diff --git a/configure.ac b/configure.ac
index a9acc371f41e90fc800882a31a6adffe2d00d6ee..d1cc4515db0fc34954388be6e980930c8ae9b979 100644
--- a/configure.ac
+++ b/configure.ac
@@ -15,7 +15,7 @@ LIBS="$LIBS -lstdc++"
dnl
dnl Solaris pkgadd support definitions
PKGADD_PKG="SFLPhoned"
-PKGADD_NAME="SFLPhone - a SIP client and deamon"
+PKGADD_NAME="SFLPhone - a SIP client and daemon"
PKGADD_VENDOR="http://www.sflphone.org/"
AC_SUBST(PKGADD_PKG)
AC_SUBST(PKGADD_NAME)
@@ -146,6 +146,7 @@ AM_CONDITIONAL(USE_ZEROCONF, test "$have_libdns_sd" = "yes")
dnl AC_CONFIG_FILES(
AC_OUTPUT(
sflphone.spec \
+sflphoned-fedora.spec \
Makefile \
src/Makefile \
src/audio/Makefile \
diff --git a/sflphone.spec.in b/sflphone.spec.in
index 80937125ecb6144ac3f157e6fbbb9ccf98f83586..ecdd21b047dd97b0340093a39e26f597ee4502cf 100644
--- a/sflphone.spec.in
+++ b/sflphone.spec.in
@@ -1,22 +1,22 @@
-%define name sflphone
+%define name sflphoned
%define version 0.5
%define release 1
%define prefix /usr
Autoreq: 0
Name: %name
-Summary: An other VoIP client
+Summary: A VoIP daemon with SIP protocol
Version: %version
Release: %release
License: GPL
Group: Networking/Utilities
URL: http://www.sflphone.org/
Packager: Cyrille Béraud <cyrille.beraud@savoirfairelinux.com>
-Source: http://www.sflphone.org/sflphone-0.5.tar.gz
+Source: http://www.sflphone.org/releases/sflphone-0.5.tar.gz
BuildRoot: /tmp/sflphone-%{version}-%{release}
%description
-SFLPhone is a VoIP client for a X11 environment.
+SFLPhoned is a VoIP daemon with SIP protocol.
%prep
%setup
CXXFLAGS="$RPM_OPT_FLAGS" ./configure --prefix %{prefix}
diff --git a/src/Makefile.am b/src/Makefile.am
index 9438d91446011fe17bb4e91b3beeb255f67a4aa1..8ef596118265105b0dea2fee604c81b60eb5a3a3 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -20,9 +20,9 @@ sflphoned_SOURCES = call.cpp eventthread.cpp error.cpp main.cpp sipvoiplink.
sflphoned_CXXFLAGS = -DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone\" $(ZEROCONFFLAGS)
-sflphoned_LDFLAGS = -static
+#sflphoned_LDFLAGS = -static
sflphoned_LDADD = gui/libguiframework.la audio/libaudio.la ../stund/libstun.la ../utilspp/libutilspp.la -lpthread $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) config/libconfig.la
-KDE_CXXFLAGS = $(USE_EXCEPTIONS)
+#KDE_CXXFLAGS = $(USE_EXCEPTIONS)
AM_CPPFLAGS = -I$(top_srcdir) -I$(srcdir)/audio/pacpp/include $(libccext2_CFLAGS) $(libccgnu2_CFLAGS) $(portaudio_CFLAGS)
noinst_HEADERS = observer.h
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 54d2f1b928a631cf6ac534f6095a769d4f29167d..82453849e3fdb19852ab3f96997ebfc0d5a443e7 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -59,10 +59,10 @@ AudioRtp::createNewSession (SipCall *ca) {
_symmetric);
// Start PortAudio
- Manager::instance().getAudioDriver()->micRingBuffer().flush();
- Manager::instance().getAudioDriver()->startStream();
+ //Manager::instance().getAudioDriver()->flushMic();
+ //Manager::instance().getAudioDriver()->startStream();
- _debug("AudioRtp::createNewSession: starting RTX thread\n");
+ //_debug("AudioRtp::createNewSession: starting RTX thread\n");
if (_RTXThread->start() != 0) {
return -1;
}
@@ -80,27 +80,27 @@ AudioRtp::closeRtpSession () {
////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
-AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioLayer* driver,
- bool sym) : _codecBuilder(0) {
- time = new ost::Time();
- _ca = sipcall;
- _sym = sym;
- _audioDevice = driver;
-
- // TODO: Change bind address according to user settings.
+AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioLayer* driver, bool sym) : _codecBuilder(0) {
+
+ time = new ost::Time();
+ _ca = sipcall;
+ _sym = sym;
+ _audioDevice = driver;
+
+ // TODO: Change bind address according to user settings.
std::string localipConfig = _ca->getLocalIp();
- ost::InetHostAddress local_ip(localipConfig.c_str());
+ ost::InetHostAddress local_ip(localipConfig.c_str());
- _debug("RTP: listening on IP %s local port : %d\n", localipConfig.c_str(), _ca->getLocalAudioPort());
- if (!_sym) {
- _sessionRecv = new ost::RTPSession (local_ip, _ca->getLocalAudioPort());
- _sessionSend = new ost::RTPSession (local_ip);
+ if (!_sym) {
+ _sessionRecv = new ost::RTPSession (local_ip, _ca->getLocalAudioPort());
+ _sessionSend = new ost::RTPSession (local_ip);
_session = NULL;
- } else {
- _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
+ } else {
+ _debug("Symmetric RTP Session on local: %s:%d\n", localipConfig.c_str(), _ca->getLocalAudioPort());
+ _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
_sessionRecv = NULL;
_sessionSend = NULL;
- }
+ }
}
AudioRtpRTX::~AudioRtpRTX () {
@@ -110,13 +110,14 @@ AudioRtpRTX::~AudioRtpRTX () {
_debug("AudioRtpRTX: try to terminate, but catch an exception...\n");
}
_ca = NULL;
- _debug("Thread: AudioRtpRTX stop session\n");
+
if (!_sym) {
delete _sessionRecv; _sessionRecv = NULL;
delete _sessionSend; _sessionSend = NULL;
} else {
delete _session; _session = NULL;
}
+
delete time; time = NULL;
}
@@ -128,8 +129,6 @@ AudioRtpRTX::initAudioRtpSession (void)
if (!remote_ip) {
_debug("RTP: Target IP address [%s] is not correct!\n", _ca->getRemoteSdpAudioIp());
return;
- } else {
- _debug("RTP: Sending to %s : %d\n", _ca->getRemoteSdpAudioIp(), _ca->getRemoteSdpAudioPort());
}
// Initialization
@@ -150,31 +149,34 @@ AudioRtpRTX::initAudioRtpSession (void)
_debug("RTX send: could not connect to port %d\n",
_ca->getRemoteSdpAudioPort());
return;
- } else {
- _debug("RTP(Send): Added destination %s : %d\n",
- remote_ip.getHostname(),
- (unsigned short) _ca->getRemoteSdpAudioPort());
}
+ _debug("RTP(Send): Added sessionSend destination %s:%d\n",
+ remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
//setPayloadFormat(StaticPayloadFormat(sptPCMU));
- _debug("Payload Format: %d\n", _ca->payload);
+ //_debug("Payload Format: %d\n", _ca->payload);
_sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
_sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
- setCancel(cancelImmediate);
- _sessionSend->setMark(true);
+ setCancel(cancelImmediate);
+ _sessionSend->setMark(true);
+
+ } else {
+
+ _debug("RTP(Send): Added session destination %s:%d\n",
+ remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
+
+ if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
+ return;
+ }
+
+ _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
+ setCancel(cancelImmediate);
+ }
- } else {
- if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteSdpAudioPort())) {
- return;
- } else {
- _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _ca->payload));
- setCancel(cancelImmediate);
- }
- }
Manager::instance().getAudioDriver()->flushMic();
Manager::instance().getAudioDriver()->flushMain();
- _debug("-----------------------\n");
+ _debug("== AudioRtpRTX::initAudioRtpSession end == \n");
}
void
@@ -306,7 +308,7 @@ AudioRtpRTX::run (void) {
audiolayer->urgentRingBuffer().flush();
// start running the packet queue scheduler.
- _debug("Thread: start session of AudioRtpRTX\n");
+ //_debug("Thread: start session of AudioRtpRTX\n");
if (!_sym) {
_sessionRecv->startRunning();
_sessionSend->startRunning();
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index 4864a7ea18cf767e5caed7166c68aac7461113aa..7852741e6f488c9285478df2b9074d08257d5ab8 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -18,7 +18,6 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include <iostream>
#include <fstream>
#include <math.h>
#include <stdlib.h>
@@ -331,7 +330,7 @@ ToneGenerator::toneHandle (unsigned int idr, const std::string& zone) {
}
buildTone(idz, idr, _buf);
tonethread = new ToneThread(_buf, totalbytes);
- _debug("Thread: start tonethread\n");
+ //_debug("Thread: start tonethread\n");
tonethread->start();
}
@@ -339,10 +338,10 @@ void
ToneGenerator::stopTone() {
_currentTone = ZT_TONE_NULL;
- _debug("Thread: delete tonethread\n");
+ //_debug("Thread: delete tonethread\n");
delete tonethread; tonethread = NULL;
// we end the last thread
- _debug("Thread: tonethread deleted\n");
+ //_debug("Thread: tonethread deleted\n");
}
/**
@@ -385,11 +384,11 @@ ToneGenerator::playRingtone (const char *fileName) {
// expandedsize is the number of bytes, not the number of int
expandedsize = _ulaw->codecDecode (_dst, (unsigned char *)_src, length);
- _debug("length (pre-ulaw) : %d\n", length);
- _debug("expandedsize (post-ulaw) : %d\n", expandedsize);
+ //_debug("length (pre-ulaw) : %d\n", length);
+ //_debug("expandedsize (post-ulaw) : %d\n", expandedsize);
if (tonethread == NULL) {
- _debug("Thread: start tonethread\n");
+ //_debug("Thread: start tonethread\n");
// send the number of int16, so device by two
tonethread = new ToneThread ((int16*)_dst, expandedsize>>1);
tonethread->start();
diff --git a/src/global.h b/src/global.h
index 57cd515b87b3fe1ed8964fdf0502643344e78759..fc299d56471bd5533b626e50ba77a556b1244c9a 100644
--- a/src/global.h
+++ b/src/global.h
@@ -27,7 +27,7 @@
typedef float float32;
typedef short int16;
-#define DEBUG
+//#define DEBUG
#ifdef DEBUG
#define _debug(...) fprintf(stderr, "[sflphoned] " __VA_ARGS__)
diff --git a/src/gui/server/tcpstreampool.cpp b/src/gui/server/tcpstreampool.cpp
index 30f4de63dd1db7f1f76d5a3693f7bcfabcc67150..e66ae2b4cb903859ffdff6d45b4a33124f56f01f 100644
--- a/src/gui/server/tcpstreampool.cpp
+++ b/src/gui/server/tcpstreampool.cpp
@@ -29,7 +29,7 @@ TCPStreamPool::run() {
while(!testCancel() && good()) {
while (isPending(ost::TCPSocket::pendingInput, 2LU)) {
std::getline(*this, input);
- _debug("TCPStreamPool getline %s\n", input.c_str());
+ //_debug("TCPStreamPool getline %s\n", input.c_str());
if (input != null && input[0]!=cr13) {
_inputPool.push(input);
}
@@ -37,7 +37,7 @@ TCPStreamPool::run() {
if (testCancel() || !good()) {break;}
}
if (_outputPool.pop(output, 2LU)) {
- _debug("TCPStreamPool send %s\n", output.c_str());
+ //_debug("TCPStreamPool send %s\n", output.c_str());
*this << output << std::endl;
}
}
diff --git a/src/main.cpp b/src/main.cpp
index ab113d05005c1de0f85ae08ecaa9ffa000c64da5..a0339fef235864b97f1ba2373426c7f612ec5c7e 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -31,7 +31,7 @@ main (int argc, char **argv) {
if (argc == 2 && strcmp(argv[1], "--help") == 0) {
- std::cout << PROGNAME << " Deamon " << SFLPHONED_VERSION << ", by Savoir-Faire Linux 2004-2005" << std::endl << std::endl;
+ std::cout << PROGNAME << " Daemon " << SFLPHONED_VERSION << ", by Savoir-Faire Linux 2004-2005" << std::endl << std::endl;
std::cout << "USAGE: sflphoned [--help]" << std::endl;
std::cout << "Parameters: " << std::endl;
std::cout << " --help for this message" << std::endl << std::endl;
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 49b7f354268d5293419a465a524bea1d210a08fc..249cd9aa80558bbacdc75fffffcccddf9eda80c4 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -613,7 +613,7 @@ SipVoIPLink::getEvent (void)
CALLID id = 0;
int returnValue = 0;
- //_debug("GetEvent : %d ", event->type);
+ _debug("GetEvent : %d ", event->type);
switch (event->type) {
// IP-Phone user receives a new call
case EXOSIP_CALL_INVITE: //
@@ -635,9 +635,7 @@ SipVoIPLink::getEvent (void)
// Generate id
id = Manager::instance().generateNewCallId();
Manager::instance().pushBackNewCall(id, Incoming);
- _debug("Incoming call with id %d [cid = %d, did = %d]\n",
- id, event->cid, event->did);
- _debug("Local audio port: %d\n", _localPort);
+ _debug("New INVITE Event: call with id %d [cid = %d, did = %d]\n",id, event->cid, event->did);
// Display the callerId-name
osip_from_t *from;
@@ -662,7 +660,7 @@ SipVoIPLink::getEvent (void)
urlUsername = url->username;
}
Manager::instance().callSetInfo(id, name, urlUsername);
- _debug("From: %s\n", name.c_str());
+ _debug("New INVITE Event: From: %s\n", name.c_str());
}
//Don't need this display text message now that we send the name
//inside the Manager to the gui
@@ -672,15 +670,31 @@ SipVoIPLink::getEvent (void)
// Associate an audio port with a call
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
+ _debug("New INVITE Event: we set the local audio to: %s:%d\n", getLocalIpAddress().c_str(), _localPort);
sipcall->newIncomingCall(event);
if (Manager::instance().incomingCall(id) < 0) {
- Manager::instance().displayErrorText(id, "Incoming call failed");
+ Manager::instance().displayErrorText(id, "New INVITE Event: Incoming call failed");
}
break;
case EXOSIP_CALL_REINVITE:
- eXosip_call_send_answer(event->tid, 403, NULL);
+ _debug("!!! EXOSIP_CALL_REINVITE: Should reinvite? !!!\n");
+ //eXosip_call_send_answer(event->tid, 403, NULL);
+ //488 as http://www.atosc.org/pipermail/public/osip/2005-June/005385.html
+ id = findCallId(event);
+ if (id != 0) {
+ sipcall = getSipCall(id);
+ if (sipcall != NULL) {
+ _debug("Call reinvite : [id = %d, cid = %d, did = %d], localport=%d\n", id, event->cid, event->did,sipcall->getLocalAudioPort());
+
+ _audiortp.closeRtpSession();
+ sipcall->newIncomingCall(event);
+ _audiortp.createNewSession(sipcall);
+ }
+ } else {
+ eXosip_call_send_answer(event->tid, 488, NULL);
+ }
break;
case EXOSIP_CALL_PROCEEDING: // 8
@@ -770,14 +784,14 @@ SipVoIPLink::getEvent (void)
// Handle 4XX errors
switch (event->response->status_code) {
case AUTH_REQUIRED:
- _debug("EXOSIP_CALL_REQUESTFAILURE :: AUTH_REQUIRED\n");
+ _debug("SIP Server ask required authentification: loging...\n");
setAuthentication();
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
break;
case UNAUTHORIZED:
- _debug("EXOSIP_CALL_REQUESTFAILURE :: UNAUTHORIZED\n");
+ _debug("Request is unauthorized. SIP Server ask authentification: loging...\n");
setAuthentication();
break;
@@ -852,7 +866,7 @@ SipVoIPLink::getEvent (void)
break;
}
}
-
+
// TODO: Que faire si rien trouve??
eXosip_lock();
if (k == _sipcallVector.size()) {
@@ -879,7 +893,7 @@ SipVoIPLink::getEvent (void)
// Get the message body
ii = osip_message_get_body(event->request, 0, &body);
if (ii != 0) {
- _debug("Cannot get body\n");
+ _debug("Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
returnValue = -1;
break;
}
@@ -1244,7 +1258,7 @@ SipVoIPLink::behindNat (void)
}
// Firewall address
- _debug("STUN server: %s\n", svr.data());
+ //_debug("STUN server: %s\n", svr.data());
Manager::instance().getStunInfo(stunSvrAddr);
return 1;
@@ -1354,11 +1368,11 @@ SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& t
if (!Manager::instance().useStun()) {
// Set random port for outgoing call if no firewall
setLocalPort(RANDOM_LOCAL_PORT);
- _debug("Local audio port: %d\n",_localPort);
+ _debug("SipVoIPLink::startCall: Local audio port: %d\n",_localPort);
} else {
// If use Stun server
if (behindNat() != 0) {
- _debug("sip invite: firewall port = %d\n",Manager::instance().getFirewallPort());
+ _debug("SipVoIPLink::startCall: sip invite: firewall port = %d\n",Manager::instance().getFirewallPort());
setLocalPort(Manager::instance().getFirewallPort());
} else {
return -1;