diff --git a/src/account.h b/src/account.h
index 764d4628e1c2d780eb18b9ce4cdc23046f150f95..faf33dc162725e39c3dd4ad23985c0bb207b83c3 100644
--- a/src/account.h
+++ b/src/account.h
@@ -123,7 +123,8 @@ class Account{
void setRegistrationState( RegistrationState state );
- //TODO inline?
+ /* inline functions */
+ /* They should be treated like macro definitions by the C++ compiler */
inline std::string getUsername( void ) { return _username; }
inline void setUsername( std::string username) { _username = username; }
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index c72378229e9a7b69f83a5560312ffa7724326414..0ce2e5b785ce3ad700f328c36ab19f6bcf9ff794 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Copyright (C) 2008 2009 Savoir-Faire Linux inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
@@ -19,14 +19,6 @@
#include "alsalayer.h"
-void* ringtoneThreadEntry( void *ptr);
-
-static pthread_t ringtone_thread;
-bool ringtone_thread_is_running;
-
-pthread_mutex_t mut = PTHREAD_MUTEX_INITIALIZER;
-pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
-
// Constructor
AlsaLayer::AlsaLayer( ManagerImpl* manager )
: AudioLayer( manager , ALSA )
@@ -40,21 +32,12 @@ pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
, IDSoundCards()
{
_debug(" Constructor of AlsaLayer called\n");
-
- // The flag to stop the ringtone thread loop
- ringtone_thread_is_running = false;
}
// Destructor
AlsaLayer::~AlsaLayer (void)
{
- _debugAlsa("Close ALSA streams\n");
- closeCaptureStream();
- closePlaybackStream();
- deviceClosed = true;
-
- ringtone_thread_is_running = false;
- //pthread_join(ringtone_thread, NULL);
+ closeLayer();
}
void
@@ -64,8 +47,6 @@ AlsaLayer::closeLayer()
closeCaptureStream();
closePlaybackStream();
deviceClosed = true;
-
- ringtone_thread_is_running = false;
}
bool
@@ -156,27 +137,6 @@ AlsaLayer::stopStream(void)
}
-void* ringtoneThreadEntry( void *ptr )
-{
- while( ringtone_thread_is_running )
- {
- ( ( AlsaLayer *) ptr) -> playTones();
- //sleep(0.1);
- }
-
- /*
- pthread_mutex_lock(&mut);
- while( ((AlsaLayer*)ptr)->_manager->getTelephoneTone() == NULL )
- {
- pthread_cond_wait(&cond, &mut);
- }
- ( AlsaLayer *) ptr -> playTones();
- pthread_mutex_unlock(&mut);*/
-
- return 0;
-}
-
-
void
AlsaLayer::fillHWBuffer( void)
{
@@ -242,13 +202,6 @@ AlsaLayer::putUrgent(void* buffer, int toCopy)
return nbBytes;
}
-void AlsaLayer::trigger_thread(void)
-{
- _debug("Wake up the ringtone thread\n");
- pthread_cond_broadcast(&cond);
-}
-
-
int
AlsaLayer::canGetMic()
{
@@ -299,13 +252,6 @@ AlsaLayer::playTones( void )
int frames;
int maxBytes;
- //pthread_mutex_lock(&mut);
- //while(!_manager-> getTelephoneTone() && !_manager->getTelephoneFile())
- //{
- _debug("Make the ringtone thread wait\n");
- pthread_cond_wait(&cond, &mut);
- //}
-
//frames = _periodSize ;
frames = 940 ;
maxBytes = frames * sizeof(SFLDataFormat) ;
@@ -324,7 +270,6 @@ AlsaLayer::playTones( void )
}
// free the temporary data buffer
free( out ); out = 0;
- //pthread_mutex_unlock(&mut);
}
@@ -445,18 +390,7 @@ bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){
}
if( type == 1 ){
- /*if( (err = snd_async_add_pcm_handler( &_AsyncHandler, pcm_handle , AlsaCallBack, this ) < 0)){
- _debugAlsa(" Unable to install the async callback handler (%s)\n", snd_strerror(err));
- return false;
- }*/
- // So the loop could start when the ringtone thread entry function is reached
- ringtone_thread_is_running = true;
- /*if( pthread_create(&ringtone_thread, NULL, ringtoneThreadEntry, this) != 0 )
- {
- _debug("Unable to start the ringtone posix thread\n");
- return false;
- }*/
}
snd_pcm_sw_params_free( swparams );
diff --git a/src/audio/alsalayer.h b/src/audio/alsalayer.h
index c9dfa016d2cf8d61d68df15d815d7cc8f13eb088..4111d42ad9d025097bfc66a9b18facfc404e3142 100644
--- a/src/audio/alsalayer.h
+++ b/src/audio/alsalayer.h
@@ -21,8 +21,7 @@
#define _ALSA_LAYER_H
#include "audiolayer.h"
-
-#include <pthread.h>
+#include <alsa/asoundlib.h>
class RingBuffer;
class ManagerImpl;
@@ -48,8 +47,6 @@ class AlsaLayer : public AudioLayer {
*/
~AlsaLayer(void);
- void trigger_thread(void);
-
void closeLayer( void );
/**
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 658ac3825deaad2d14fec6c31e547d60648d4ed3..c1273ab27dada2310618333c97c5f60d661de297 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -22,21 +22,15 @@
#ifndef _AUDIO_LAYER_H
#define _AUDIO_LAYER_H
-#include "../global.h"
-#include "../manager.h"
+#include "global.h"
#include "audiodevice.h"
#include "ringbuffer.h"
+#include "manager.h"
#include <cc++/thread.h> // for ost::Mutex
-#include <vector>
-#include <alsa/asoundlib.h>
#include <pulse/pulseaudio.h>
-#include <iostream>
-#include <istream>
-#include <sstream>
-
#define FRAME_PER_BUFFER 160
@@ -86,8 +80,6 @@ class AudioLayer {
virtual void closeLayer( void ) = 0;
- virtual void trigger_thread(void)=0;
-
/**
* Check if no devices are opened, otherwise close them.
* Then open the specified devices by calling the private functions open_device
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 85ec5fea464f198be3e410b68d8cd130dc562bb9..73dbcdfc10136ad503a16fb900d3d82f864f51b7 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -52,14 +52,16 @@ AudioRtp::~AudioRtp (void) {
int
AudioRtp::createNewSession (SIPCall *ca) {
- //ost::MutexLock m(_threadMutex);
-
- // something should stop the thread before...
- if ( _RTXThread != 0 ) {
- _debug("! ARTP Failure: Thread already exists..., stopping it\n");
- delete _RTXThread; _RTXThread = 0;
- //return -1;
- }
+ ost::MutexLock m(_threadMutex);
+
+ // something should stop the thread before...
+ if ( _RTXThread != 0 ) {
+ _debug("**********************************************************\n");
+ _debug("! ARTP Failure: Thread already exists..., stopping it\n");
+ _debug("**********************************************************\n");
+ delete _RTXThread; _RTXThread = 0;
+ //return -1;
+ }
// Start RTP Send/Receive threads
_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
@@ -79,7 +81,7 @@ AudioRtp::createNewSession (SIPCall *ca) {
void
AudioRtp::closeRtpSession () {
- //ost::MutexLock m(_threadMutex);
+ ost::MutexLock m(_threadMutex);
// This will make RTP threads finish.
// _debug("Stopping AudioRTP\n");
try {
@@ -88,8 +90,8 @@ AudioRtp::closeRtpSession () {
_debugException("! ARTP Exception: when stopping audiortp\n");
throw;
}
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- audiolayer->stopStream();
+ //AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ //audiolayer->stopStream();
}
////////////////////////////////////////////////////////////////////////////////
@@ -99,7 +101,7 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _
_sym(sym), micData(NULL), micDataConverted(NULL), micDataEncoded(NULL), spkrDataDecoded(NULL), spkrDataConverted(NULL),
converter(NULL), _layerSampleRate(),_codecSampleRate(), _layerFrameSize(), _audiocodec(NULL)
{
- setCancel(cancelDeferred);
+ setCancel(cancelDefault);
// AudioRtpRTX should be close if we change sample rate
// TODO: Change bind address according to user settings.
// TODO: this should be the local ip not the external (router) IP
diff --git a/src/global.h b/src/global.h
index 7a2a1b5793dcdb59213d5f3f684a57c864b40272..45332563dc0bbb28a0f22a777d997deafa5c32f5 100644
--- a/src/global.h
+++ b/src/global.h
@@ -29,6 +29,8 @@
#include <string>
#include <stdlib.h>
#include <sstream>
+#include <map>
+#include <vector>
#define SFLPHONED_VERSION "0.9.2-4" /** Version number */
diff --git a/src/iaxaccount.cpp b/src/iaxaccount.cpp
index 4de19c54cc995f4d39954775851299b67860451a..504a1cc7106dfa7d1c441a64b7a186fc584152bd 100644
--- a/src/iaxaccount.cpp
+++ b/src/iaxaccount.cpp
@@ -22,8 +22,8 @@
#include "iaxaccount.h"
#include "iaxvoiplink.h"
- IAXAccount::IAXAccount(const AccountID& accountID)
-: Account(accountID, "iax2")
+IAXAccount::IAXAccount(const AccountID& accountID)
+ : Account(accountID, "iax2")
{
_link = new IAXVoIPLink(accountID);
}
@@ -37,17 +37,12 @@ IAXAccount::~IAXAccount()
int IAXAccount::registerVoIPLink()
{
- IAXVoIPLink *thislink;
-
_link->init();
- thislink = dynamic_cast<IAXVoIPLink*> (_link);
- if (thislink) {
- // Stuff needed for IAX registration
- thislink->setHost(Manager::instance().getConfigString(_accountID, HOSTNAME));
- thislink->setUser(Manager::instance().getConfigString(_accountID, USERNAME));
- thislink->setPass(Manager::instance().getConfigString(_accountID, PASSWORD));
- }
+ // Stuff needed for IAX registration
+ setHostname(Manager::instance().getConfigString(_accountID, HOSTNAME));
+ setUsername(Manager::instance().getConfigString(_accountID, USERNAME));
+ setPassword(Manager::instance().getConfigString(_accountID, PASSWORD));
_link->sendRegister( _accountID );
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 679e2527ce887adab5dbaeefbf9ec043d17af82f..f301014b686462ba999989ae0919967024cf508b 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -18,15 +18,12 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "iaxvoiplink.h"
-#include "global.h" // for _debug
#include "iaxcall.h"
#include "eventthread.h"
#include "iaxaccount.h"
-
#include "manager.h"
#include "audio/audiolayer.h"
-//#include <iax/iax-client.h>
#include <math.h>
#include <dlfcn.h>
@@ -41,182 +38,182 @@
#define MUSIC_ONHOLD true
#define CHK_VALID_CALL if (call == NULL) { _debug("IAX: Call doesn't exists\n"); \
- return false; }
+ return false; }
- IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
+ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID)
{
- _evThread = new EventThread(this);
- _regSession = NULL;
- _nextRefreshStamp = 0;
+ _evThread = new EventThread(this);
+ _regSession = NULL;
+ _nextRefreshStamp = 0;
- // to get random number for RANDOM_PORT
- srand (time(NULL));
+ // to get random number for RANDOM_PORT
+ srand (time(NULL));
- audiolayer = NULL;
+ audiolayer = NULL;
- converter = new SamplerateConverter();
+ converter = new SamplerateConverter();
- int nbSamplesMax = (int) ( converter->getFrequence() * converter->getFramesize() / 1000 );
+ int nbSamplesMax = (int) ( converter->getFrequence() * converter->getFramesize() / 1000 );
- micData = new SFLDataFormat[nbSamplesMax];
- micDataConverted = new SFLDataFormat[nbSamplesMax];
- micDataEncoded = new unsigned char[nbSamplesMax];
+ micData = new SFLDataFormat[nbSamplesMax];
+ micDataConverted = new SFLDataFormat[nbSamplesMax];
+ micDataEncoded = new unsigned char[nbSamplesMax];
- spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
+ spkrDataConverted = new SFLDataFormat[nbSamplesMax];
+ spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
}
IAXVoIPLink::~IAXVoIPLink()
{
- delete _evThread; _evThread = NULL;
- _regSession = NULL; // shall not delete it
- terminate();
+ delete _evThread; _evThread = NULL;
+ _regSession = NULL; // shall not delete it
+ terminate();
- audiolayer = NULL;
+ audiolayer = NULL;
- delete [] micData; micData = NULL;
- delete [] micDataConverted; micDataConverted = NULL;
- delete [] micDataEncoded; micDataEncoded = NULL;
+ delete [] micData; micData = NULL;
+ delete [] micDataConverted; micDataConverted = NULL;
+ delete [] micDataEncoded; micDataEncoded = NULL;
- delete [] spkrDataDecoded; spkrDataDecoded = NULL;
- delete [] spkrDataConverted; spkrDataConverted = NULL;
+ delete [] spkrDataDecoded; spkrDataDecoded = NULL;
+ delete [] spkrDataConverted; spkrDataConverted = NULL;
}
- bool
+ bool
IAXVoIPLink::init()
{
- // If it was done, don't do it again, until we call terminate()
- if (_initDone)
- return false;
-
- bool returnValue = false;
- //_localAddress = "127.0.0.1";
- // port 0 is default
- // iax_enable_debug(); have to enable debug when compiling iax...
- int port = IAX_DEFAULT_PORTNO;
- int last_port = 0;
- int nbTry = 3;
-
- while (port != IAX_FAILURE && nbTry) {
- last_port = port;
- port = iax_init(port);
- if ( port < 0 ) {
- _debug("IAX Warning: already initialize on port %d\n", last_port);
- port = RANDOM_IAX_PORT;
- } else if (port == IAX_FAILURE) {
- _debug("IAX Fail to start on port %d", last_port);
- port = RANDOM_IAX_PORT;
- } else {
- _debug("IAX Info: listening on port %d\n", last_port);
- _localPort = last_port;
- returnValue = true;
- _evThread->start();
+ // If it was done, don't do it again, until we call terminate()
+ if (initDone())
+ return false;
- audiolayer = Manager::instance().getAudioDriver();
- break;
+ bool returnValue = false;
+ //_localAddress = "127.0.0.1";
+ // port 0 is default
+ // iax_enable_debug(); have to enable debug when compiling iax...
+ int port = IAX_DEFAULT_PORTNO;
+ int last_port = 0;
+ int nbTry = 3;
+
+ while (port != IAX_FAILURE && nbTry) {
+ last_port = port;
+ port = iax_init(port);
+ if ( port < 0 ) {
+ _debug("IAX Warning: already initialize on port %d\n", last_port);
+ port = RANDOM_IAX_PORT;
+ } else if (port == IAX_FAILURE) {
+ _debug("IAX Fail to start on port %d", last_port);
+ port = RANDOM_IAX_PORT;
+ } else {
+ _debug("IAX Info: listening on port %d\n", last_port);
+ _localPort = last_port;
+ returnValue = true;
+ _evThread->start();
+
+ audiolayer = Manager::instance().getAudioDriver();
+ break;
+ }
+ nbTry--;
+
+ initDone(true);
}
- nbTry--;
-
- _initDone = true;
- }
- if (port == IAX_FAILURE || nbTry==0) {
- _debug("Fail to initialize iax\n");
+ if (port == IAX_FAILURE || nbTry==0) {
+ _debug("Fail to initialize iax\n");
- _initDone = false;
- }
+ initDone(false);
+ }
- return returnValue;
+ return returnValue;
}
- void
+ void
IAXVoIPLink::terminate()
{
- // If it was done, don't do it again, until we call init()
- if (!_initDone)
- return;
+ // If it was done, don't do it again, until we call init()
+ if (!initDone())
+ return;
- // iaxc_shutdown();
+ // iaxc_shutdown();
- // Hangup all calls
- terminateIAXCall();
+ // Hangup all calls
+ terminateIAXCall();
- _initDone = false;
+ initDone(false);
}
- void
+ void
IAXVoIPLink::terminateIAXCall()
{
- std::string reason = "Dumped Call";
- ost::MutexLock m(_callMapMutex);
- CallMap::iterator iter = _callMap.begin();
- IAXCall *call;
- while( iter != _callMap.end() ) {
- call = dynamic_cast<IAXCall*>(iter->second);
- if (call) {
- _mutexIAX.enterMutex();
- iax_hangup(call->getSession(), (char*)reason.c_str());
- _mutexIAX.leaveMutex();
- call->setSession(NULL);
- delete call; call = NULL;
+ std::string reason = "Dumped Call";
+ ost::MutexLock m(_callMapMutex);
+ CallMap::iterator iter = _callMap.begin();
+ IAXCall *call;
+ while( iter != _callMap.end() ) {
+ call = dynamic_cast<IAXCall*>(iter->second);
+ if (call) {
+ _mutexIAX.enterMutex();
+ iax_hangup(call->getSession(), (char*)reason.c_str());
+ _mutexIAX.leaveMutex();
+ call->setSession(NULL);
+ delete call; call = NULL;
+ }
+ iter++;
}
- iter++;
- }
- _callMap.clear();
+ _callMap.clear();
}
- void
+ void
IAXVoIPLink::getEvent()
{
- IAXCall* call = NULL;
+ IAXCall* call = NULL;
- // lock iax_ stuff..
- _mutexIAX.enterMutex();
- iax_event* event = NULL;
-
- while ( (event = iax_get_event(IAX_NONBLOCKING)) != NULL ) {
- // If we received an 'ACK', libiax2 tells apps to ignore them.
- if (event->etype == IAX_EVENT_NULL) {
- continue;
+ // lock iax_ stuff..
+ _mutexIAX.enterMutex();
+ iax_event* event = NULL;
+
+ while ( (event = iax_get_event(IAX_NONBLOCKING)) != NULL ) {
+ // If we received an 'ACK', libiax2 tells apps to ignore them.
+ if (event->etype == IAX_EVENT_NULL) {
+ continue;
+ }
+
+ //_debug ("Receive IAX Event: %d (0x%x)\n", event->etype, event->etype);
+
+ call = iaxFindCallBySession(event->session);
+
+ if (call) {
+ // We know that call, deal with it
+ iaxHandleCallEvent(event, call);
+ //_audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( call -> getAudioCodec() );
+ }
+ else if (event->session && event->session == _regSession) {
+ // This is a registration session, deal with it
+ iaxHandleRegReply(event);
+ }
+ else {
+ // We've got an event before it's associated with any call
+ iaxHandlePrecallEvent(event);
+ }
+
+ iax_event_free(event);
}
+ _mutexIAX.leaveMutex();
- //_debug ("Receive IAX Event: %d (0x%x)\n", event->etype, event->etype);
-
- call = iaxFindCallBySession(event->session);
+ // Do the doodle-moodle to send audio from the microphone to the IAX channel.
+ sendAudioFromMic();
- if (call) {
- // We know that call, deal with it
- iaxHandleCallEvent(event, call);
- //_audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( call -> getAudioCodec() );
- }
- else if (event->session && event->session == _regSession) {
- // This is a registration session, deal with it
- iaxHandleRegReply(event);
+ // Refresh registration.
+ if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
+ sendRegister("");
}
- else {
- // We've got an event before it's associated with any call
- iaxHandlePrecallEvent(event);
- }
-
- iax_event_free(event);
- }
- _mutexIAX.leaveMutex();
-
- // Do the doodle-moodle to send audio from the microphone to the IAX channel.
- sendAudioFromMic();
- // Refresh registration.
- if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
- sendRegister("");
- }
-
- // thread wait 3 millisecond
- _evThread->sleep(3);
- free(event);
+ // thread wait 3 millisecond
+ _evThread->sleep(3);
+ free(event);
}
- void
+ void
IAXVoIPLink::sendAudioFromMic(void)
{
int maxBytesToGet, availBytesFromMic, bytesAvail, nbSample, compSize;
@@ -227,7 +224,7 @@ IAXVoIPLink::sendAudioFromMic(void)
updateAudiolayer();
IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
-
+
if (!currentCall) {
// Let's mind our own business.
return;
@@ -236,90 +233,90 @@ IAXVoIPLink::sendAudioFromMic(void)
if( currentCall -> getAudioCodec() < 0 )
return;
- // Just make sure the currentCall is in state to receive audio right now.
- //_debug("Here we get: connectionState: %d state: %d \n",
- //currentCall->getConnectionState(),
- //currentCall->getState());
+ // Just make sure the currentCall is in state to receive audio right now.
+ //_debug("Here we get: connectionState: %d state: %d \n",
+ //currentCall->getConnectionState(),
+ //currentCall->getState());
- if (currentCall->getConnectionState() != Call::Connected ||
- currentCall->getState() != Call::Active) {
- return;
- }
+ if (currentCall->getConnectionState() != Call::Connected ||
+ currentCall->getState() != Call::Active) {
+ return;
+ }
- ac = currentCall->getCodecMap().getCodec( currentCall -> getAudioCodec() );
- if (!ac) {
- // Audio codec still not determined.
- if (audiolayer) {
- // To keep latency low..
- //audiolayer->flushMic();
+ ac = currentCall->getCodecMap().getCodec( currentCall -> getAudioCodec() );
+ if (!ac) {
+ // Audio codec still not determined.
+ if (audiolayer) {
+ // To keep latency low..
+ //audiolayer->flushMic();
+ }
+ return;
}
- return;
- }
- // Send sound here
- if (audiolayer) {
+ // Send sound here
+ if (audiolayer) {
- // we have to get 20ms of data from the mic *20/1000 = /50
- // rate/50 shall be lower than IAX__20S_48KHZ_MAX
- maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
-
- // available bytes inside ringbuffer
- availBytesFromMic = audiolayer->canGetMic();
+ // we have to get 20ms of data from the mic *20/1000 = /50
+ // rate/50 shall be lower than IAX__20S_48KHZ_MAX
+ maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
- if (availBytesFromMic < maxBytesToGet) {
- // We need packets full!
- return;
- }
+ // available bytes inside ringbuffer
+ availBytesFromMic = audiolayer->canGetMic();
- // take the lowest
- bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
- //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
+ if (availBytesFromMic < maxBytesToGet) {
+ // We need packets full!
+ return;
+ }
- // Get bytes from micRingBuffer to data_from_mic
- nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
+ // take the lowest
+ bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+ //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
- // resample
- nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample );
+ // Get bytes from micRingBuffer to data_from_mic
+ nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
- // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+ // resample
+ nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample );
- // Send it out!
- _mutexIAX.enterMutex();
- // Make sure the session and the call still exists.
- if (currentCall->getSession()) {
- if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
- _debug("IAX: Error sending voice data.\n");
- }
+ // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
+ compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+
+ // Send it out!
+ _mutexIAX.enterMutex();
+ // Make sure the session and the call still exists.
+ if (currentCall->getSession()) {
+ if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
+ _debug("IAX: Error sending voice data.\n");
+ }
+ }
+ _mutexIAX.leaveMutex();
}
- _mutexIAX.leaveMutex();
- }
}
- IAXCall*
+ IAXCall*
IAXVoIPLink::getIAXCall(const CallID& id)
{
- Call* call = getCall(id);
- if (call) {
- return dynamic_cast<IAXCall*>(call);
- }
- return NULL;
+ Call* call = getCall(id);
+ if (call) {
+ return dynamic_cast<IAXCall*>(call);
+ }
+ return NULL;
}
- int
+ int
IAXVoIPLink::sendRegister(AccountID id)
{
IAXAccount *account;
bool result;
result = false;
- account = dynamic_cast<IAXAccount *> (Manager::instance().getAccount(id));
- if (_host.empty()) {
+ account = dynamic_cast<IAXAccount *> (getAccountPtr());
+ if (account->getHostname().empty()) {
return false;
}
- if (_user.empty()) {
+ if (account->getUsername().empty()) {
return false;
}
@@ -338,16 +335,16 @@ IAXVoIPLink::sendRegister(AccountID id)
} else {
// refresh
// last reg
- char host[_host.length()+1];
- strcpy(host, _host.c_str());
- char user[_user.length()+1];
- strcpy(user, _user.c_str());
- char pass[_pass.length()+1];
- strcpy(pass, _pass.c_str());
+ //char host[_host.length()+1];
+ //strcpy(host, _host.c_str());
+ //char user[_user.length()+1];
+ //strcpy(user, _user.c_str());
+ //char pass[_pass.length()+1];
+ //strcpy(pass, _pass.c_str());
// iax_register doesn't use const char*
- _debug("IAX Sending registration to %s with user %s\n", host, user);
- int val = iax_register(_regSession, host, user, pass, 120);
+ _debug("IAX Sending registration to %s with user %s\n", account->getHostname().c_str() , account->getUsername().c_str() );
+ int val = iax_register(_regSession, account->getHostname().data(), account->getUsername().data(), account->getPassword().data(), 120);
_debug ("Return value: %d\n", val);
// set the time-out to 15 seconds, after that, resend a registration request.
// until we unregister.
@@ -355,432 +352,438 @@ IAXVoIPLink::sendRegister(AccountID id)
result = true;
account->setRegistrationState(Trying);
- }
+ }
- // unlock
- _mutexIAX.leaveMutex();
+ // unlock
+ _mutexIAX.leaveMutex();
- return result;
+ return result;
}
-
-
- int
+ int
IAXVoIPLink::sendUnregister(AccountID id)
{
- _mutexIAX.enterMutex();
- if (_regSession) {
- /** @todo Should send a REGREL in sendUnregister()... */
+ IAXAccount *account;
- //iax_send_regrel(); doesn't exist yet :)
- iax_destroy(_regSession);
+ account = dynamic_cast<IAXAccount*>(getAccountPtr());
+ if(!account)
+ return 1;
- _regSession = NULL;
- }
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ if (_regSession) {
+ /** @todo Should send a REGREL in sendUnregister()... */
+ //iax_send_regrel(); doesn't exist yet :)
+ iax_destroy(_regSession);
+ _regSession = NULL;
+ }
+ _mutexIAX.leaveMutex();
- _nextRefreshStamp = 0;
+ _nextRefreshStamp = 0;
- _debug("IAX2 send unregister\n");
- //setRegistrationState(Unregistered);
+ _debug("IAX2 send unregister\n");
+ account->setRegistrationState(Unregistered);
- return SUCCESS;
+ return SUCCESS;
}
- Call*
+ Call*
IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
- IAXCall* call = new IAXCall(id, Call::Outgoing);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
-
-
- if (call) {
- call->setPeerNumber(toUrl);
+ IAXCall* call = new IAXCall(id, Call::Outgoing);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- if ( iaxOutgoingInvite(call) ) {
- call->setConnectionState(Call::Progressing);
- call->setState(Call::Active);
- addCall(call);
- } else {
- delete call; call = NULL;
+ if (call) {
+ call->setPeerNumber(toUrl);
+
+ if ( iaxOutgoingInvite(call) ) {
+ call->setConnectionState(Call::Progressing);
+ call->setState(Call::Active);
+ addCall(call);
+ } else {
+ delete call; call = NULL;
+ }
}
- }
- return call;
+ return call;
}
- bool
+ bool
IAXVoIPLink::answer(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ IAXCall* call = getIAXCall(id);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- _mutexIAX.enterMutex();
- iax_answer(call->getSession());
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_answer(call->getSession());
+ _mutexIAX.leaveMutex();
- call->setState(Call::Active);
- call->setConnectionState(Call::Connected);
- // Start audio
- audiolayer->startStream();
- //audiolayer->flushMic();
+ call->setState(Call::Active);
+ call->setConnectionState(Call::Connected);
+ // Start audio
+ audiolayer->startStream();
+ //audiolayer->flushMic();
- return true;
+ return true;
}
- bool
+ bool
IAXVoIPLink::hangup(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
- std::string reason = "Dumped Call";
- CHK_VALID_CALL;
-
- _mutexIAX.enterMutex();
- iax_hangup(call->getSession(), (char*) reason.c_str());
- _mutexIAX.leaveMutex();
- call->setSession(NULL);
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- removeCall(id);
- return true;
+ IAXCall* call = getIAXCall(id);
+ std::string reason = "Dumped Call";
+ CHK_VALID_CALL;
+
+ _mutexIAX.enterMutex();
+ iax_hangup(call->getSession(), (char*) reason.c_str());
+ _mutexIAX.leaveMutex();
+ call->setSession(NULL);
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
+ removeCall(id);
+ return true;
}
- bool
+ bool
IAXVoIPLink::onhold(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- //if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
+ //if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
- _mutexIAX.enterMutex();
- iax_quelch_moh(call->getSession() , MUSIC_ONHOLD);
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_quelch_moh(call->getSession() , MUSIC_ONHOLD);
+ _mutexIAX.leaveMutex();
- call->setState(Call::Hold);
- return true;
+ call->setState(Call::Hold);
+ return true;
}
- bool
+ bool
IAXVoIPLink::offhold(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- //if (call->getState() == Call::Active) { _debug("Call is already active\n"); return false; }
- _mutexIAX.enterMutex();
- iax_unquelch(call->getSession());
- _mutexIAX.leaveMutex();
- audiolayer->startStream();
- call->setState(Call::Active);
- return true;
+ //if (call->getState() == Call::Active) { _debug("Call is already active\n"); return false; }
+ _mutexIAX.enterMutex();
+ iax_unquelch(call->getSession());
+ _mutexIAX.leaveMutex();
+ audiolayer->startStream();
+ call->setState(Call::Active);
+ return true;
}
- bool
+ bool
IAXVoIPLink::transfer(const CallID& id, const std::string& to)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- char callto[to.length()+1];
- strcpy(callto, to.c_str());
+ char callto[to.length()+1];
+ strcpy(callto, to.c_str());
- _mutexIAX.enterMutex();
- iax_transfer(call->getSession(), callto);
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_transfer(call->getSession(), callto);
+ _mutexIAX.leaveMutex();
- return true;
+ return true;
- // should we remove it?
- // removeCall(id);
+ // should we remove it?
+ // removeCall(id);
}
- bool
+ bool
IAXVoIPLink::refuse(const CallID& id)
{
- IAXCall* call = getIAXCall(id);
- std::string reason = "Call rejected manually.";
+ IAXCall* call = getIAXCall(id);
+ std::string reason = "Call rejected manually.";
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- _mutexIAX.enterMutex();
- iax_reject(call->getSession(), (char*) reason.c_str());
- _mutexIAX.leaveMutex();
- removeCall(id);
+ _mutexIAX.enterMutex();
+ iax_reject(call->getSession(), (char*) reason.c_str());
+ _mutexIAX.leaveMutex();
+ removeCall(id);
- return true;
+ return true;
}
- bool
+ bool
IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
- IAXCall* call = getIAXCall(id);
+ IAXCall* call = getIAXCall(id);
- CHK_VALID_CALL;
+ CHK_VALID_CALL;
- _mutexIAX.enterMutex();
- iax_send_dtmf(call->getSession(), code);
- _mutexIAX.leaveMutex();
+ _mutexIAX.enterMutex();
+ iax_send_dtmf(call->getSession(), code);
+ _mutexIAX.leaveMutex();
- return true;
+ return true;
}
- bool
+ bool
IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
{
- struct iax_session *newsession;
- ost::MutexLock m(_mutexIAX);
-
- newsession = iax_session_new();
- if (!newsession) {
- _debug("IAX Error: Can't make new session for a new call\n");
- return false;
- }
- call->setSession(newsession);
- /* reset activity and ping "timers" */
- // iaxc_note_activity(callNo);
-
- std::string strNum = _user + ":" + _pass + "@" + _host + "/" + call->getPeerNumber();
+ struct iax_session *newsession;
+ ost::MutexLock m(_mutexIAX);
+ std::string username, strNum;
+ char *lang=NULL;
+ int wait, audio_format_preferred, audio_format_capability;
+ char user[username.length()+1];
+ char num[strNum.length()+1];
+ IAXAccount *account;
- char user[_user.length()+1];
- strcpy(user, _user.c_str());
+ newsession = iax_session_new();
+ if (!newsession) {
+ _debug("IAX Error: Can't make new session for a new call\n");
+ return false;
+ }
+ call->setSession(newsession);
- char num[strNum.length()+1];
- strcpy(num, strNum.c_str());
+ account = dynamic_cast<IAXAccount*> (getAccountPtr());
+ username = account->getUsername();
+ strNum = username + ":" + account->getPassword() + "@" + account->getHostname() + "/" + call->getPeerNumber();
+ strcpy(user, username.c_str());
+ strcpy(num, strNum.c_str());
- char* lang = NULL;
- int wait = 0;
- /** @todo Make preference dynamic, and configurable */
- int audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat());
- int audio_format_capability = call->getSupportedFormat();
+ wait = 0;
+ /** @todo Make preference dynamic, and configurable */
+ audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat());
+ audio_format_capability = call->getSupportedFormat();
- _debug("IAX New call: %s\n", num);
- iax_call(newsession, user, user, num, lang, wait, audio_format_preferred, audio_format_capability);
+ _debug("IAX New call: %s\n", num);
+ iax_call(newsession, user, user, num, lang, wait, audio_format_preferred, audio_format_capability);
- return true;
+ return true;
}
- IAXCall*
+ IAXCall*
IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
{
- // access to callMap shoud use that
- // the code below is like findSIPCallWithCid()
- ost::MutexLock m(_callMapMutex);
- IAXCall* call = NULL;
- CallMap::iterator iter = _callMap.begin();
- while(iter != _callMap.end()) {
- call = dynamic_cast<IAXCall*>(iter->second);
- if (call && call->getSession() == session) {
- return call;
+ // access to callMap shoud use that
+ // the code below is like findSIPCallWithCid()
+ ost::MutexLock m(_callMapMutex);
+ IAXCall* call = NULL;
+ CallMap::iterator iter = _callMap.begin();
+ while(iter != _callMap.end()) {
+ call = dynamic_cast<IAXCall*>(iter->second);
+ if (call && call->getSession() == session) {
+ return call;
+ }
+ iter++;
}
- iter++;
- }
- return NULL; // not found
+ return NULL; // not found
}
- void
+ void
IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
{
- // call should not be 0
- // note activity?
- //
- CallID id = call->getCallId();
-
- switch(event->etype) {
- case IAX_EVENT_HANGUP:
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_REJECT:
- //Manager::instance().peerHungupCall(id);
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- call->setConnectionState(Call::Connected);
- call->setState(Call::Error);
- Manager::instance().callFailure(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_ACCEPT:
- // Call accepted over there by the computer, not the user yet.
- if (event->ies.format) {
- call->setFormat(event->ies.format);
- }
- break;
-
- case IAX_EVENT_ANSWER:
- if (call->getConnectionState() != Call::Connected){
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
-
- if (event->ies.format) {
- // Should not get here, should have been set in EVENT_ACCEPT
- call->setFormat(event->ies.format);
- }
-
- Manager::instance().peerAnsweredCall(id);
- //audiolayer->flushMic();
- audiolayer->startStream();
- // start audio here?
- } else {
- // deja connecté ?
- }
- break;
-
- case IAX_EVENT_BUSY:
- call->setConnectionState(Call::Connected);
- call->setState(Call::Busy);
- Manager::instance().callBusy(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_VOICE:
- //_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
- if( !audiolayer -> isCaptureActive())
- audiolayer->startStream();
- iaxHandleVoiceEvent(event, call);
- break;
-
- case IAX_EVENT_TEXT:
- break;
-
- case IAX_EVENT_RINGA:
- call->setConnectionState(Call::Ringing);
- Manager::instance().peerRingingCall(call->getCallId());
- break;
-
- case IAX_IE_MSGCOUNT:
- break;
- case IAX_EVENT_PONG:
- break;
-
- case IAX_EVENT_URL:
- break;
-
- // case IAX_EVENT_CNG: ??
- // break;
-
- case IAX_EVENT_TIMEOUT:
- break;
-
- case IAX_EVENT_TRANSFER:
- break;
-
- default:
- _debug("Unknown event type (in call event): %d\n", event->etype);
-
- }
+ // call should not be 0
+ // note activity?
+ //
+ CallID id = call->getCallId();
+
+ switch(event->etype) {
+ case IAX_EVENT_HANGUP:
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_REJECT:
+ //Manager::instance().peerHungupCall(id);
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Error);
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_ACCEPT:
+ // Call accepted over there by the computer, not the user yet.
+ if (event->ies.format) {
+ call->setFormat(event->ies.format);
+ }
+ break;
+
+ case IAX_EVENT_ANSWER:
+ if (call->getConnectionState() != Call::Connected){
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ if (event->ies.format) {
+ // Should not get here, should have been set in EVENT_ACCEPT
+ call->setFormat(event->ies.format);
+ }
+
+ Manager::instance().peerAnsweredCall(id);
+ //audiolayer->flushMic();
+ audiolayer->startStream();
+ // start audio here?
+ } else {
+ // deja connecté ?
+ }
+ break;
+
+ case IAX_EVENT_BUSY:
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Busy);
+ Manager::instance().callBusy(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_VOICE:
+ //_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
+ //TODO Check this method
+ if( !audiolayer -> isCaptureActive())
+ audiolayer->startStream();
+ iaxHandleVoiceEvent(event, call);
+ break;
+
+ case IAX_EVENT_TEXT:
+ break;
+
+ case IAX_EVENT_RINGA:
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ break;
+
+ case IAX_IE_MSGCOUNT:
+ break;
+ case IAX_EVENT_PONG:
+ break;
+
+ case IAX_EVENT_URL:
+ break;
+
+ // case IAX_EVENT_CNG: ??
+ // break;
+
+ case IAX_EVENT_TIMEOUT:
+ break;
+
+ case IAX_EVENT_TRANSFER:
+ break;
+
+ default:
+ _debug("Unknown event type (in call event): %d\n", event->etype);
+
+ }
}
/* Handle audio event, VOICE packet received */
- void
+ void
IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
{
-
unsigned char *data;
unsigned int size, max, nbInt16;
int expandedSize, nbSample;
AudioCodec *ac;
- // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
- // were triggered
- if (!event->datalen) {
- // Skip this empty packet.
- //_debug("IAX: Skipping empty jitter-buffer interpolated packet\n");
- return;
- }
-
- if (audiolayer) {
- // On-the-fly codec changing (normally, when we receive a full packet)
- // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
- // - subclass holds the voiceformat property.
- if (event->subclass && event->subclass != call->getFormat()) {
- call->setFormat(event->subclass);
+ // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
+ // were triggered
+ if (!event->datalen) {
+ // Skip this empty packet.
+ //_debug("IAX: Skipping empty jitter-buffer interpolated packet\n");
+ return;
}
- //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
- ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- data = (unsigned char*)event->data;
- size = event->datalen;
+ if (audiolayer) {
+ // On-the-fly codec changing (normally, when we receive a full packet)
+ // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
+ // - subclass holds the voiceformat property.
+ if (event->subclass && event->subclass != call->getFormat()) {
+ call->setFormat(event->subclass);
+ }
+ //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
+ ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- // Decode data with relevant codec
- max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
+ data = (unsigned char*)event->data;
+ size = event->datalen;
- if (size > max) {
- _debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
- size = max;
- }
+ // Decode data with relevant codec
+ max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
- expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
- nbInt16 = expandedSize/sizeof(int16);
+ if (size > max) {
+ _debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
+ size = max;
+ }
- if (nbInt16 > max) {
- _debug("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.\n", nbInt16, max);
- nbInt16 = max;
- }
+ expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
+ nbInt16 = expandedSize/sizeof(int16);
- nbSample = nbInt16;
- // resample
- nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
+ if (nbInt16 > max) {
+ _debug("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.\n", nbInt16, max);
+ nbInt16 = max;
+ }
- audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+ nbSample = nbInt16;
+ // resample
+ nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
- } else {
- _debug("IAX: incoming audio, but no sound card open");
- }
+ audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+
+ } else {
+ _debug("IAX: incoming audio, but no sound card open");
+ }
}
/**
* Handle the registration process
*/
- void
+ void
IAXVoIPLink::iaxHandleRegReply(iax_event* event)
{
int new_voicemails;
std::string account_id;
-
+ IAXAccount *account;
+
+ account_id = getAccountID();
+ account = dynamic_cast<IAXAccount *>(Manager::instance().getAccount(account_id));
+
if (event->etype == IAX_EVENT_REGREJ) {
/* Authentication failed! */
_mutexIAX.enterMutex();
iax_destroy(_regSession);
_mutexIAX.leaveMutex();
_regSession = NULL;
- //TODO Restore that
- //setRegistrationState(ErrorAuth);
+ // Update the account registration state
+ account->setRegistrationState(ErrorAuth);
}
-
+
else if (event->etype == IAX_EVENT_REGACK) {
/* Authentication succeeded */
_mutexIAX.enterMutex();
// Looking for the voicemail information
//if( event->ies != 0 )
- new_voicemails = processIAXMsgCount(event->ies.msgcount);
- _debug("iax voicemail number notification: %i\n", new_voicemails);
+ //new_voicemails = processIAXMsgCount(event->ies.msgcount);
+ //_debug("iax voicemail number notification: %i\n", new_voicemails);
// Notify the client if new voicemail waiting for the current account
- account_id = getAccountID();
- Manager::instance().startVoiceMessageNotification(account_id.c_str(), new_voicemails);
+ //account_id = getAccountID();
+ //Manager::instance().startVoiceMessageNotification(account_id.c_str(), new_voicemails);
iax_destroy(_regSession);
_mutexIAX.leaveMutex();
@@ -789,13 +792,13 @@ IAXVoIPLink::iaxHandleRegReply(iax_event* event)
// I mean, save the timestamp, so that we re-register again in the REFRESH time.
// Defaults to 60, as per draft-guy-iax-03.
_nextRefreshStamp = time(NULL) + (event->ies.refresh ? event->ies.refresh : 60);
- //nsetRegistrationState(Registered);
+ account->setRegistrationState(Registered);
}
}
int IAXVoIPLink::processIAXMsgCount( int msgcount )
{
-
+
// IAX sends the message count under a specific format:
// 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
@@ -809,118 +812,108 @@ int IAXVoIPLink::processIAXMsgCount( int msgcount )
// Thus:
// 0 <= msgcount <= 255 => msgcount new messages, 0 old messages
// msgcount >= 256 => msgcount/256 old messages , msgcount%256 new messages (RULES)
-
+
return msgcount%256;
}
- void
+ void
IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
{
- IAXCall* call = NULL;
- CallID id;
- std::string reason = "Error ringing user.";
+ IAXCall* call = NULL;
+ CallID id;
+ std::string reason = "Error ringing user.";
- switch(event->etype) {
- case IAX_EVENT_REGACK:
- case IAX_EVENT_REGREJ:
- _debug("IAX Registration Event in a pre-call setup\n");
- break;
+ switch(event->etype) {
+ case IAX_EVENT_REGACK:
+ case IAX_EVENT_REGREJ:
+ _debug("IAX Registration Event in a pre-call setup\n");
+ break;
- case IAX_EVENT_REGREQ:
- // Received when someone wants to register to us!?!
- // Asterisk receives and answers to that, not us, we're a phone.
- _debug("Registration by a peer, don't allow it\n");
- break;
+ case IAX_EVENT_REGREQ:
+ // Received when someone wants to register to us!?!
+ // Asterisk receives and answers to that, not us, we're a phone.
+ _debug("Registration by a peer, don't allow it\n");
+ break;
- case IAX_EVENT_CONNECT:
- // We've got an incoming call! Yikes!
- _debug("> IAX_EVENT_CONNECT (receive)\n");
+ case IAX_EVENT_CONNECT:
+ // We've got an incoming call! Yikes!
+ _debug("> IAX_EVENT_CONNECT (receive)\n");
- id = Manager::instance().getNewCallID();
+ id = Manager::instance().getNewCallID();
+ call = new IAXCall(id, Call::Incoming);
- call = new IAXCall(id, Call::Incoming);
+ if (!call) {
+ _debug("! IAX Failure: unable to create an incoming call");
+ return;
+ }
- if (!call) {
- _debug("! IAX Failure: unable to create an incoming call");
- return;
- }
+ // Setup the new IAXCall
+ // Associate the call to the session.
+ call->setSession(event->session);
- // Setup the new IAXCall
- // Associate the call to the session.
- call->setSession(event->session);
+ // setCallAudioLocal(call);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ call->setConnectionState(Call::Progressing);
- // setCallAudioLocal(call);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- call->setConnectionState(Call::Progressing);
+ if (event->ies.calling_number)
+ call->setPeerNumber(std::string(event->ies.calling_number));
+ if (event->ies.calling_name)
+ call->setPeerName(std::string(event->ies.calling_name));
- if (event->ies.calling_number)
- call->setPeerNumber(std::string(event->ies.calling_number));
- if (event->ies.calling_name)
- call->setPeerName(std::string(event->ies.calling_name));
+ if (Manager::instance().incomingCall(call, getAccountID())) {
+ /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
+ * l'établissement du codec de transmission */
- if (Manager::instance().incomingCall(call, getAccountID())) {
- /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
- * l'établissement du codec de transmission */
+ // Remote lists its capabilities
+ int format = call->getFirstMatchingFormat(event->ies.capability);
+ // Remote asks for preferred codec voiceformat
+ int pref_format = call->getFirstMatchingFormat(event->ies.format);
- // Remote lists its capabilities
- int format = call->getFirstMatchingFormat(event->ies.capability);
- // Remote asks for preferred codec voiceformat
- int pref_format = call->getFirstMatchingFormat(event->ies.format);
+ // Priority to remote's suggestion. In case it's a forwarding, no transcoding
+ // will be needed from the server, thus less latency.
+ if (pref_format)
+ format = pref_format;
- // Priority to remote's suggestion. In case it's a forwarding, no transcoding
- // will be needed from the server, thus less latency.
- if (pref_format)
- format = pref_format;
+ iax_accept(event->session, format);
+ iax_ring_announce(event->session);
- iax_accept(event->session, format);
- iax_ring_announce(event->session);
+ addCall(call);
+ } else {
+ // reject call, unable to add it
+ iax_reject(event->session, (char*)reason.c_str());
- addCall(call);
- } else {
- // reject call, unable to add it
- iax_reject(event->session, (char*)reason.c_str());
+ delete call; call = NULL;
+ }
- delete call; call = NULL;
- }
+ break;
- break;
+ case IAX_EVENT_HANGUP:
+ // Remote peer hung up
+ call = iaxFindCallBySession(event->session);
+ id = call->getCallId();
- case IAX_EVENT_HANGUP:
- // Remote peer hung up
- call = iaxFindCallBySession(event->session);
- id = call->getCallId();
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ break;
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- break;
+ case IAX_EVENT_TIMEOUT: // timeout for an unknown session
- case IAX_EVENT_TIMEOUT: // timeout for an unknown session
+ break;
- break;
+ case IAX_IE_MSGCOUNT:
+ //_debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
+ break;
- case IAX_IE_MSGCOUNT:
- //_debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
- break;
-
- default:
- _debug("Unknown event type (in precall): %d\n", event->etype);
- }
-
-}
+ default:
+ _debug("Unknown event type (in precall): %d\n", event->etype);
+ }
- int
-IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
-{
- CodecOrder map = call->getCodecMap().getActiveCodecs();
- printf("taytciatcia = %i\n", map.size());
- return 0;
}
-
void IAXVoIPLink::updateAudiolayer( void )
{
_mutexIAX.enterMutex();
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 8ad4082882f41b731a610714172167bb94e15f95..b82949f640760b474913204c596616018ece1c98 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -166,21 +166,7 @@ class IAXVoIPLink : public VoIPLink
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; }
-
+
void updateAudiolayer( void );
void setStunServer( const std::string &server ) {};
@@ -194,7 +180,7 @@ class IAXVoIPLink : public VoIPLink
* @param msgcount The value sent by IAX in the REGACK message
* @return int The number of new messages waiting for the current registered user
*/
- int processIAXMsgCount( int msgcount );
+ int processIAXMsgCount( int msgcount );
/**
@@ -253,28 +239,12 @@ class IAXVoIPLink : public VoIPLink
*/
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;
-
/** 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. */
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index c8c25d2b8432821c06f587c7b17ae0d3b38fa651..0b3b46b719f59d2949c91a4ed2a91116cae15d5f 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -57,20 +57,20 @@ ManagerImpl::ManagerImpl (void)
: _hasTriedToRegister(false)
, _config()
, _currentCallId2()
- //, _currentCallMutex()
+ , _currentCallMutex()
, _codecBuilder(NULL)
, _audiodriver(NULL)
, _dtmfKey(NULL)
, _codecDescriptorMap()
- //, _toneMutex()
+ , _toneMutex()
, _telephoneTone(NULL)
, _audiofile()
, _spkr_volume(0)
, _mic_volume(0)
- //, _mutex()
+ , _mutex()
, _dbus(NULL)
, _waitingCall()
- //, _waitingCallMutex()
+ , _waitingCallMutex()
, _nbIncomingWaitingCall(0)
, _path("")
, _exist(0)
@@ -79,7 +79,7 @@ ManagerImpl::ManagerImpl (void)
, _firewallAddr("")
, _hasZeroconf(false)
, _callAccountMap()
- //, _callAccountMapMutex()
+ , _callAccountMapMutex()
, _accountMap()
{
@@ -158,13 +158,11 @@ void ManagerImpl::terminate()
bool
ManagerImpl::isCurrentCall(const CallID& callId) {
- //ost::MutexLock m(_currentCallMutex);
return (_currentCallId2 == callId ? true : false);
}
bool
ManagerImpl::hasCurrentCall() {
- //ost::MutexLock m(_currentCallMutex);
_debug("Current call ID = %s\n", _currentCallId2.c_str());
if ( _currentCallId2 != "") {
return true;
@@ -174,13 +172,12 @@ ManagerImpl::hasCurrentCall() {
const CallID&
ManagerImpl::getCurrentCallId() {
- //ost::MutexLock m(_currentCallMutex);
return _currentCallId2;
}
void
ManagerImpl::switchCall(const CallID& id ) {
- //ost::MutexLock m(_currentCallMutex);
+ ost::MutexLock m(_currentCallMutex);
_currentCallId2 = id;
}
@@ -403,7 +400,6 @@ ManagerImpl::saveConfig (void)
ManagerImpl::initRegisterAccounts()
{
int status;
- //TODO What the flag is for ??
bool flag = true;
AccountMap::iterator iter;
@@ -527,20 +523,19 @@ ManagerImpl::playDtmf(char code, bool isTalking)
// Multi-thread
bool
ManagerImpl::incomingCallWaiting() {
- //ost::MutexLock m(_waitingCallMutex);
return (_nbIncomingWaitingCall > 0) ? true : false;
}
void
ManagerImpl::addWaitingCall(const CallID& id) {
- //ost::MutexLock m(_waitingCallMutex);
+ ost::MutexLock m(_waitingCallMutex);
_waitingCall.insert(id);
_nbIncomingWaitingCall++;
}
void
ManagerImpl::removeWaitingCall(const CallID& id) {
- //ost::MutexLock m(_waitingCallMutex);
+ ost::MutexLock m(_waitingCallMutex);
// should return more than 1 if it erase a call
if (_waitingCall.erase(id)) {
_nbIncomingWaitingCall--;
@@ -549,7 +544,6 @@ ManagerImpl::removeWaitingCall(const CallID& id) {
bool
ManagerImpl::isWaitingCall(const CallID& id) {
- //ost::MutexLock m(_waitingCallMutex);
CallIDSet::iterator iter = _waitingCall.find(id);
if (iter != _waitingCall.end()) {
return false;
@@ -701,9 +695,9 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
if (!hasToPlayTone) return false;
if (_telephoneTone != 0) {
- //_toneMutex.enterMutex();
+ _toneMutex.enterMutex();
_telephoneTone->setCurrentTone(toneId);
- //_toneMutex.leaveMutex();
+ _toneMutex.leaveMutex();
AudioLoop* audioloop = getTelephoneTone();
unsigned int nbSampling = audioloop->getSize();
@@ -740,16 +734,16 @@ ManagerImpl::stopTone(bool stopAudio=true) {
}
- //_toneMutex.enterMutex();
+ _toneMutex.enterMutex();
if (_telephoneTone != 0) {
_telephoneTone->setCurrentTone(Tone::TONE_NULL);
}
- //_toneMutex.leaveMutex();
+ _toneMutex.leaveMutex();
// for ringing tone..
- //_toneMutex.enterMutex();
+ _toneMutex.enterMutex();
_audiofile.stop();
- //_toneMutex.leaveMutex();
+ _toneMutex.leaveMutex();
}
/**
@@ -786,7 +780,6 @@ ManagerImpl::congestion () {
void
ManagerImpl::ringback () {
playATone(Tone::TONE_RINGTONE);
- getAudioDriver()->trigger_thread();
}
/**
@@ -813,13 +806,13 @@ ManagerImpl::ringtone()
int sampleRate = audiolayer->getSampleRate();
AudioCodec* codecForTone = _codecDescriptorMap.getFirstCodecAvailable();
- //_toneMutex.enterMutex();
+ _toneMutex.enterMutex();
bool loadFile = _audiofile.loadFile(ringchoice, codecForTone , sampleRate);
- //_toneMutex.leaveMutex();
+ _toneMutex.leaveMutex();
if (loadFile) {
- //_toneMutex.enterMutex();
+ _toneMutex.enterMutex();
_audiofile.start();
- //_toneMutex.leaveMutex();
+ _toneMutex.leaveMutex();
if(CHECK_INTERFACE( layer, ALSA )){
/*int size = _audiofile.getSize();
SFLDataFormat output[ size ];
@@ -846,7 +839,7 @@ ManagerImpl::ringtone()
ManagerImpl::getTelephoneTone()
{
if(_telephoneTone != 0) {
- //ost::MutexLock m(_toneMutex);
+ ost::MutexLock m(_toneMutex);
return _telephoneTone->getCurrentTone();
}
else {
@@ -857,7 +850,7 @@ ManagerImpl::getTelephoneTone()
AudioLoop*
ManagerImpl::getTelephoneFile()
{
- //ost::MutexLock m(_toneMutex);
+ ost::MutexLock m(_toneMutex);
if(_audiofile.isStarted()) {
return &_audiofile;
} else {
@@ -1735,7 +1728,7 @@ int ManagerImpl::getSipPort( void )
ManagerImpl::getCallStatus(const std::string& sequenceId UNUSED)
{
if (!_dbus) { return false; }
- //ost::MutexLock m(_callAccountMapMutex);
+ ost::MutexLock m(_callAccountMapMutex);
CallAccountMap::iterator iter = _callAccountMap.begin();
TokenList tk;
std::string code;
@@ -2032,7 +2025,7 @@ ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accou
{
if (getAccountFromCall(callID) == AccountNULL) { // nothing with the same ID
if ( accountExists(accountID) ) { // account id exist in AccountMap
- //ost::MutexLock m(_callAccountMapMutex);
+ ost::MutexLock m(_callAccountMapMutex);
_callAccountMap[callID] = accountID;
_debug("Associate Call %s with Account %s\n", callID.data(), accountID.data());
return true;
@@ -2047,7 +2040,7 @@ ManagerImpl::associateCallToAccount(const CallID& callID, const AccountID& accou
AccountID
ManagerImpl::getAccountFromCall(const CallID& callID)
{
- //ost::MutexLock m(_callAccountMapMutex);
+ ost::MutexLock m(_callAccountMapMutex);
CallAccountMap::iterator iter = _callAccountMap.find(callID);
if ( iter == _callAccountMap.end()) {
return AccountNULL;
@@ -2059,7 +2052,7 @@ ManagerImpl::getAccountFromCall(const CallID& callID)
bool
ManagerImpl::removeCallAccount(const CallID& callID)
{
- //ost::MutexLock m(_callAccountMapMutex);
+ ost::MutexLock m(_callAccountMapMutex);
if ( _callAccountMap.erase(callID) ) {
return true;
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index 222c7154788fd86d4cc79d2fd3ad16c048584f82..cc3411f3ade26470030f558b6cba32e18795bfdd 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -865,7 +865,7 @@ class ManagerImpl {
CallID _currentCallId2;
/** Protected current call access */
- //ost::Mutex _currentCallMutex;
+ ost::Mutex _currentCallMutex;
/** Vector of CodecDescriptor */
CodecDescriptor* _codecBuilder;
@@ -883,7 +883,7 @@ class ManagerImpl {
/////////////////////
// Protected by Mutex
/////////////////////
- //ost::Mutex _toneMutex;
+ ost::Mutex _toneMutex;
TelephoneTone* _telephoneTone;
AudioFile _audiofile;
@@ -896,7 +896,7 @@ class ManagerImpl {
// Multithread variable (protected by _mutex)
//
/** Mutex to protect access to code section */
- //ost::Mutex _mutex;
+ ost::Mutex _mutex;
// Multithread variable (non protected)
DBusManagerImpl * _dbus;
@@ -905,7 +905,7 @@ class ManagerImpl {
CallIDSet _waitingCall;
/** Protect waiting call list, access by many voip/audio threads */
- //ost::Mutex _waitingCallMutex;
+ ost::Mutex _waitingCallMutex;
/** Number of waiting call, synchronize with waitingcall callidvector */
unsigned int _nbIncomingWaitingCall;
@@ -953,7 +953,7 @@ class ManagerImpl {
CallAccountMap _callAccountMap;
/** Mutex to lock the call account map (main thread + voiplink thread) */
- //ost::Mutex _callAccountMapMutex;
+ ost::Mutex _callAccountMapMutex;
/** Associate a new CallID to a AccountID
* Protected by mutex
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index 5880f927f00fd0708ed0a7f30fedea5ef140af24..61c8bfd05b2a762891c61ebd8fee8f22fd3f2dda 100644
--- a/src/sipaccount.cpp
+++ b/src/sipaccount.cpp
@@ -52,19 +52,13 @@ SIPAccount::~SIPAccount()
int SIPAccount::registerVoIPLink()
{
- int status, useStun;
- SIPVoIPLink *thislink;
+ int status;
/* Retrieve the account information */
/* Stuff needed for SIP registration */
setHostname(Manager::instance().getConfigString(_accountID,HOSTNAME));
setUsername(Manager::instance().getConfigString(_accountID, USERNAME));
setPassword(Manager::instance().getConfigString(_accountID, PASSWORD));
- /* Retrieve STUN stuff */
- /* STUN configuration is attached to a voiplink because it is applied to every accounts (PJSIP limitation)*/
- thislink = dynamic_cast<SIPVoIPLink*> (_link);
- if (thislink) {
- }
/* Start registration */
status = _link->sendRegister( _accountID );
diff --git a/src/sipaccount.h b/src/sipaccount.h
index b716086fbe5fc94fe57146228d20d4605a92169d..a49af2d00864a29c674a2ac152dcd2663d75620c 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -77,22 +77,10 @@ class SIPAccount : public Account
bool fullMatch(const std::string& username, const std::string& hostname);
bool userMatch(const std::string& username);
- inline std::string getHostname( void ) { return _hostname; }
- inline void setHostname( std::string hostname) { _hostname = hostname; }
-
- inline std::string getPassword( void ) { return _password; }
- inline void setPassword( std::string password ) { _password = password; }
-
- inline std::string getAlias( void ) { return _alias; }
- inline void setAlias( std::string alias ) { _alias = alias; }
-
- inline std::string getType( void ) { return _type; }
- inline void setType( std::string type ) { _type = type; }
-
pjsip_regc* getRegistrationInfo( void ) { return _regc; }
void setRegistrationInfo( pjsip_regc *regc ) { _regc = regc; }
- //TODO See if it useful
+ /* Registration flag */
bool isRegister() {return _bRegister;}
void setRegister(bool result) {_bRegister = result;}
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index a37e68318704433019f1836d553157ab4f0910c0..e3fc11b7920e38dd1e813a486f4fdae20ac110f4 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -211,13 +211,13 @@ SIPVoIPLink::terminate()
SIPVoIPLink::terminateSIPCall()
{
- //ost::MutexLock m(_callMapMutex);
+ ost::MutexLock m(_callMapMutex);
CallMap::iterator iter = _callMap.begin();
SIPCall *call;
while( iter != _callMap.end() ) {
call = dynamic_cast<SIPCall*>(iter->second);
if (call) {
- //TODO terminate the sip call
+ // terminate the sip call
delete call; call = 0;
}
iter++;
@@ -517,17 +517,17 @@ SIPVoIPLink::onhold(const CallID& id)
if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
- _mutexSIP.enterMutex();
// Stop sound
call->setAudioStart(false);
call->setState(Call::Hold);
_debug("* SIP Info: Stopping AudioRTP for onhold action\n");
- _audiortp->closeRtpSession();
+ //_mutexSIP.enterMutex();
+ _audiortp->closeRtpSession();
+ //_mutexSIP.leaveMutex();
+
local_sdp = call->getLocalSDPSession();
- _mutexSIP.leaveMutex();
-
if( local_sdp == NULL ){
_debug("! SIP Failure: unable to find local_sdp\n");
return false;
@@ -1145,7 +1145,6 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
// Init the callback for INVITE session:
- // TODO The invite session as global ?
pj_bzero(&inv_cb, sizeof (inv_cb));
inv_cb.on_state_changed = &call_on_state_changed;
@@ -1171,7 +1170,7 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
_debug("UserAgent: pjsip version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
- //TODO Create the secondary thread to poll sip events
+ // Create the secondary thread to poll sip events
_evThread->start();
/* Done! */
@@ -1435,9 +1434,6 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
SIPVoIPLink *link;
pjsip_msg *msg;
- _debug("UserAgent: TSX Changed! The tsx->state is %d; tsx->role is %d; code is %d; method id is %.*s.\n",
- tsx->state, tsx->role, tsx->status_code, (int)tsx->method.name.slen, tsx->method.name.ptr);
-
if(pj_strcmp2(&tsx->method.name, "INFO") == 0) {
// Receive a INFO message, ingore it!
return;
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 0aca792fc392fd6ab2a84bea7760c5d6a7870d01..03b12e6c38ba2d0dc8eae5dd73f3a9a9916f5720 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -42,7 +42,7 @@ class AudioRtp;
#define RANDOM_SIP_PORT rand() % 64000 + 1024
// To set the verbosity. From 0 (min) to 6 (max)
-#define PJ_LOG_LEVEL 5
+#define PJ_LOG_LEVEL 1
/**
* @file sipvoiplink.h
diff --git a/src/useragent.h b/src/useragent.h
deleted file mode 100644
index 728b074cd70eb9fb14d09256cfabeaa36cd653c0..0000000000000000000000000000000000000000
--- a/src/useragent.h
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yun Liu <yun.liu@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-
-#ifndef _SIPMANAGER_H
-#define _SIPMANAGER_H
-
-#include <pjsip.h>
-#include <pjlib-util.h>
-#include <pjlib.h>
-#include <pjnath/stun_config.h>
-
-//TODO Remove this include if we don't need anything from it
-#include <pjsip_simple.h>
-
-#include <pjsip_ua.h>
-#include <pjmedia/sdp.h>
-#include <pjmedia/sdp_neg.h>
-
-#include <string>
-#include <vector>
-
-#define PJ_LOG_LEVEL 1
-
-typedef std::string AccountID;
-
-class SIPCall;
-
-class UserAgent
-{
-private:
- /** PJSIP Endpoint */
- pjsip_endpoint *_endpt;
- pj_sock_t _sock;
- //pjsip_module _appMod;
- pj_caching_pool _cp;
- pj_pool_t *_pool;
- pj_mutex_t *_mutex; /** Mutex protection for this data */
- pjsip_module _mod; /** PJSIP module. */
- pjsip_module _options_handler;
- bool _useStun;
- pj_str_t _stunHost;
-
-
- pj_thread_t *_thread;
-
- static UserAgent *_current;
-
- /* Sleep with polling */
- void busy_sleep(unsigned msec);
-public:
- UserAgent();
- ~UserAgent();
-
- pj_status_t sipCreate();
-
- /**
- * This method is used to initialize the pjsip
- */
- pj_status_t sipInit();
-
-
- void sipDestory();
-
- /** Set whether it will use stun server */
- void setStunServer(const char *server);
-
- /** Set the port number user designated */
- void setSipPort(int port) { _regPort = port; }
-
- int getSipPort() { return _regPort; }
-
- pj_str_t getStunServer() { return _stunHost; }
-
- bool addAccount(AccountID id, pjsip_regc **regc, const std::string& server, const std::string& user, const std::string& passwd
- , const int& timeout);
- bool removeAccount(pjsip_regc *regc);
-
- pj_str_t buildContact(char *userName);
-
- bool loadSIPLocalIP();
-
-
- pjsip_endpoint* getEndPoint() {return _endpt;}
-
- std::string getLocalIP() {return _localExternAddress;}
-
- int getModId() {return _mod.id;}
-
- bool setCallAudioLocal(SIPCall* call);
-
- int answer(SIPCall* call);
-
- bool hangup(SIPCall* call);
-
- bool refuse(SIPCall* call);
-
- bool onhold(SIPCall *call);
- bool offhold(SIPCall *call);
-
- bool transfer(SIPCall *call, const std::string& to);
-
- void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata);
-
- bool makeOutgoingCall(const std::string& to, SIPCall* call, const AccountID& id);
-
- bool carryingDTMFdigits(SIPCall* call, char *msgBody);
-
- pj_pool_t *getAppPool() {return _pool;}
- static pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata);
- static pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata UNUSED) {return PJ_SUCCESS;}
- static void regc_cb(struct pjsip_regc_cbparam *param);
- static void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event);
- static void xfer_svr_cb(pjsip_evsub *sub, pjsip_event *event);
- static void call_on_media_update( pjsip_inv_session *inv UNUSED, pj_status_t status UNUSED) {}
- static void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e);
- static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
- static void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
- static int start_thread(void *arg);
- static UserAgent* getInstance() {return _current;}
-
- static void set_voicemail_info( AccountID account, pjsip_msg_body *body );
-private:
-
- // Copy Constructor
- UserAgent(const UserAgent& rh);
-
- // Assignment Operator
- UserAgent& operator=( const UserAgent& rh);
-};
-
-
-#endif /* _SIPMANAGER_H */
-
diff --git a/src/voiplink.cpp b/src/voiplink.cpp
index e4ff653d48211f0912f639bc8da37a75ada65593..77a0c95ed11f7cc05646e0baa21a311501310ab8 100644
--- a/src/voiplink.cpp
+++ b/src/voiplink.cpp
@@ -76,3 +76,11 @@ bool VoIPLink::clearCallMap()
return true;
}
+Account* VoIPLink::getAccountPtr(void)
+{
+ Account* account;
+ AccountID id;
+
+ id = getAccountID();
+ return Manager::instance().getAccount(id);
+}
diff --git a/src/voiplink.h b/src/voiplink.h
index 45f0c0e34e4f25fdd02e6574966f7fce637d6f99..4038394ccf6446663bba7e5d0d465d7cd955c2a8 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -27,6 +27,7 @@
#include "account.h"
class AudioCodec;
+class Account;
/** Define AccountID type */
typedef std::string AccountID;
@@ -182,6 +183,8 @@ class VoIPLink {
*/
inline AccountID& getAccountID(void) { return _accountID; }
+ Account* getAccountPtr(void);
+
/**
* @param accountID The account identifier
*/
diff --git a/src/zeroconf/DNSQueryThread.cpp b/src/zeroconf/DNSQueryThread.cpp
deleted file mode 100644
index a5e9dd3fd2208eb5466264ce7d5c078961a9eedc..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSQueryThread.cpp
+++ /dev/null
@@ -1,69 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#include "DNSQueryThread.h"
-#include "DNSService.h"
-
-/**
- * Construct a DNSQueryThread and initialize the cancel to deferred
- */
-DNSQueryThread::DNSQueryThread(DNSService *parent, const char *regtype) : ost::Thread()
-{
- _parent = parent;
- _regtype = regtype;
- _serviceRef = NULL;
- setCancel(cancelDeferred);
-}
-
-/**
- * Destruct a DNSQueryThread
- */
-DNSQueryThread::~DNSQueryThread()
-{
- if (_serviceRef) {
- DNSServiceRefDeallocate(_serviceRef);
- }
- terminate();
- _parent = NULL;
- _regtype = NULL;
- _serviceRef = NULL;
-}
-
-/**
- * Running loop
- */
-void
-DNSQueryThread::run() {
- DNSServiceErrorType theErr=0; // NULL;
- DNSServiceFlags resultFlags=0;
-
- theErr = DNSServiceBrowse(&_serviceRef,
- resultFlags,
- 0, // all interfaces
- _regtype,
- NULL,
- DNSServiceAddServicesCallback,
- (void*)_parent);
-
- if (theErr == kDNSServiceErr_NoError) {
- while(!testCancel()) {
- DNSServiceProcessResult(_serviceRef); // blockage if none...
- }
- }
-}
diff --git a/src/zeroconf/DNSQueryThread.h b/src/zeroconf/DNSQueryThread.h
deleted file mode 100644
index 2ea0bc2cab61d49f1493cccee103e8e8fddd9d33..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSQueryThread.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#ifndef __DNSSD_DNSQUERYTHREAD_H__
-#define __DNSSD_DNSQUERYTHREAD_H__
-
-#include <cc++/thread.h>
-#include <dns_sd.h>
-
-class DNSService;
-class DNSQueryThread : public ost::Thread
-{
-public:
- DNSQueryThread(DNSService *parent, const char *regtype);
- ~DNSQueryThread();
- virtual void run(); // looking for services
-
-private:
- DNSService *_parent; // parent service
- DNSServiceRef _serviceRef; // service reference
- const char *_regtype; // service type and socket type (_sip._udp by example)
-};
-
-
-#endif // __DNSSD_DNSQUERYTHREAD_H__
diff --git a/src/zeroconf/DNSService.cpp b/src/zeroconf/DNSService.cpp
deleted file mode 100644
index e2919fe27d93ec7c431ec7e30cb53d1469b462f2..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSService.cpp
+++ /dev/null
@@ -1,237 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
- /**
- * Inspired by http://braden.machacking.net/zerobrowse.cpp and
- * http://developer.kde.org/documentation/library/3.4-api/dnssd/html/remoteservice_8cpp-source.html
- */
-#include "DNSService.h"
-#include "DNSServiceTXTRecord.h"
-#include "DNSQueryThread.h"
-
-#include <cc++/thread.h>
-
-/**
- * Simple Empty Constructor
- */
-DNSService::DNSService()
-{
- _start = false;
- _regtypeList.push_back("_sip._udp");
-#ifdef USE_IAX2
- _regtypeList.push_back("_iax._udp");
-#endif
-
- // for the thread, the ifdef add a dynamic _regtypeList problem
- for (std::list<std::string>::iterator iterThread=_regtypeList.begin();
- iterThread!=_regtypeList.end();
- iterThread++) {
- _queryThread.push_back(new DNSQueryThread(this, (*iterThread).c_str()));
- }
-}
-
-/**
- * Simple Empty Destructor
- */
-DNSService::~DNSService()
-{
- int cntThread = _queryThread.size();
- for (int iThread=0;iThread<cntThread;iThread++) {
- delete _queryThread[iThread]; _queryThread[iThread] = NULL;
- }
-}
-
-/**
- * Look for zeroconf services and add them to _services
- */
-void
-DNSService::startScanServices()
-{
- for (std::vector<DNSQueryThread *>::iterator iter = _queryThread.begin();iter!=_queryThread.end();iter++) {
- (*iter)->start();
- }
- _start = true;
-}
-
-/**
- * Add one service to the list of actual services
- * @param service Service to add to the list
- */
-void DNSService::addService(const std::string &service)
-{
- DNSServiceTXTRecord txtRecord;
- _mutex.enterMutex();
- _services[service] = txtRecord;
- // we leave before the queryService since, each
- // thread will modify a DNSServiceTXTRecord of a difference services
- _mutex.leaveMutex();
- notify();
- queryService(service);
-}
-
-/**
- * Remove one service to the list of actual services
- * @param service Service to remove to the list
- */
-void DNSService::removeService(const std::string &service)
-{
- _mutex.enterMutex();
- _services.erase(service);
- _mutex.leaveMutex();
- notify();
-}
-
-/**
- * Return every services
- */
-DNSServiceMap
-DNSService::getServices()
-{
- ost::MutexLock m(_mutex);
- return _services;
-}
-
-
-/**
- * Query a service and wait for the anwser
- * the queryCallback will show the result
- * @param service The service full adress
- */
-void
-DNSService::queryService(const std::string &service)
-{
- DNSServiceErrorType theErr=0;
- DNSServiceRef myServRef=0;
- DNSServiceFlags resultFlags=0;
-
- theErr = DNSServiceQueryRecord(&myServRef,
- resultFlags,
- 0,
- service.c_str(),
- kDNSServiceType_TXT,
- kDNSServiceClass_IN,
- DNSServiceQueryRecordCallback,
- (void*)this);
- if (theErr == kDNSServiceErr_NoError) {
- DNSServiceProcessResult(myServRef); // blockage...
- DNSServiceRefDeallocate(myServRef);
- }
-}
-
-/**
- * Overloadding queryService
- * @param service service name
- * @param regtype registred type of service
- * @param domain domain (habitually local.)
- */
-void
-DNSService::queryService(const char *service, const char *regtype, const char *domain)
-{
- char serviceName[kDNSServiceMaxDomainName+1];
- DNSServiceConstructFullName(serviceName, service, regtype, domain);
- queryService(std::string(serviceName));
-}
-
-/**
- * Add a txt record with the queryService callback answser data
- * @param rdlen the length of the txt record data
- * @param rdata txt record data
- */
-void
-DNSService::addTXTRecord(const char *fullname, uint16_t rdlen, const void *rdata)
-{
- char key[256];
-
- const char *value;
- uint8_t valueLen; // 0 to 256 by type restriction
- char valueTab[256];
-
-
- uint16_t keyCount = TXTRecordGetCount(rdlen, rdata);
- for (int iKey=0; iKey<keyCount; iKey++) {
- TXTRecordGetItemAtIndex (rdlen, rdata, iKey, 256, key, &valueLen, (const void **)(&value));
- if (value) {
- bcopy(value, valueTab, valueLen);
- valueTab[valueLen]='\0';
- _mutex.enterMutex(); // extra-careful
- _services[fullname].addKeyValue(key, valueTab);
- _mutex.leaveMutex();
- } else {
- _mutex.enterMutex();
- _services[fullname].removeKey(key);
- _mutex.leaveMutex();
- }
- }
-
- notify();
-}
-
-void
-DNSServiceAddServicesCallback(DNSServiceRef,
- DNSServiceFlags flags,
- uint32_t,
- DNSServiceErrorType errorCode,
- const char *serviceName,
- const char *replyType,
- const char *replyDomain,
- void *context)
-{
- if (errorCode==kDNSServiceErr_NoError) {
-
- if (flags) {
- DNSService *service = (DNSService*)context;
- std::string tempService;
- tempService = std::string(serviceName) + "." + std::string(replyType) + std::string(replyDomain);
- if (flags&kDNSServiceFlagsAdd) {
-// _debug("DNSServiceAddServicesCallback call addService\n");
- service->addService(tempService);
- } else {
-// _debug("DNSServiceAddServicesCallback call removeService\n");
- service->removeService(tempService);
- }
- }
- } else {
- // TODO: error handling
- }
-}
-
-void
-DNSServiceQueryRecordCallback(
- DNSServiceRef,
- DNSServiceFlags flags,
- uint32_t,
- DNSServiceErrorType errorCode,
- const char *fullname,
- uint16_t,
- uint16_t,
- uint16_t rdlen,
- const void *rdata,
- uint32_t,
- void *context)
-{
- if (errorCode==kDNSServiceErr_NoError) {
- if (flags&kDNSServiceFlagsAdd) {
-// _debug("DNSServiceQueryRecordCallback call addTXTRecord\n");
- ((DNSService *)context)->addTXTRecord(fullname, rdlen, rdata);
- } else {
-// _debug("DNSServiceQueryRecordCallback call removeService\n");
- ((DNSService *)context)->removeService(fullname);
- }
- }
-}
diff --git a/src/zeroconf/DNSService.h b/src/zeroconf/DNSService.h
deleted file mode 100644
index b1a840bf1cf6bf5b8076d9fb79c2785f84cb6ae4..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSService.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#ifndef __DNSSD_DNSSERVICE_H__
-#define __DNSSD_DNSSERVICE_H__
-
-#include <string>
-#include <map>
-#include <list>
-#include <vector>
-
-#include <dns_sd.h>
-#include <cc++/thread.h>
-#include "../observer.h"
-
-class DNSQueryThread;
-class DNSServiceTXTRecord;
-
-typedef std::map<std::string, DNSServiceTXTRecord> DNSServiceMap;
-class DNSService : public Pattern::Subject
-{
-public:
- DNSService();
- ~DNSService();
-
- void startScanServices(); // looking for services
- void addService(const std::string &service); // adding every services
- void removeService(const std::string &service); // remove a service
- DNSServiceMap getServices(); // get all DNS Service
- void stop(); // after the browsing loop stop
-
- void queryService(const std::string &service); // query the TXT record of a service
- void queryService(const char *service, const char *regtype, const char *domain);
- void addTXTRecord(const char *fullname, uint16_t rdlen, const void *rdata);
- //void removeTXTRecord(const char *fullname);
-
- bool isStart() const { return _start; }
-
-private:
- DNSServiceMap _services; //map
-
- std::vector<DNSQueryThread *> _queryThread;
- /**
- * Mutex to protect access to _services on add/erase
- */
- ost::Mutex _mutex;
- /**
- * RegType List contains zeroconf services to register, like sip, iax2, ...
- * It will be use to initialize the DNSQueryThread
- */
- std::list<std::string> _regtypeList;
-
- bool _start;
-};
-
-void DNSServiceAddServicesCallback(DNSServiceRef sdRef,
- DNSServiceFlags flags,
- uint32_t interfaceIndex,
- DNSServiceErrorType errorCode,
- const char *serviceName,
- const char *replyType,
- const char *replyDomain,
- void *context);
-
-void DNSServiceQueryRecordCallback(DNSServiceRef DNSServiceRef,
- DNSServiceFlags flags,
- uint32_t interfaceIndex,
- DNSServiceErrorType errorCode,
- const char *fullname,
- uint16_t rrtype,
- uint16_t rrclass,
- uint16_t rdlen,
- const void *rdata,
- uint32_t ttl,
- void *context);
-
-#endif // __DNSSD_DNSSERVICE_H__
diff --git a/src/zeroconf/DNSServiceTXTRecord.cpp b/src/zeroconf/DNSServiceTXTRecord.cpp
deleted file mode 100644
index f12ed15f5b2b1f9468ffa72f8b71d3567102f5a2..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSServiceTXTRecord.cpp
+++ /dev/null
@@ -1,76 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#include "DNSServiceTXTRecord.h"
-
-/**
- * Simple constructor
- */
-DNSServiceTXTRecord::DNSServiceTXTRecord()
-{
-}
-
-/**
- * Simple destructor
- */
-DNSServiceTXTRecord::~DNSServiceTXTRecord()
-{
-}
-
-/**
- * add a pair of key/value inside the associative std::map
- * @param key unique key inside the std::map
- * @param value value associated to the key
- */
-void
-DNSServiceTXTRecord::addKeyValue(const std::string &key, const std::string &value)
-{
- _map[key] = value;
-}
-
-/**
- * remove a key inside the map or do nothing if it doesn't exist
- * @param key unique key inside the std::map
- */
-void
-DNSServiceTXTRecord::removeKey(const std::string &key)
-{
- _map.erase(key);
-}
-
-/**
- * get a value from a key
- * @param key unique key inside the std::map
- * @return the value or empty
- */
-const std::string &
-DNSServiceTXTRecord::getValue(const std::string &key)
-{
- return _map[key]; // return std::string("") if it's not there
-}
-
-/**
- * get a value from a key
- * @param key unique key inside the std::map
- * @return the value or empty
- */
-const std::string &
-DNSServiceTXTRecord::getValue(const char* key)
-{
- return getValue(std::string(key));
-}
diff --git a/src/zeroconf/DNSServiceTXTRecord.h b/src/zeroconf/DNSServiceTXTRecord.h
deleted file mode 100644
index d6d2742fd6efa5091dc04f63064f3eb28e555397..0000000000000000000000000000000000000000
--- a/src/zeroconf/DNSServiceTXTRecord.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/**
- * Copyright (C) 2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-#ifndef __DNSSD_DNSSERVICETXTRECORD_H__
-#define __DNSSD_DNSSERVICETXTRECORD_H__
-
-#include <string>
-#include <map>
-
-typedef std::map<std::string, std::string> TXTRecordMap;
-class DNSServiceTXTRecord
-{
-public:
- DNSServiceTXTRecord();
- ~DNSServiceTXTRecord();
- void addKeyValue(const std::string &key, const std::string &value);
- void removeKey(const std::string &key);
- const std::string &getValue(const std::string &key);
- const std::string &getValue(const char *key);
- inline void clear(void) { _map.clear(); };
- inline int size(void) { return _map.size(); };
- TXTRecordMap getTXTRecords() { return _map; }
-
-private:
- TXTRecordMap _map;
-};
-
-#endif // __DNSSD_DNSSERVICETXTRECORD_H__
diff --git a/src/zeroconf/Makefile.am b/src/zeroconf/Makefile.am
deleted file mode 100644
index d2ed5b4e4a56d0330b7bad354ea7f2afec291086..0000000000000000000000000000000000000000
--- a/src/zeroconf/Makefile.am
+++ /dev/null
@@ -1,12 +0,0 @@
-SUBDIRS =
-
-noinst_LTLIBRARIES = libzeroconf.la
-
-libzeroconf_la_SOURCES = \
- DNSQueryThread.cpp DNSQueryThread.h \
- DNSService.cpp DNSService.h \
- DNSServiceTXTRecord.cpp DNSServiceTXTRecord.h
-
-AM_CXXFLAGS = $(libccext2_CFLAGS)
-libzeroconf_la_LIBADD = $(LIB_DNSSD)
-