
+ * @param value The new string value
+ * @return bool true on success
+ * false otherwise
+ */
+ bool setConfig(const std::string& section, const std::string& name, const std::string& value);
+
+ /**
+ * Change a specific value in the configuration tree.
+ * This value will then be saved in the user config file sflphonedrc
+ * @param section The section name
+ * @param name The parameter name
+ * @param value The new int value
+ * @return bool true on success
+ * false otherwise
+ */
+ bool setConfig(const std::string& section, const std::string& name, int value);
+
+ /**
+ * Get a int from the configuration tree
+ * Throw an Conf::ConfigTreeItemException if not found
+ * @param section The section name to look in
+ * @param name The parameter name
+ * @return int The int value
+ */
+ int getConfigInt(const std::string& section, const std::string& name);
+
+ /**
+ * Get a string from the configuration tree
+ * Throw an Conf::ConfigTreeItemException if not found
+ * @param section The section name to look in
+ * @param name The parameter name
+ * @return sdt::string The string value
+ */
+ std::string getConfigString(const std::string& section, const std::string& name);
+
+ /**
+ * Retrieve the soundcards index in the user config file and try to open audio devices
+ * with a specific alsa plugin.
+ * Set the audio layer sample rate
+ */
+ void selectAudioDriver(void);
+
+ /**
+ * 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 ();
+
+ /**
+ * Handle played music when a congestion occurs
+ */
+ void congestion ();
+
+ /**
+ * Handle played sound when a call can not be conpleted because of a busy recipient
+ */
+ void callBusy(const CallID& id);
+
+ /**
+ * Handle played sound when a failure occurs
+ */
+ void callFailure(const CallID& id);
+
+ /**
+ * Retrieve the current telephone tone
+ * @return AudioLoop* The audio tone or 0 if no tone (init before calling this function)
+ */
+ AudioLoop* getTelephoneTone();
+
+ /**
+ * Retrieve the current telephone file
+ * @return AudioLoop* The audio file or 0 if the wav is stopped
+ */
+ AudioLoop* getTelephoneFile();
+
+ /**
+ * @return true is there is one or many incoming call waiting
+ * new call, not anwsered or refused
+ */
+ bool incomingCallWaiting(void);
+
+ /**
+ * Notification of incoming call when you are already busy
+ */
+ void notificationIncomingCall(void);
+
+ /*
+ * Inline functions to manage speaker volume control
+ * Read by main thread and AudioLayer thread
+ * Write by main thread only
+ * @return unsigned short The volume value
+ */
+ unsigned short getSpkrVolume(void) { return _spkr_volume; }
+
+ /*
+ * Inline functions to manage speaker volume control
+ * Read by main thread and AudioLayer thread
+ * Write by main thread only
+ * @param spkr_vol The volume value
+ */
+ void setSpkrVolume(unsigned short spkr_vol) { _spkr_volume = spkr_vol; }
+
+ /*
+ * Inline functions to manage mic volume control
+ * Read by main thread and AudioLayer thread
+ * Write by main thread only
+ * @return unsigned short The volume value
+ */
+ unsigned short getMicVolume(void) { return _mic_volume; }
+
+ /*
+ * Inline functions to manage mic volume control
+ * Read by main thread and AudioLayer thread
+ * Write by main thread only
+ * @param mic_vol The volume value
+ */
+ void setMicVolume(unsigned short mic_vol) { _mic_volume = mic_vol; }
+
+ // 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 functions to manage firewall settings
+ * @return int The firewall port
+ */
+ inline int getFirewallPort(void) { return _firewallPort; }
+
+ /*
+ * Inline functions to manage firewall settings
+ * @param port The firewall port
+ */
+ inline void setFirewallPort(int port) { _firewallPort = port; }
+
+ /*
+ * Inline functions to manage firewall settings
+ * @return std::string The firewall address
+ */
+ 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 svr Server on which to send request
+ * @param port On which port we want to listen to
+ * @return true if we are behind a NAT (without error)
+ */
+ bool behindNat(const std::string& svr, int port);
+
+ /**
+ * Init default values for the different fields in the config file.
+ * Fills the local _config (Conf::ConfigTree) with the default contents.
+ * Called in main.cpp, just before Manager::init().
+ */
+ void initConfigFile (void);
+
+ /**
+ * Tell if the setup was already loaded
+ * @return bool True if yes
+ * false otherwise
+ */
+ bool hasLoadedSetup() { return _setupLoaded; }
+
+ /**
+ * Return a new random callid that is not present in the list
+ * @return CallID A brand new callid
+ */
+ CallID getNewCallID();
+
+ /**
+ * Get the current call id
+ * @return CallID The call id or ""
+ */
+ const CallID& getCurrentCallId();
+
+ /**
+ * Check if a call is the current one
+ * @param callId the new callid
+ * @return bool True if the id is the current call
+ */
+ bool isCurrentCall(const CallID& callId);
+
+ private:
+ /**
+ * Create .PROGNAME directory in home user and create
+ * configuration tree from the settings file if this file exists.
+ *
+ * @return 0 if creating file failed
+ * 1 if config-file exists
+ * 2 if file doesn't exist yet.
+ */
+ int createSettingsPath (void);
+
+ /*
+ * Initialize audiocodec with config setting
+ */
+ void initAudioCodec(void);
+
+ /*
+ * Initialize audiodriver
+ */
+ void initAudioDriver(void);
+
+ /*
+ * Initialize zeroconf module and scanning
+ */
+ void initZeroconf(void);
+
+ /*
+ * Init the volume for speakers/micro from 0 to 100 value
+ */
+ void initVolume();
+
+ /**
+ * Tell if there is a current call processed
+ * @return bool True if there is a current call
+ */
+ bool hasCurrentCall();
+
+ /**
+ * Switch of current call id
+ * @param id The new callid
+ */
+ void switchCall(const CallID& id);
+
+ /*
+ * Play one tone
+ * @return false if the driver is uninitialize
+ */
+ bool playATone(Tone::TONEID toneId);
+
+ /** The configuration tree. It contains accounts parameters, general user settings ,audio settings, ... */
+ Conf::ConfigTree _config;
+
+ /** Current Call ID */
+ CallID _currentCallId2;
+
+ /** Protected current call access */
+ ost::Mutex _currentCallMutex;
+
+ /** Vector of CodecDescriptor */
+ CodecDescriptor* _codecBuilder;
+
+ /** Audio layer */
+ AudioLayer* _audiodriver;
+
+ // Main thread
+
+ DTMF* _dtmfKey;
+
+ // map of codec (for configlist request)
+ CodecDescriptor _codecDescriptorMap;
+
+ /////////////////////
+ // Protected by Mutex
+ /////////////////////
+ ost::Mutex _toneMutex;
+ TelephoneTone* _telephoneTone;
+ AudioFile _audiofile;
+
+ // To handle volume control
+ short _spkr_volume;
+ short _mic_volume;
+ // End of sound variable
+
+
+ // Multithread variable (protected by _mutex)
+ //
+ /** Mutex to protect access to code section */
+ ost::Mutex _mutex;
+
+ // Multithread variable (non protected)
+ DBusManagerImpl * _dbus;
+
+ /** 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;
+
+ /**
+ * Add incoming callid to the waiting list
+ * @param id CallID to add
+ */
+ 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 bool True if the call is waiting
+ */
+ bool isWaitingCall(const CallID& id);
+
+ /**
+ * Path of the ConfigFile
+ */
+ std::string _path;
+ int _exist;
+ int _setupLoaded;
+
+ // To handle firewall
+ int _firewallPort;
+ std::string _firewallAddr;
+
+ // tell if we have zeroconf is enabled
+ int _hasZeroconf;
#ifdef USE_ZEROCONF
- // DNSService contain every zeroconf services
- // configuration detected on the network
- DNSService *_DNSService;
+ // DNSService contain every zeroconf services
+ // configuration detected on the network
+ DNSService *_DNSService;
#endif
- /** Map to associate a CallID to the good account */
- CallAccountMap _callAccountMap;
- /** Mutex to lock the call account map (main thread + voiplink thread) */
- ost::Mutex _callAccountMapMutex;
-
- /** Associate a new CallID to a AccountID
- * Protected by mutex
- * @param callID the new CallID not in the list yet
- * @param accountID the known accountID present in accountMap
- * @return true if the new association is create
- */
- 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);
-
- /** Remove a CallID/AccountID association
- * Protected by mutex
- * @param callID the CallID to remove
- * @return true if association is removed
- */
- bool removeCallAccount(const CallID& callID);
-
- /** Contains a list of account (sip, aix, etc) and their respective voiplink/calls */
- AccountMap _accountMap;
-
- /**
- * Load the account from configuration
- * @return number of account
- */
- short loadAccountMap();
-
- /**
- * Unload the account (delete them)
- */
- void unloadAccountMap();
-
- /**
- * Tell if an account exists
- * @param accountID account ID check
- */
- bool accountExists(const AccountID& accountID);
-
- /**
- * Get an account pointer
- * @param accountID account ID to get
- * @param the account pointer or 0
- */
- Account* getAccount(const AccountID& accountID);
-
- /**
- * Get the voip link from the account pointer
- * @param accountID account ID to get
- * @param the voip link from the account pointer or 0
- */
- VoIPLink* getAccountLink(const AccountID& accountID);
-
-
-
- #ifdef TEST
- bool testCallAccountMap();
- bool testAccountMap();
- #endif
+ /** Map to associate a CallID to the good account */
+ CallAccountMap _callAccountMap;
+
+ /** Mutex to lock the call account map (main thread + voiplink thread) */
+ ost::Mutex _callAccountMapMutex;
+
+ /** Associate a new CallID to a AccountID
+ * Protected by mutex
+ * @param callID the new CallID not in the list yet
+ * @param accountID the known accountID present in accountMap
+ * @return bool True if the new association is create
+ */
+ 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 AccountID The accountID associated or "" if the callID is not found
+ */
+ AccountID getAccountFromCall(const CallID& callID);
+
+ /** Remove a CallID/AccountID association
+ * Protected by mutex
+ * @param callID the CallID to remove
+ * @return bool True if association is removed
+ */
+ bool removeCallAccount(const CallID& callID);
+
+ /** Contains a list of account (sip, aix, etc) and their respective voiplink/calls */
+ AccountMap _accountMap;
+
+ /**
+ * Load the account from configuration
+ * @return short Number of account
+ */
+ short loadAccountMap();
+
+ /**
+ * Unload the account (delete them)
+ */
+ void unloadAccountMap();
+
+ /**
+ * Tell if an account exists
+ * @param accountID account ID check
+ * @return bool True if the account exists
+ * false otherwise
+ */
+ bool accountExists(const AccountID& accountID);
+
+ /**
+ * Get an account pointer
+ * @param accountID account ID to get
+ * @return Account* The account pointer or 0
+ */
+ Account* getAccount(const AccountID& accountID);
+
+ /**
+ * Get the voip link from the account pointer
+ * @param accountID Account ID to get
+ * @return VoIPLink* The voip link from the account pointer or 0
+ */
+ VoIPLink* getAccountLink(const AccountID& accountID);
+
+#ifdef TEST
+ bool testCallAccountMap();
+ bool testAccountMap();
+#endif
};
diff --git a/src/sipaccount.h b/src/sipaccount.h
index 23c2fc3f529f09821820f238c9faa3e626d261dd..e542032b02cd4ede155d3548d120aaf4e9c5cb6d 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -24,19 +24,37 @@
/**
- * A SIP Account specify SIP specific functions and object (SIPCall/SIPVoIPLink)
- * @author Yan Morin <yan.morin@gmail.com>
+ * @file sipaccount.h
+ * @brief A SIP Account specify SIP specific functions and object (SIPCall/SIPVoIPLink)
*/
+
class SIPAccount : public Account
{
public:
+ /**
+ * Constructor
+ * @param accountID The account identifier
+ */
SIPAccount(const AccountID& accountID);
+ /**
+ * Virtual destructor
+ */
virtual ~SIPAccount();
- /** Actually unuseful, since config loading is done in init() */
+ /**
+ * Actually unuseful, since config loading is done in init()
+ */
void loadConfig();
+
+ /**
+ * Initialize the SIP voip link with the account parameters and send registration
+ */
void registerVoIPLink();
+
+ /**
+ * Send unregistration and clean all related stuff ( calls , thread )
+ */
void unregisterVoIPLink();
private:
diff --git a/src/sipcall.h b/src/sipcall.h
index 3883380e3f69cc60e93d0cb0f80a263333893bcc..a74885500707958e326a034690f66c20d119a52c 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -28,112 +28,147 @@
class AudioCodec;
/**
- * SIPCall are SIP implementation of a normal Call
- * @author Yan Morin <yan.morin@gmail.com>
+ * @file sipcall.h
+ * @brief SIPCall are SIP implementation of a normal Call
*/
class SIPCall : public Call
{
-public:
+ public:
+
+ /**
+ * Constructor
+ * @param id The call identifier
+ * @param type The type of the call. Could be Incoming
+ * Outgoing
+ */
SIPCall(const CallID& id, Call::CallType type);
+ /**
+ * Destructor
+ */
~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; }
-
- /**
- * Setup incoming call, and verify for errors, before ringing the user.
- * @param event eXosip Event
- */
- 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
- */
- bool SIPCallReinvite(eXosip_event_t *event);
-
- /**
- * 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
- */
- bool SIPCallAnsweredWithoutHold(eXosip_event_t *event);
-
- //TODO: humm?
- int sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg);
-
-
-private:
-
- // TODO: hum???
- int sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med);
- /**
- * 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;
+ /**
+ * Call Identifier
+ * @return int SIP call id : protected by eXosip lock
+ */
+ int getCid() { return _cid; }
+
+ /**
+ * Call Identifier
+ * @param cid SIP call id : protected by eXosip lock
+ */
+ void setCid(int cid) { _cid = cid ; }
+
+ /**
+ * Domain identifier
+ * @return int SIP domain id : protected by eXosip lock
+ */
+ int getDid() { return _did; }
+
+ /**
+ * Domain identifier
+ * @param did SIP domain id : protected by eXosip lock
+ */
+ void setDid(int did) { _did = did; }
+
+ /**
+ * Transaction identifier
+ * @return int SIP transaction id : protected by eXosip lock
+ */
+ int getTid() { return _tid; }
+
+ /**
+ * Transaction identifier
+ * @param tid SIP transaction id : protected by eXosip lock
+ */
+ void setTid(int tid) { _tid = tid; }
+
+ /**
+ * Setup incoming call, and verify for errors, before ringing the user.
+ * @param event eXosip Event
+ * @return bool True on success
+ * false otherwise
+ */
+ 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 bool True if ok
+ */
+ bool SIPCallReinvite(eXosip_event_t *event);
+
+ /**
+ * Peer answered to a call (on hold or not)
+ * @param event eXosip Event
+ * @return bool True if ok
+ */
+ bool SIPCallAnswered(eXosip_event_t *event);
+
+ /**
+ * We retreive final SDP info if they changed
+ * @param event eXosip Event
+ * @return bool True if ok (change / no change) or false on error
+ */
+ bool SIPCallAnsweredWithoutHold(eXosip_event_t *event);
+
+ //TODO: humm?
+ int sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg);
+
+
+ private:
+
+ // TODO: hum???
+ int sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med);
+
+ /**
+ * Set peer name and number with event->request->from
+ * @param event eXosip event
+ * @return bool False if 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 sdp_message_t* A 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 sdp_media_t* A 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 bool 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 bool 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;
};
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 741fd963e7a621ea6e15b1e6066ac2bc179f1db4..47320dce08b5c1f05ace808242036f79b4c9d7d5 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -215,12 +215,6 @@ SIPVoIPLink::loadSIPLocalIP()
return returnValue;
}
-void
-SIPVoIPLink::parseRequestUri( osip_uri_t* req )
-{
- // _debug("%d\n",req->url_header);
-}
-
void
SIPVoIPLink::getEvent()
{
@@ -313,7 +307,6 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_CALL_TIMEOUT: /** 17 < announce that call has failed */
_debugMid(" !EXOSIP_CALL_TIMEOUT\n");
- Manager::instance().displayError(" !EXOSIP Call Error not implemented yet");
break;
/* Request related events within calls (except INVITE) */
@@ -339,7 +332,6 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_CALL_MESSAGE_GLOBALFAILURE: /** 24 < announce a failure. */
_debugMid(" !EXOSIP_CALL_MESSAGE_GLOBALFAILURE\n");
- Manager::instance().displayError(" !EXOSIP Call Message not implemented yet");
break;
case EXOSIP_CALL_CLOSED: /** 25 < a BYE was received for this call */
@@ -367,23 +359,18 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_MESSAGE_REDIRECTED: /** 30 < announce a failure. */
_debugMid(" !EXOSIP_MESSAGE_REDIRECTED\n");
- Manager::instance().displayError(" !EXOSIP Message not implemented yet");
break;
case EXOSIP_MESSAGE_REQUESTFAILURE: /** 31 < announce a failure. */
_debugMid(" !EXOSIP_MESSAGE_REQUESTFAILURE\n");
- if (event->response !=0 && event->response->status_code == SIP_METHOD_NOT_ALLOWED) {
+ 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. */
_debugMid(" !EXOSIP_MESSAGE_SERVERFAILURE\n");
break;
case EXOSIP_MESSAGE_GLOBALFAILURE: /** 33 < announce a failure. */
_debugMid(" !EXOSIP_MESSAGE_GLOBALFAILURE\n");
- Manager::instance().displayError(" !EXOSIP Message not implemented yet");
break;
/* Presence and Instant Messaging */
@@ -392,7 +379,6 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_SUBSCRIPTION_CLOSED: /** 35 < announce end of subscription. */
_debugMid(" !EXOSIP_SUBSCRIPTION_CLOSED\n");
- Manager::instance().displayError(" !EXOSIP Subscription not implemented yet");
break;
case EXOSIP_SUBSCRIPTION_NOANSWER: /** 37 < announce no answer */
@@ -400,7 +386,6 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_SUBSCRIPTION_PROCEEDING: /** 38 < announce a 1xx */
_debugMid(" !EXOSIP_SUBSCRIPTION_PROCEEDING\n");
- Manager::instance().displayError(" !EXOSIP Subscription response not implemented yet");
break;
case EXOSIP_SUBSCRIPTION_ANSWERED: /** 39 < announce a 200ok */
_debugMid(" !EXOSIP_SUBSCRIPTION_ANSWERED\n");
@@ -435,7 +420,6 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_SUBSCRIPTION_RELEASED: /** 45 < call context is cleared. */
_debugMid(" !EXOSIP_SUBSCRIPTION_RELEASED\n");
- Manager::instance().displayError(" !EXOSIP Subscription response not implemented yet.");
break;
case EXOSIP_IN_SUBSCRIPTION_NEW: /** 46 < announce new incoming SUBSCRIBE.*/
@@ -443,7 +427,6 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_IN_SUBSCRIPTION_RELEASED: /** 47 < announce end of subscription. */
_debugMid(" !EXOSIP_IN_SUBSCRIPTION_RELEASED\n");
- Manager::instance().displayError(" !EXOSIP Subscription not implemented yet");
break;
case EXOSIP_EVENT_COUNT: /** 48 < MAX number of events */
@@ -461,19 +444,15 @@ SIPVoIPLink::sendRegister()
{
if (_eXosipRegID != EXOSIP_ERROR_STD) {
- _debug("nlvnslvnlsnvaljsdnvjlasnvlsfvbnnns sjvlsvn\n");
- Manager::instance().displayError("! SIP Error: Registration already sent. Try to unregister");
return false;
}
std::string hostname = getHostName();
if (hostname.empty()) {
- Manager::instance().displayConfigError("Fill host part field");
return false;
}
if (_userpart.empty()) {
- Manager::instance().displayConfigError("Fill user part field");
return false;
}
@@ -531,12 +510,6 @@ SIPVoIPLink::SIPFromHeader(const std::string& userpart, const std::string& hostp
return ("\"" + getFullName() + "\"" + " <sip:" + userpart + "@" + hostpart + ">");
}
-std::string
-SIPVoIPLink::SIPFromHeaderAlternate(const std::string& userpart, const std::string& hostpart)
-{
- return ("<sip:" + userpart + "@" + hostpart + ">");
-}
-
bool
SIPVoIPLink::sendSIPAuthentification()
{
@@ -546,12 +519,10 @@ SIPVoIPLink::sendSIPAuthentification()
}
if (login.empty()) {
/** @todo Ajouter ici un call à setRegistrationState(Error, "Fill balh") ? */
- Manager::instance().displayConfigError("Fill authentification name");
return false;
}
if (_password.empty()) {
/** @todo Même chose ici ? */
- Manager::instance().displayConfigError("Fill password field");
return false;
}
eXosip_lock();
@@ -942,11 +913,9 @@ SIPVoIPLink::sendMessage(const std::string& to, const std::string& body)
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;
}
@@ -1085,11 +1054,9 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
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;
}
@@ -1177,9 +1144,6 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
return true;
}
-/**
- * Get the Sip FROM url (add sip:, add @host, etc...)
- */
std::string
SIPVoIPLink::getSipFrom() {
@@ -1191,9 +1155,6 @@ SIPVoIPLink::getSipFrom() {
return SIPFromHeader(_userpart, host);
}
-/**
- * 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
@@ -1209,10 +1170,6 @@ SIPVoIPLink::getSipTo(const std::string& to_url) {
return SIPToHeader(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 = _proxy;
@@ -1409,7 +1366,6 @@ SIPVoIPLink::SIPCallRequestFailure(eXosip_event_t *event)
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);
}
@@ -1430,13 +1386,11 @@ SIPVoIPLink::SIPCallRequestFailure(eXosip_event_t *event)
case SIP_NOT_ACCEPTABLE_HERE: // 488 */
// Display error on the screen phone
{
- _debug("--------------------------------------------------------------error message: %s\n", event->response->reason_phrase);
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);
}
@@ -1610,11 +1564,7 @@ SIPVoIPLink::SIPMessageNew(eXosip_event_t *event)
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");
@@ -1697,10 +1647,6 @@ SIPVoIPLink::getSIPCall(const CallID& id)
return NULL;
}
-/**
- * Handle an INFO with application/dtmf-relay content-type
- * @param event eXosip Event
- */
bool
SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 25490c9d5856a67afac6a7dc6c7a34c877bdcc80..a62aa7b9297c7ec7a35d9a472ddba88b8d2a3a60 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -1,7 +1,8 @@
/*
- * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -30,263 +31,413 @@ class EventThread;
class SIPCall;
/**
- * Specific VoIPLink for SIP (SIP core for incoming and outgoing events)
+ * @file sipvoiplink.h
+ * @brief Specific VoIPLink for SIP (SIP core for incoming and outgoing events)
*/
class SIPVoIPLink : public VoIPLink
{
-public:
- SIPVoIPLink(const AccountID& accountID);
-
- ~SIPVoIPLink();
-
- int eXosip_running;
-
- /** try to initiate the eXosip engine/thread and set config */
- bool init(void);
- void terminate(void);
- bool checkNetwork(void);
- void getEvent(void);
-
- bool sendRegister(void);
- bool sendUnregister(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);
- bool isContactPresenceSupported();
- void subscribePresenceForContact(Contact* contact);
- void publishPresenceStatus(std::string status);
-
- // TODO Not used yet
- void sendMessageToContact(const CallID& id, const std::string& message);
-
- // SIP Specific
-
- /** If set to true, we check for a firewall
- * @param use true if we use STUN
- */
- void setUseStun(bool use) { _useStun = use; }
-
- /** The name of the STUN server
- * @param server Server FQDN/IP
- */
- void setStunServer(const std::string& server) { _stunServer = server; }
-
- /** Set the SIP proxy
- * @param proxy Proxy FQDN/IP
- */
- 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:
- void parseRequestUri( osip_uri_t* );
- /** 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);
- std::string SIPFromHeaderAlternate(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);
-
- /**
- * Use a exisiting call to restart the audio
- * @param event eXosip Event
- */
- void SIPCallReinvite(eXosip_event_t *event);
-
- /**
- * 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);
-
- void SIPRegistrationFailure( eXosip_event_t *event );
- /**
- * Handling ack (restart audio if reinvite)
- * @param event eXosip Event
- */
- void SIPCallAck(eXosip_event_t *event);
-
- /**
- * Handling message inside a call (like dtmf)
- * @param event eXosip Event
- */
- void SIPCallMessageNew(eXosip_event_t *event);
- bool handleDtmfRelay(eXosip_event_t *event);
-
- /**
- * Peer close the connection
- * @param event eXosip Event
- */
- void SIPCallClosed(eXosip_event_t *event);
-
- /**
- * The call pointer was released
- * If the call was not cleared before, report an error
- * @param event eXosip 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 _initDone;
-
- /** 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;
-
- /** What is the stun server? */
- std::string _stunServer;
-
- /** Local Extern Address is the IP address seen by peers for SIP listener */
- std::string _localExternAddress;
-
- /** Local Extern Port is the port seen by peers for SIP listener */
- unsigned int _localExternPort;
-
- /** SIP Proxy URL */
- std::string _proxy;
-
- /** SIP UserPart */
- std::string _userpart;
-
- /** SIP Authenfication name */
- std::string _authname;
-
- /** SIP Authenfication password */
- std::string _password;
-
- /** Starting sound */
- AudioRtp _audiortp;
+ public:
+
+ /**
+ * Constructor
+ * @param accountID The account identifier
+ */
+ SIPVoIPLink(const AccountID& accountID);
+
+ /**
+ * Destructor
+ */
+ ~SIPVoIPLink();
+
+ int eXosip_running;
+
+ /**
+ * Try to initiate the eXosip engine/thread and set config
+ * @return bool True if OK
+ */
+ bool init(void);
+
+ /**
+ * Delete link-related stuuf like calls
+ */
+ void terminate(void);
+
+ /**
+ * Check if a local IP can be found
+ * @return bool True if network is reachable
+ */
+ bool checkNetwork(void);
+
+ /**
+ * Event listener. Each event send by the call manager is received and handled from here
+ */
+ void getEvent(void);
+
+ /**
+ * Build and send SIP registration request
+ * @return bool True on success
+ * false otherwise
+ */
+ bool sendRegister(void);
+
+ /**
+ * Build and send SIP unregistration request
+ * @return bool True on success
+ * false otherwise
+ */
+ bool sendUnregister(void);
+
+ /**
+ * Place a new call
+ * @param id The call identifier
+ * @param toUrl The Sip address of the recipient of the call
+ * @return Call* The current call
+ */
+ Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
+
+ /**
+ * Answer the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool answer(const CallID& id);
+
+ /**
+ * Hang up the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool hangup(const CallID& id);
+
+ /**
+ * Cancel the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool cancel(const CallID& id);
+
+ /**
+ * Put the call on hold
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool onhold(const CallID& id);
+
+ /**
+ * Put the call off hold
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool offhold(const CallID& id);
+
+ /**
+ * Transfer the call
+ * @param id The call identifier
+ * @param to The recipient of the transfer
+ * @return bool True on success
+ */
+ bool transfer(const CallID& id, const std::string& to);
+
+ /**
+ * Refuse the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ bool refuse (const CallID& id);
+
+ /**
+ * Send DTMF
+ * @param id The call identifier
+ * @param code The char code
+ * @return bool True on success
+ */
+ bool carryingDTMFdigits(const CallID& id, char code);
+
+ bool sendMessage(const std::string& to, const std::string& body);
+
+ bool isContactPresenceSupported();
+
+ void subscribePresenceForContact(Contact* contact);
+
+ void publishPresenceStatus(std::string status);
+
+ // TODO Not used yet
+ void sendMessageToContact(const CallID& id, const std::string& message);
+
+ /**
+ * If set to true, we check for a firewall
+ * @param use true if we use STUN
+ */
+ void setUseStun(bool use) { _useStun = use; }
+
+ /**
+ * The name of the STUN server
+ * @param server Server FQDN/IP
+ */
+ void setStunServer(const std::string& server) { _stunServer = server; }
+
+ /**
+ * Set the SIP proxy
+ * @param proxy Proxy FQDN/IP
+ */
+ void setProxy(const std::string& proxy) { _proxy = proxy; }
+
+ /**
+ * Set the user part
+ * @param userpart User part
+ */
+ void setUserPart(const std::string& userpart) { _userpart = userpart; }
+
+ /**
+ * Set the authentification name
+ * @param authname The authentification name
+ */
+ void setAuthName(const std::string& authname) { _authname = authname; }
+
+ /**
+ * Set the password
+ * @param password Password
+ */
+ void setPassword(const std::string& password) { _password = password; }
+
+ private:
+
+ /**
+ * 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 bool false if not found
+ */
+ bool loadSIPLocalIP();
+
+ /**
+ * send SIP authentification
+ * @return bool true if sending succeed
+ */
+ bool sendSIPAuthentification();
+
+ /**
+ * Get a SIP From header ("fullname" <sip:userpart@hostpart>)
+ * @param userpart User part
+ * @param hostpart Host name
+ * @return std::string 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
+ * @param to The header of the recipient
+ * @return std::string Result as a string
+ */
+ std::string SIPToHeader(const std::string& to);
+
+ /**
+ * Check if an url is sip-valid
+ * @param url The url to check
+ * @return bool True if osip tell that is valid
+ */
+ bool SIPCheckUrl(const std::string& url);
+
+
+ /**
+ * Send an outgoing call invite
+ * @param call The current call
+ * @return bool True if all is correct
+ */
+ bool SIPOutgoingInvite(SIPCall* call);
+
+ /**
+ * Start a SIP Call
+ * @param call The current call
+ * @param subject Undocumented
+ * @return true if all is correct
+ */
+ bool SIPStartCall(SIPCall* call, const std::string& subject);
+
+ /**
+ * Get the Sip FROM url (add sip:, add @host, etc...)
+ * @return std::string The From url
+ */
+ std::string getSipFrom();
+
+ /**
+ * Get the sip proxy (add sip: if there is one)
+ * @return std::string Empty string or <sip:proxy;lr> url
+ */
+ std::string getSipRoute();
+
+ /**
+ * Get the Sip TO url (add sip:, add @host, etc...)
+ * @param to_url The To url
+ * @return std::string The SIP to address
+ */
+ std::string getSipTo(const std::string& to_url);
+
+ /**
+ * Set audio (SDP) configuration for a call
+ * localport, localip, localexternalport
+ * @param call a SIPCall valid pointer
+ * @return bool True
+ */
+ bool setCallAudioLocal(SIPCall* call);
+
+ /**
+ * Create a new call and send a incoming call notification to the user
+ * @param event eXosip Event
+ */
+ void SIPCallInvite(eXosip_event_t *event);
+
+ /**
+ * Use a exisiting call to restart the audio
+ * @param event eXosip Event
+ */
+ void SIPCallReinvite(eXosip_event_t *event);
+
+ /**
+ * 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);
+
+ /**
+ * Handle registration failure cases ( SIP_FORBIDDEN , SIP_UNAUTHORIZED )
+ * @param event eXosip event
+ */
+ void SIPRegistrationFailure( eXosip_event_t *event );
+
+ /**
+ * Handling ack (restart audio if reinvite)
+ * @param event eXosip Event
+ */
+ void SIPCallAck(eXosip_event_t *event);
+
+ /**
+ * Handling message inside a call (like dtmf)
+ * @param event eXosip Event
+ */
+ void SIPCallMessageNew(eXosip_event_t *event);
+
+ /**
+ * Handle an INFO with application/dtmf-relay content-type
+ * @param event eXosip Event
+ */
+ bool handleDtmfRelay(eXosip_event_t *event);
+
+ /**
+ * Peer close the connection
+ * @param event eXosip Event
+ */
+ void SIPCallClosed(eXosip_event_t *event);
+
+ /**
+ * The call pointer was released
+ * If the call was not cleared before, report an error
+ * @param event eXosip 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 SIPCall* SIPCall pointer or 0
+ */
+ SIPCall* findSIPCallWithCid(int cid);
+
+ /**
+ * Find a SIPCall with cid and did from eXosip Event
+ * @param cid call ID
+ * @param did domain ID
+ * @return SIPCall* SIPCall pointer or 0
+ */
+ SIPCall* findSIPCallWithCidDid(int cid, int did);
+
+ /**
+ * SIPCall accessor
+ * @param id The call identifier
+ * @return SIPCall* A pointer on SIPCall object
+ */
+ 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 _initDone;
+
+ /** 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;
+
+ /** What is the stun server? */
+ std::string _stunServer;
+
+ /** Local Extern Address is the IP address seen by peers for SIP listener */
+ std::string _localExternAddress;
+
+ /** Local Extern Port is the port seen by peers for SIP listener */
+ unsigned int _localExternPort;
+
+ /** SIP Proxy URL */
+ std::string _proxy;
+
+ /** SIP UserPart */
+ std::string _userpart;
+
+ /** SIP Authenfication name */
+ std::string _authname;
+
+ /** SIP Authenfication password */
+ std::string _password;
+
+ /** Starting sound */
+ AudioRtp _audiortp;
};
#endif
diff --git a/src/user_cfg.h b/src/user_cfg.h
index bafce381485eb5282879e8b20d37eada4829ba7d..6c94010836b867eeced149c586273b6d81b5ba6d 100644
--- a/src/user_cfg.h
+++ b/src/user_cfg.h
@@ -1,7 +1,8 @@
/*
- * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
+ * Authoe: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -23,76 +24,63 @@
#include <stdlib.h>
-// Home directory
-#define HOMEDIR (getenv ("HOME"))
+#define HOMEDIR (getenv ("HOME")) /** Home directory */
-// TODO: change for a \ in Windows Environment
-#define DIR_SEPARATOR_CH '/'
-#define DIR_SEPARATOR_STR "/"
+#define DIR_SEPARATOR_CH '/' /** Directory separator string */
+#define DIR_SEPARATOR_STR "/" /** Directory separator char */
-// Main menu
-#define SIGNALISATION "VoIPLink"
-#define AUDIO "Audio"
-#define VIDEO "Video"
-#define NETWORK "Network"
-#define PREFERENCES "Preferences"
+#define ACCOUNT_SIP0 "SIP0" /** Account type SIP */
+#define ACCOUNT_IAX0 "IAX0" /** Account type IAX */
-#define ACCOUNT_SIP0 "SIP0"
-#define ACCOUNT_IAX0 "IAX0"
+/** User configuration file fields */
+#define AUDIO "Audio" /** Section Audio */
+#define CODECS "ActiveCodecs" /** List of active codecs */
+#define ALSA_CARD_ID_IN "Alsa.cardID_In" /** Soundcard index to use for capture */
+#define ALSA_CARD_ID_OUT "Alsa.cardID_Out" /** Soundcard index to use for playback */
+#define ALSA_FRAME_SIZE "Alsa.framesize" /** Audio layer frame size */
+#define ALSA_PLUGIN "Alsa.plugin" /** Alsa plugin */
+#define ALSA_SAMPLE_RATE "Alsa.sampleRate" /** Audio layer sample rate */
+#define RING_CHOICE "Rings.ringChoice" /** Ringtone */
+#define VOLUME_SPKR "Volume.speakers" /** Speaker volume */
+#define VOLUME_MICRO "Volume.micro" /** Mic volume */
+#define VIDEO "Video" /** Section Video */
-// Fields to fill
-#define SYMMETRIC "VoIPLink.symmetric"
+#define PREFERENCES "Preferences" /** Section Preferences */
+#define CONFIG_DIALPAD "Dialpad.display" /** Display dialpad preferences */
+#define ZONE_TONE "Options.zoneToneChoice" /** Country tone */
+#define VOICEMAIL_NUM "Options.voicemailNumber" /** Voicemail number */
+#define CONFIG_RINGTONE "Ringtones.enable" /** Ringtones preferences */
+#define CONFIG_START "Start.hidden" /** SFLphone starts in the systm tray or not */
+#define CONFIG_POPUP "Window.popup" /** SFLphone pops up on incoming calls or not */
+#define CONFIG_ZEROCONF "Zeroconf.enable" /** Zero configuration networking module */
-#define PLAY_DTMF "DTMF.playDtmf"
-#define PLAY_TONES "DTMF.playTones"
-#define PULSE_LENGTH "DTMF.pulseLength"
-#define SEND_DTMF_AS "DTMF.sendDTMFas"
-#define ALSA_CARD_ID_IN "Alsa.cardID_In"
-#define ALSA_CARD_ID_OUT "Alsa.cardID_Out"
-#define ALSA_SAMPLE_RATE "Alsa.sampleRate"
-#define ALSA_FRAME_SIZE "Alsa.framesize"
-#define ALSA_PLUGIN "Alsa.plugin"
-#define CODECS "ActiveCodecs"
-#define RING_CHOICE "Rings.ringChoice"
-#define ACCOUNT_SIP_COUNT_DEFAULT 4
-#define ACCOUNT_IAX_COUNT_DEFAULT 4
+#define SIGNALISATION "VoIPLink" /** Section Signalisation */
+#define PLAY_DTMF "DTMF.playDtmf" /** Whether or not should play dtmf */
+#define PLAY_TONES "DTMF.playTones" /** Whether or not should play tones */
+#define PULSE_LENGTH "DTMF.pulseLength" /** Length of the DTMF in millisecond */
+#define SEND_DTMF_AS "DTMF.sendDTMFas" /** DTMF send mode */
+#define SYMMETRIC "VoIPLink.symmetric" /** VoIP link type */
-// speakers and volume 0 to 100
-#define VOLUME_SPKR "Volume.speakers"
-#define VOLUME_MICRO "Volume.micro"
-#define ZONE_TONE "Options.zoneToneChoice"
-#define VOICEMAIL_NUM "Options.voicemailNumber"
-// zeroconfig module
-#define CONFIG_ZEROCONF "Zeroconf.enable"
-#define CONFIG_RINGTONE "Ringtones.enable"
-#define CONFIG_DIALPAD "Dialpad.display"
-#define CONFIG_START "Start.hidden"
-#define CONFIG_POPUP "Window.popup"
+#define EMPTY_FIELD "" /** Default value for empty field */
+#define DFT_STUN_SERVER "stun.fwdnet.net:3478" /** Default STUN server address */
+#define YES_STR "1" /** Default YES value */
+#define NO_STR "0" /** Default NO value */
+#define DFT_PULSE_LENGTH_STR "250" /** Default DTMF lenght */
+#define SIP_INFO_STR "0" /** Default DTMF transport mode */
+#define ALSA_DFT_CARD "0" /** Default sound card index */
+#define DFT_VOL_SPKR_STR "100" /** Default speaker volume */
+#define DFT_VOL_MICRO_STR "100" /** Default mic volume */
+#define DFT_RINGTONE "konga.ul" /** Default ringtone */
+#define DFT_ZONE "North America" /** Default geographical zone */
+#define DFT_VOICEMAIL "888" /** Default voicemail number */
+#define DFT_FRAME_SIZE "20" /** Default frame size in millisecond */
+#define DFT_SAMPLE_RATE "44100" /** Default sample rate in HZ */
-// Default values
-#define EMPTY_FIELD ""
-#define DFT_STUN_SERVER "stun.fwdnet.net:3478"
-#define YES_STR "1"
-#define NO_STR "0"
-#define DFT_PULSE_LENGTH_STR "250"
-#define SIP_INFO_STR "0"
-#define ALSA_DFT_CARD "0"
-// volume by default 100%
-#define DFT_VOL_SPKR_STR "100"
-#define DFT_VOL_MICRO_STR "100"
-
-#define DFT_RINGTONE "konga.ul"
-#define DFT_ZONE "North America"
-#define DFT_VOICEMAIL "888"
-#define DFT_FRAME_SIZE "20"
-#define DFT_SAMPLE_RATE "44100"
-
-// zeroconfig default value
#ifdef USE_ZEROCONF
-#define CONFIG_ZEROCONF_DEFAULT_STR "1"
+#define CONFIG_ZEROCONF_DEFAULT_STR "1" /** Default Zero configuration networking module value */
#else
-#define CONFIG_ZEROCONF_DEFAULT_STR "0"
+#define CONFIG_ZEROCONF_DEFAULT_STR "0" /** Default Zero configuration networking module value */
#endif
#endif // __USER_CFG_H__
diff --git a/src/voiplink.h b/src/voiplink.h
index ad7ee15a7e054d35f1a1a6932de3849a2c603fab..4f4672e73e69af1cd87ea51edff18289cdb5e643 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -30,195 +30,290 @@
class AudioCodec;
-//#include "account.h" // for AccountID
-// replaced by:
+/** Define AccountID type */
typedef std::string AccountID;
+/** Define a map that associate a Call object to a call identifier */
typedef std::map<CallID, Call*> CallMap;
/**
- * Listener and manager interface for each VoIP protocol
+ * @file voiplink.h
+ * @brief Listener and manager interface for each VoIP protocol
*/
class VoIPLink {
-public:
- VoIPLink(const AccountID& accountID);
- virtual ~VoIPLink (void);
-
- enum RegistrationState {Unregistered, Trying, Registered, Error, ErrorAuth , ErrorNetwork , ErrorHost};
-
- // Pure virtual functions
- virtual void getEvent (void) = 0;
- virtual bool init (void) = 0;
- virtual bool checkNetwork (void) = 0;
- virtual void terminate (void) = 0;
- virtual bool sendRegister (void) = 0;
- virtual bool sendUnregister (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;
-
- /**
- * Hang up a call
- */
- virtual bool hangup(const CallID& id) = 0;
-
- /**
- * Cancel the call dialing
- */
- virtual bool cancel(const CallID& id) = 0;
-
- /**
- * Put a call on hold
- */
- virtual bool onhold(const CallID& id) = 0;
-
- /**
- * Resume a call from hold state
- */
- virtual bool offhold(const CallID& id) = 0;
-
- /**
- * Transfer a call to specified URI
- */
- virtual bool transfer(const CallID& id, const std::string& to) = 0;
-
- /**
- * Refuse incoming call
- */
- virtual bool refuse(const CallID& id) = 0;
-
- virtual bool carryingDTMFdigits(const CallID& id, char code) = 0;
-
- /**
- * Send text message
- */
- virtual bool sendMessage(const std::string& to, const std::string& body) = 0;
-
- // NOW
- /**
- * Determine if link supports presence information
- */
- virtual bool isContactPresenceSupported() = 0;
-
- /**
- * Register contacts for presence information if supported
- */
- virtual void subscribePresenceForContact(Contact* contact);
-
- /**
- * Publish presence status to server
- */
- virtual void publishPresenceStatus(std::string status);
-
- // 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; }
-
- /**
- * Return parent Account's ID
- */
- AccountID& getAccountID(void) { return _accountID; }
-
- /**
- * Set parent Account's ID
- */
- void setAccountID( const AccountID& accountID) { _accountID = 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);
-
- /**
- * Get registration state
- */
- enum RegistrationState getRegistrationState() { return _registrationState; }
-
- /**
- * Get registration error message, if set.
- */
- int getRegistrationError() { return _registrationError; }
-
- /**
- * Set new registration state
- *
- * We use this function, in case the server needs to PUSH to the
- * GUI when the state changes.
- */
- void setRegistrationState(const enum RegistrationState state,
- const int& errorCode);
-
- /**
- * Same, but with default error value to ""
- */
- void setRegistrationState(const enum RegistrationState state);
-
-private:
- /**
- * Full name used as outgoing Caller ID
- */
- std::string _fullname;
-
- /**
- * Host name used for authentication
- */
- std::string _hostname;
-
- /**
- * ID of parent's Account
- */
- AccountID _accountID;
-
- /**
- * State of registration
- */
- enum RegistrationState _registrationState;
-
- /**
- * Registration error code -> refers to global.h
- */
- int _registrationError;
-
-protected:
- /** 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;
-
-
- /** Whether init() was called already or not
- *
- * This should be used in [IAX|SIP]VoIPLink::init() and terminate(), to
- * indicate that init() was called, or reset by terminate().
- */
- bool _initDone;
+ public:
+
+ /**
+ * Constructor
+ * @param accountID The account identifier
+ */
+ VoIPLink(const AccountID& accountID);
+
+ /**
+ * Virtual destructor
+ */
+ virtual ~VoIPLink (void);
+
+ /** Contains all the state an Voip can be in */
+ enum RegistrationState {Unregistered, Trying, Registered, Error, ErrorAuth , ErrorNetwork , ErrorHost};
+
+ /**
+ * Virtual method
+ * Event listener. Each event send by the call manager is received and handled from here
+ */
+ virtual void getEvent (void) = 0;
+
+ /**
+ * Virtual method
+ * Try to initiate the eXosip engine/thread and set config
+ * @return bool True if OK
+ */
+ virtual bool init (void) = 0;
+
+ /**
+ * Virtual method
+ * Check if a local IP can be found
+ * @return bool True if network is reachable
+ */
+ virtual bool checkNetwork (void) = 0;
+
+ /**
+ * Virtual method
+ * Delete link-related stuuf like calls
+ */
+ virtual void terminate (void) = 0;
+
+ /**
+ * Virtual method
+ * Build and send SIP registration request
+ * @return bool True on success
+ * false otherwise
+ */
+ virtual bool sendRegister (void) = 0;
+
+ /**
+ * Virtual method
+ * Build and send SIP unregistration request
+ * @return bool True on success
+ * false otherwise
+ */
+ virtual bool sendUnregister (void) = 0;
+
+ /**
+ * Place a new call
+ * @param id The call identifier
+ * @param toUrl The Sip address of the recipient of the call
+ * @return Call* The current call
+ */
+ virtual Call* newOutgoingCall(const CallID& id, const std::string& toUrl) = 0;
+ /**
+ * Answer the call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool answer(const CallID& id) = 0;
+
+ /**
+ * Hang up a call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool hangup(const CallID& id) = 0;
+
+ /**
+ * Cancel the call dialing
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool cancel(const CallID& id) = 0;
+
+ /**
+ * Put a call on hold
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool onhold(const CallID& id) = 0;
+
+ /**
+ * Resume a call from hold state
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool offhold(const CallID& id) = 0;
+
+ /**
+ * Transfer a call to specified URI
+ * @param id The call identifier
+ * @param to The recipient of the call
+ * @return bool True on success
+ */
+ virtual bool transfer(const CallID& id, const std::string& to) = 0;
+
+ /**
+ * Refuse incoming call
+ * @param id The call identifier
+ * @return bool True on success
+ */
+ virtual bool refuse(const CallID& id) = 0;
+
+ /**
+ * Send DTMF
+ * @param id The call identifier
+ * @param code The char code
+ * @return bool True on success
+ */
+ virtual bool carryingDTMFdigits(const CallID& id, char code) = 0;
+
+ /**
+ * Send text message
+ */
+ virtual bool sendMessage(const std::string& to, const std::string& body) = 0;
+
+ // NOW
+ /**
+ * Determine if link supports presence information
+ */
+ virtual bool isContactPresenceSupported() = 0;
+
+ /**
+ * Register contacts for presence information if supported
+ */
+ virtual void subscribePresenceForContact(Contact* contact);
+
+ /**
+ * Publish presence status to server
+ */
+ virtual void publishPresenceStatus(std::string status);
+
+ /**
+ * Set the account full name
+ * @param fullname The full name
+ */
+ void setFullName (const std::string& fullname) { _fullname = fullname; }
+
+ /**
+ * Get the account full name
+ * @return std::string The full name
+ */
+ std::string& getFullName (void) { return _fullname; }
+
+ /**
+ * Set the account host name
+ * @param hostname The host name
+ */
+ void setHostName (const std::string& hostname) { _hostname = hostname; }
+
+ /**
+ * Get the account host name
+ * @return std::string The host name
+ */
+ std::string& getHostName (void) { return _hostname; }
+
+ /**
+ * @return AccountID parent Account's ID
+ */
+ AccountID& getAccountID(void) { return _accountID; }
+
+ /**
+ * @param accountID The account identifier
+ */
+ void setAccountID( const AccountID& accountID) { _accountID = accountID; }
+
+ /** Get the call pointer from the call map (protected by mutex)
+ * @param id A Call ID
+ * @return Call* Call pointer or 0
+ */
+ Call* getCall(const CallID& id);
+
+ /**
+ * Get registration state
+ * @return RegistrationState
+ */
+ enum RegistrationState getRegistrationState() { return _registrationState; }
+
+ /**
+ * Get registration error message, if set.
+ */
+ int getRegistrationError() { return _registrationError; }
+
+ /**
+ * Set new registration state
+ * We use this function, in case the server needs to PUSH to the
+ * GUI when the state changes.
+ * @param state The registration state
+ * @param errorCode The error code
+ */
+ void setRegistrationState(const enum RegistrationState state,
+ const int& errorCode);
+
+ /**
+ * Set new registration state
+ * @param state The registration state
+ */
+ void setRegistrationState(const enum RegistrationState state);
+
+ private:
+ /**
+ * Full name used as outgoing Caller ID
+ */
+ std::string _fullname;
+
+ /**
+ * Host name used for authentication
+ */
+ std::string _hostname;
+
+ /**
+ * ID of parent's Account
+ */
+ AccountID _accountID;
+
+ /**
+ * State of registration
+ */
+ enum RegistrationState _registrationState;
+
+ /**
+ * Registration error code -> refers to global.h
+ */
+ int _registrationError;
+
+ protected:
+ /** Add a call to the call map (protected by mutex)
+ * @param call A call pointer with a unique pointer
+ * @return bool 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 bool True if the call was correctly removed
+ */
+ bool removeCall(const CallID& id);
+
+ /**
+ * Remove all the call from the map
+ * @return bool True on success
+ */
+ 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;
+
+ /** Whether init() was called already or not
+ * This should be used in [IAX|SIP]VoIPLink::init() and terminate(), to
+ * indicate that init() was called, or reset by terminate().
+ */
+ bool _initDone;
};
#endif // __VOIP_LINK_H__