From 897f34d9ce4643a85ed8a112710e70cd94a75726 Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Tue, 1 Nov 2005 20:17:54 +0000
Subject: [PATCH] Only switch call when no one are selected
---
src/audio/audiortp.cpp | 13 ++++++++++---
src/managerimpl.cpp | 34 +++++++++++++++++++++++-----------
src/managerimpl.h | 2 ++
src/sipvoiplink.cpp | 20 ++++++++++++--------
4 files changed, 47 insertions(+), 22 deletions(-)
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 2b5ec76a83..39c24ab7db 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -38,16 +38,23 @@
// AudioRtp
////////////////////////////////////////////////////////////////////////////////
AudioRtp::AudioRtp () {
- _RTXThread = NULL;
+ _RTXThread = 0;
}
AudioRtp::~AudioRtp (void) {
- delete _RTXThread; _RTXThread = NULL;
+ delete _RTXThread; _RTXThread = 0;
}
int
AudioRtp::createNewSession (SipCall *ca) {
ost::MutexLock m(_threadMutex);
+
+ // something should stop the thread before...
+ if ( _RTXThread != 0 ) {
+ _debug("AudioRTP error: try to create a new audio rtp thread...\n");
+ return -1;
+ }
+
// Start RTP Send/Receive threads
_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
_RTXThread = new AudioRtpRTX (ca, Manager::instance().getAudioDriver(), _symmetric);
@@ -70,7 +77,7 @@ AudioRtp::closeRtpSession () {
// This will make RTP threads finish.
_debug("waiting start signal...\n");
_debug("receive start signal...");
- delete _RTXThread; _RTXThread = NULL;
+ delete _RTXThread; _RTXThread = 0;
}
////////////////////////////////////////////////////////////////////////////////
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index c8cee61164..b8aaa4bc01 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -209,6 +209,7 @@ ManagerImpl::pushBackNewCall (CALLID id, enum CallType type)
Call*
ManagerImpl::getCall (CALLID id)
{
+ _debug("CALL: Getting call %d\n", id);
Call* call = NULL;
unsigned int size = _callVector.size();
for (unsigned int i = 0; i < size; i++) {
@@ -228,6 +229,7 @@ ManagerImpl::getCall (CALLID id)
void
ManagerImpl::deleteCall (CALLID id)
{
+ _debug("CALL: Deleting call %d\n", id);
CallVector::iterator iter = _callVector.begin();
while(iter!=_callVector.end()) {
Call *call = *iter;
@@ -244,6 +246,13 @@ ManagerImpl::deleteCall (CALLID id)
}
}
+void
+ManagerImpl::setCurrentCallId(CALLID id)
+{
+ _debug("CALL: Setting current callid %d to %d\n", _currentCallId, id);
+ _currentCallId = id;
+}
+
///////////////////////////////////////////////////////////////////////////////
// Management of events' IP-phone user
///////////////////////////////////////////////////////////////////////////////
@@ -255,7 +264,6 @@ ManagerImpl::outgoingCall (const std::string& to)
{
CALLID id = generateNewCallId();
Call *call = pushBackNewCall(id, Outgoing);
- _debug("Outgoing Call with identifiant %d\n", id);
ost::MutexLock m(_mutex);
call->setState(Call::Progressing);
call->setCallerIdNumber(to);
@@ -326,8 +334,10 @@ ManagerImpl::answerCall (CALLID id)
decWaitingCall();
call->setFlagNotAnswered(false);
}
- switchCall(id);
- stopTone(); // before answer, don't stop the audio stream after open it...
+ if (call->getState() != Call::OnHold) {
+ switchCall(id);
+ }
+ stopTone(); // before answer, don't stop the audio stream after open it
return call->answer();
}
@@ -347,7 +357,7 @@ ManagerImpl::onHoldCall (CALLID id)
if ( call->getState() == Call::OnHold || call->isNotAnswered()) {
return 1;
}
- _currentCallId = 0;
+ setCurrentCallId(0);
return call->onHold();
}
@@ -367,7 +377,8 @@ ManagerImpl::offHoldCall (CALLID id)
if (call->getState() == Call::OffHold) {
return 1;
}
- _currentCallId = id;
+ _debug("CALL: setting current id = %d\n", id);
+ setCurrentCallId(id);
int returnValue = call->offHold();
// start audio if it's ok
if (returnValue != -1) {
@@ -388,7 +399,7 @@ ManagerImpl::transferCall (CALLID id, const std::string& to)
if (call == 0) {
return -1;
}
- _currentCallId = 0;
+ setCurrentCallId(0);
return call->transfer(to);
}
@@ -431,7 +442,7 @@ ManagerImpl::refuseCall (CALLID id)
}
int refuse = call->refuse();
- _currentCallId = 0;
+ setCurrentCallId(0);
deleteCall(id);
stopTone();
return refuse;
@@ -780,7 +791,7 @@ ManagerImpl::peerHungupCall (CALLID id)
deleteCall(id);
call->setState(Call::Hungup);
- _currentCallId = 0;
+ setCurrentCallId(0);
return 1;
}
@@ -1580,10 +1591,11 @@ ManagerImpl::getDirListing(const std::string& sequenceId, const std::string& pat
void
ManagerImpl::switchCall(CALLID id)
{
- if (_currentCallId!=0 && id!=_currentCallId) {
- //onHoldCall(_currentCallId); <-- this function block _mutex...
+ // we can only switch the current call id if we
+ // it's not selected yet..
+ if (_currentCallId == 0 ) {
+ setCurrentCallId(id);
}
- _currentCallId = id;
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index cf885e066a..f986eddc35 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -290,9 +290,11 @@ private:
/*
* Erase the Call(id) from the CallVector
+ * Protected by other function by _mutex lock
*/
void deleteCall (CALLID id);
Call* getCall (CALLID id);
+ void setCurrentCallId(CALLID id);
/*
* Play one tone
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 0471f0344c..75a8fe4c26 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -409,7 +409,9 @@ SipVoIPLink::hangup (CALLID id)
// Release RTP channels
_debug("Stopping AudioRTP\n");
- _audiortp.closeRtpSession();
+ if (id == Manager::instance().getCurrentCallId()) {
+ _audiortp.closeRtpSession();
+ }
deleteSipCall(id);
return i;
@@ -684,8 +686,10 @@ SipVoIPLink::getEvent (void)
if (sipcall != 0) {
_debug(" Info [id = %d, cid = %d, did = %d], localport=%d\n", id, event->cid, event->did,sipcall->getLocalAudioPort());
- _debug("Stopping AudioRTP\n");
- _audiortp.closeRtpSession();
+ if ( id == Manager::instance().getCurrentCallId() ) {
+ _debug("Stopping AudioRTP\n");
+ _audiortp.closeRtpSession();
+ }
sipcall->newReinviteCall(event);
// we should receive an ack after that...
}
@@ -728,14 +732,13 @@ SipVoIPLink::getEvent (void)
sipcall->answeredCall_without_hold(event);
Manager::instance().peerAnsweredCall(id);
- if(!Manager::instance().callIsOnHold(id)) {
+ if(!Manager::instance().callIsOnHold(id) && Manager::instance().getCurrentCallId()==id) {
// Outgoing call is answered, start the sound channel.
_debug("Starting AudioRTP\n");
- if (_audiortp.createNewSession (sipcall) < 0) {
+ if (_audiortp.createNewSession(sipcall) < 0) {
_debug("FATAL: Unable to start sound (%s:%d)\n",
__FILE__, __LINE__);
returnValue = -1;
- break;
}
}
}
@@ -745,11 +748,11 @@ SipVoIPLink::getEvent (void)
sipcall->answeredCall(event);
}
}
- break;
} else {
returnValue = -1;
}
}
+ break;
case EXOSIP_CALL_REDIRECTED: // 11
break;
@@ -762,7 +765,7 @@ SipVoIPLink::getEvent (void)
sipcall->receivedAck(event);
if (sipcall->isReinvite()) {
sipcall->endReinvite();
- if(!Manager::instance().callIsOnHold(id)) {
+ if(!Manager::instance().callIsOnHold(id) && Manager::instance().getCurrentCallId()==id) {
_debug("Starting AudioRTP\n");
_audiortp.createNewSession(sipcall);
}
@@ -1502,6 +1505,7 @@ SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& t
}
// this is the cid (call id from exosip)
+ _debug("< INVITE (%d)", id);
int cid = eXosip_call_send_initial_invite (invite);
_debug("> Start a new Call: Send INVITE\n");
--
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