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Commit a1f6c4ac authored by Alexandre Bourget's avatar Alexandre Bourget
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IAX: Dynamic codecs + Fixed samplerate conversion + Ability to receive calls 

Added documentation.
parent 77751a2f
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src/iaxcall.cpp
+ 83
− 3
View file @ a1f6c4ac
......@@ -20,13 +20,93 @@
#include "iaxcall.h"
#include "global.h" // for _debug
IAXCall::IAXCall(const CallID& id, Call::CallType type) : Call(id, type), _session(0)
IAXCall::IAXCall(const CallID& id, Call::CallType type) : Call(id, type), _session(NULL)
{
}
IAXCall::~IAXCall()
{
_session = 0; // just to be sure to don't have unknown pointer, do not delete it!
_session = NULL; // just to be sure to don't have unknown pointer, do not delete it!
}
void
IAXCall::setFormat(int format)
{
_format = format;
switch(format) {
case AST_FORMAT_ULAW:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ULAW)); break;
case AST_FORMAT_GSM:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_GSM)); break;
case AST_FORMAT_ALAW:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ALAW)); break;
case AST_FORMAT_ILBC:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ILBC)); break;
case AST_FORMAT_SPEEX:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_SPEEX)); break;
default:
setAudioCodec(NULL);
break;
}
}
int
IAXCall::getSupportedFormat()
{
CodecMap map = getCodecMap().getMap();
int format = 0;
CodecMap::iterator iter = map.begin();
while(iter != map.end()) {
switch(iter->first) {
case PAYLOAD_CODEC_ULAW:
format |= AST_FORMAT_ULAW; break;
case PAYLOAD_CODEC_GSM:
format |= AST_FORMAT_GSM; break;
case PAYLOAD_CODEC_ALAW:
format |= AST_FORMAT_ALAW; break;
case PAYLOAD_CODEC_ILBC:
format |= AST_FORMAT_ILBC; break;
case PAYLOAD_CODEC_SPEEX:
format |= AST_FORMAT_SPEEX; break;
default:
break;
}
iter++;
}
return format;
}
int
IAXCall::getFirstMatchingFormat(int needles)
{
CodecMap map = getCodecMap().getMap();
int format = 0;
CodecMap::iterator iter = map.begin();
while(iter != map.end()) {
switch(iter->first) {
case PAYLOAD_CODEC_ULAW:
format = AST_FORMAT_ULAW; break;
case PAYLOAD_CODEC_GSM:
format = AST_FORMAT_GSM; break;
case PAYLOAD_CODEC_ALAW:
format = AST_FORMAT_ALAW; break;
case PAYLOAD_CODEC_ILBC:
format = AST_FORMAT_ILBC; break;
case PAYLOAD_CODEC_SPEEX:
format = AST_FORMAT_SPEEX; break;
default:
break;
}
// Return the first that matches
if (format & needles)
return format;
iter++;
}
return 0;
}
src/iaxcall.h
+ 42
− 4
View file @ a1f6c4ac
......@@ -21,6 +21,7 @@
#include "call.h"
#include <iax/iax-client.h>
#include <iax/frame.h>
/**
* IAXCall are IAX implementation of a normal Call
......@@ -33,7 +34,7 @@ public:
~IAXCall();
/** Get the session pointer or 0 */
/** Get the session pointer or NULL */
struct iax_session* getSession() { return _session; }
/** Set the session pointer
......@@ -41,12 +42,49 @@ public:
*/
void setSession(struct iax_session* session) { _session = session; }
void setFormat(int format) { _format = format; }
/**
* Set format (one single bit
*
* This function sets the _audioCodec variable with the correct
* codec.
*/
void setFormat(int format);
/**
* Get format
*/
int getFormat() { return _format; }
/**
* Get the bitwise list of supported formats
*/
int getSupportedFormat();
/**
* Return a format (int) with the first matching codec selected.
*
* This considers the order of the appearance in the CodecMap,
* thus, the order of preference.
*
* NOTE: Everything returned is bound to the content of the local
* CodecMap, so it won't return format values that aren't valid
* in this call context.
*
* @param needles The format(s) (bitwise) you are looking for to match
* @return The matching format, thus 0 if none matches
*/
int getFirstMatchingFormat(int needles);
private:
// each call is associate to a session
/** Each call is associated with an iax_session */
struct iax_session* _session;
/**
* Format currently in use in the conversation,
* sent in each outgoing voice packet.
*/
int _format;
};
......
src/iaxvoiplink.cpp
+ 349
− 203
View file @ a1f6c4ac
......@@ -27,6 +27,7 @@
#include <samplerate.h>
#include <iax/iax-client.h>
#include <math.h>
#define IAX_BLOCKING 1
......@@ -39,33 +40,14 @@
// from IAXC : iaxclient.h
/* payload formats : WARNING: must match libiax values!!! */
/* Data formats for capabilities and frames alike */
#define IAX__FORMAT_G723_1 (1 << 0) /* G.723.1 compression */
#define IAX__FORMAT_GSM (1 << 1) /* GSM compression */
#define IAX__FORMAT_ULAW (1 << 2) /* Raw mu-law data (G.711) */
#define IAX__FORMAT_ALAW (1 << 3) /* Raw A-law data (G.711) */
#define IAX__FORMAT_G726 (1 << 4) /* ADPCM, 32kbps */
#define IAX__FORMAT_ADPCM (1 << 5) /* ADPCM IMA */
#define IAX__FORMAT_SLINEAR (1 << 6) /* Raw 16-bit Signed Linear (8000 Hz) PCM */
#define IAX__FORMAT_LPC10 (1 << 7) /* LPC10, 180 samples/frame */
#define IAX__FORMAT_G729A (1 << 8) /* G.729a Audio */
#define IAX__FORMAT_SPEEX (1 << 9) /* Speex Audio */
#define IAX__FORMAT_ILBC (1 << 10) /* iLBC Audio */
#define IAX__FORMAT_MAX_AUDIO (1 << 15) /* Maximum audio format */
#define IAX__FORMAT_JPEG (1 << 16) /* JPEG Images */
#define IAX__FORMAT_PNG (1 << 17) /* PNG Images */
#define IAX__FORMAT_H261 (1 << 18) /* H.261 Video */
#define IAX__FORMAT_H263 (1 << 19) /* H.263 Video */
#define IAX__FORMAT_H263_PLUS (1 << 20) /* H.263+ Video */
#define IAX__FORMAT_MPEG4 (1 << 21) /* MPEG4 Video */
#define IAX__FORMAT_H264 (1 << 23) /* H264 Video */
#define IAX__FORMAT_THEORA (1 << 24) /* Theora Video */
#define IAX__20S_8KHZ_MAX 320 // 320 samples
#define IAX__20S_48KHZ_MAX 1920 // 320*6 samples, 6 = 48000/8000
#define CHK_VALID_CALL if (call == NULL) { _debug("IAX: Call doesn't exists\n"); \
return false; }
IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID)
{
......@@ -75,7 +57,6 @@ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
// to get random number for RANDOM_PORT
srand (time(NULL));
audiocodec = NULL;
audiolayer = NULL;
_receiveDataDecoded = new int16[IAX__20S_48KHZ_MAX];
......@@ -86,6 +67,11 @@ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
_floatBuffer8000 = new float32[IAX__20S_8KHZ_MAX];
_floatBuffer48000 = new float32[IAX__20S_48KHZ_MAX];
_intBuffer8000 = new int16[IAX__20S_8KHZ_MAX];
// libsamplerate-related
_src_state_mic = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
_src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
}
......@@ -95,7 +81,6 @@ IAXVoIPLink::~IAXVoIPLink()
_regSession = NULL; // shall not delete it
terminate();
audiocodec = NULL;
audiolayer = NULL;
delete [] _intBuffer8000; _intBuffer8000 = NULL;
delete [] _floatBuffer48000; _floatBuffer48000 = NULL;
......@@ -104,6 +89,10 @@ IAXVoIPLink::~IAXVoIPLink()
delete [] _sendDataEncoded; _sendDataEncoded = NULL;
delete [] _receiveDataDecoded; _receiveDataDecoded = NULL;
// libsamplerate-related
_src_state_mic = src_delete(_src_state_mic);
_src_state_spkr = src_delete(_src_state_spkr);
}
bool
......@@ -133,8 +122,6 @@ IAXVoIPLink::init()
_evThread->start();
// audio stuff, not dynamic yet
audiocodec = Manager::instance().getCodecDescriptorMap().getCodec((CodecType)0);
audiolayer = Manager::instance().getAudioDriver();
break;
}
......@@ -167,8 +154,8 @@ IAXVoIPLink::terminateIAXCall()
_mutexIAX.enterMutex();
iax_hangup(call->getSession(),"Dumped Call");
_mutexIAX.leaveMutex();
call->setSession(0);
delete call; call = 0;
call->setSession(NULL);
delete call; call = NULL;
}
iter++;
}
......@@ -183,45 +170,45 @@ IAXVoIPLink::getEvent()
iax_event* event = NULL;
IAXCall* call = NULL;
while ( (event = iax_get_event(IAX_BLOCKING)) != NULL ) {
//_debug ("Receive IAX Event: %d\n", event->etype);
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 != 0) {
if (call) {
_debug(" - We've got an associated call, handle call event\n");
iaxHandleCallEvent(event, call);
} else if (event->session != 0 && event->session == _regSession) {
} else if (event->session && event->session == _regSession) {
_debug(" - We've got an associated REGISTRATION session, handle registration process\n");
// in iaxclient, there is many session handling, here, only one
iaxHandleRegReply(event);
} else {
switch(event->etype) {
case IAX_EVENT_REGACK:
case IAX_EVENT_REGREJ:
_debug("Unknown IAX Registration Event\n");
break;
case IAX_EVENT_REGREQ:
_debug("Registration by a peer, don't allow it\n");
break;
case IAX_EVENT_CONNECT: // new call
// New incoming call!
break;
case IAX_EVENT_TIMEOUT: // timeout for an unknown session
} else {
break;
_debug (" - We've got some other event, deal with them alone.\n");
iaxHandlePrecallEvent(event);
default:
_debug("Unknown event type: %d\n", event->etype);
}
}
iax_event_free(event);
}
// Woah, we should do that in another thread, which will always send out stuff..
// send sound here
if(_currentCall != 0 && audiolayer != 0) {
if(_currentCall && audiolayer) {
int samples = audiolayer->canGetMic();
if (samples != 0) {
int datalen = audiolayer->getMic(_sendDataEncoded, samples);
_debug("iax_send_voice(%p, %d, ,%d, %d)\n", _currentCall->getSession(), _currentCall->getFormat(), datalen, samples);
//int datalen = audiolayer->getMic(_sendDataEncoded, samples);
//_debug("iax_send_voice(%p, %d, ,%d, %d)\n", _currentCall->getSession(), _currentCall->getFormat(), datalen, samples);
//if ( iax_send_voice(_currentCall->getSession(), _currentCall->getFormat(), (char*)_sendDataEncoded, datalen, samples) == -1) {
// // error sending voice
//}
......@@ -236,6 +223,22 @@ IAXVoIPLink::getEvent()
_evThread->sleep(5);
}
IAXCall*
IAXVoIPLink::getIAXCall(const CallID& id)
{
Call* call = getCall(id);
if (call) {
return dynamic_cast<IAXCall*>(call);
}
return 0;
}
bool
IAXVoIPLink::setRegister()
{
......@@ -278,10 +281,13 @@ IAXVoIPLink::setRegister()
return result;
}
bool
IAXVoIPLink::setUnregister()
{
if (_regSession == NULL) {
if (_regSession) {
// lock here
_mutexIAX.enterMutex();
iax_destroy(_regSession);
......@@ -297,31 +303,36 @@ Call*
IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
IAXCall* call = new IAXCall(id, Call::Outgoing);
if (call) {
call->setPeerNumber(toUrl);
// we have to add the codec before using it in SIPOutgoingInvite...
//call->setCodecMap(Manager::instance().getCodecDescriptorMap());
if ( iaxOutgoingInvite(call) ) {
call->setConnectionState(Call::Progressing);
call->setState(Call::Active);
addCall(call);
} else {
delete call; call = 0;
delete call; call = NULL;
}
}
return call;
}
bool
IAXVoIPLink::answer(const CallID& id)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
_mutexIAX.enterMutex();
iax_answer(call->getSession());
_mutexIAX.leaveMutex();
call->setState(Call::Active);
call->setConnectionState(Call::Connected);
return true;
}
......@@ -329,12 +340,13 @@ bool
IAXVoIPLink::hangup(const CallID& id)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
_mutexIAX.enterMutex();
iax_hangup(call->getSession(), "Dumped Call");
_mutexIAX.leaveMutex();
call->setSession(0);
call->setSession(NULL);
if (Manager::instance().isCurrentCall(id)) {
// stop audio
}
......@@ -346,11 +358,15 @@ bool
IAXVoIPLink::onhold(const CallID& id)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
//if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
_mutexIAX.enterMutex();
iax_quelch(call->getSession());
_mutexIAX.leaveMutex();
call->setState(Call::Hold);
return true;
}
......@@ -359,7 +375,9 @@ bool
IAXVoIPLink::offhold(const CallID& id)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
//if (call->getState() == Call::Active) { _debug("Call is already active\n"); return false; }
_mutexIAX.enterMutex();
iax_unquelch(call->getSession());
......@@ -372,7 +390,8 @@ bool
IAXVoIPLink::transfer(const CallID& id, const std::string& to)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
char callto[to.length()+1];
strcpy(callto, to.c_str());
......@@ -389,7 +408,9 @@ bool
IAXVoIPLink::refuse(const CallID& id)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
_mutexIAX.enterMutex();
iax_reject(call->getSession(), "Call rejected manually.");
_mutexIAX.leaveMutex();
......@@ -400,23 +421,25 @@ bool
IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
IAXCall* call = getIAXCall(id);
if (call==0) { _debug("Call doesn't exists\n"); return false; }
CHK_VALID_CALL;
_mutexIAX.enterMutex();
iax_send_dtmf(call->getSession(), code);
_mutexIAX.leaveMutex();
}
bool
IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
{
struct iax_session *newsession;
// lock here
_mutexIAX.enterMutex();
ost::MutexLock m(_mutexIAX);
newsession = iax_session_new();
if (!newsession) {
_debug("IAX Error: Can't make new session for a new call\n");
// unlock here
_mutexIAX.leaveMutex();
return false;
}
call->setSession(newsession);
......@@ -433,14 +456,13 @@ IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
char* lang = NULL;
int wait = 0;
int audio_format_preferred = IAX__FORMAT_ULAW;
int audio_format_capability = IAX__FORMAT_ULAW; // | IAX__FORMAT_ALAW | IAX__FORMAT_GSM | IAX__FORMAT_SPEEX;
/** @todo Make preference dynamic, and configurable */
int audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat());
int 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);
// unlock here
_mutexIAX.leaveMutex();
return true;
}
......@@ -451,7 +473,7 @@ IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
// access to callMap shoud use that
// the code below is like findSIPCallWithCid()
ost::MutexLock m(_callMapMutex);
IAXCall* call = 0;
IAXCall* call = NULL;
CallMap::iterator iter = _callMap.begin();
while(iter != _callMap.end()) {
call = dynamic_cast<IAXCall*>(iter->second);
......@@ -460,7 +482,7 @@ IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
}
iter++;
}
return 0; // not found
return NULL; // not found
}
void
......@@ -496,7 +518,11 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
case IAX_EVENT_ACCEPT:
// accept
//
if (event->ies.format) {
call->setFormat(event->ies.format);
}
break;
case IAX_EVENT_ANSWER:
......@@ -504,7 +530,10 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
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);
_currentCall = call;
......@@ -525,18 +554,42 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
break;
case IAX_EVENT_VOICE:
if (audiocodec != NULL && audiolayer != NULL) {
//_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
if (audiolayer) {
AudioCodec* audiocodec = call->getAudioCodec();
if (!audiocodec) {
// libiax2 stores the voiceformat in the 'subclass' element.
if (event->subclass) {
// Set the format, with the first voice packet
call->setFormat(event->subclass);
audiocodec = call->getAudioCodec();
} else {
// Send a VNAK, because they sent a Mini packet before
// a full VOICE packet (with the format inside)
_debug("IAX: sending VNAK, received mini packet before full VOICE packet.");
iax_vnak(event->session);
break;
}
}
_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
unsigned char* data = (unsigned char*)event->data;
unsigned int size = event->datalen;
if (size > IAX__20S_8KHZ_MAX) {
_debug("the size %d is bigger than expected %d. crop\n", size, IAX__20S_8KHZ_MAX);
_debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, IAX__20S_8KHZ_MAX);
size = IAX__20S_8KHZ_MAX;
}
// On pourrait ajuster le codec dynamiquement ici, comme c'est fait dans SIP.
// à moins que IAX ne permette pas de changer le codec à chaque paquet.
int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
int nbInt16 = expandedSize/sizeof(int16);
if (nbInt16 > IAX__20S_8KHZ_MAX) {
_debug("We have decoded a IAX packet larger than expected: %s VS %s. crop\n", nbInt16, IAX__20S_8KHZ_MAX);
_debug("We have decoded a IAX VOICE packet larger than expected: %s VS %s. Cropping.\n", nbInt16, IAX__20S_8KHZ_MAX);
nbInt16 = IAX__20S_8KHZ_MAX;
}
......@@ -544,18 +597,21 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
int nbSample = nbInt16;
int nbSampleMaxRate = nbInt16 * 6;
if ( audiolayer->getSampleRate() != audiocodec->getClockRate() ) {
// convert here
if ( audiolayer->getSampleRate() != audiocodec->getClockRate() && nbSample ) {
// Do sample rate conversion
double factord = (double) audiolayer->getSampleRate() / audiocodec->getClockRate();
// SRC_DATA from samplerate.h
SRC_DATA src_data;
src_data.data_in = _floatBuffer8000;
src_data.data_out = _floatBuffer48000;
src_data.input_frames = nbSample;
src_data.output_frames = nbSampleMaxRate;
src_data.output_frames = (int) floor(factord * nbSample);
src_data.src_ratio = factord;
src_data.end_of_input = 0; /* More data will come */
src_short_to_float_array(_receiveDataDecoded, _floatBuffer8000, nbSample);
src_simple (&src_data, SRC_SINC_BEST_QUALITY/*SRC_SINC_MEDIUM_QUALITY*/, 1); // 1=mono channel
// samplerate convert, go!
src_process(_src_state_spkr, &src_data);
nbSample = ( src_data.output_frames_gen > IAX__20S_48KHZ_MAX) ? IAX__20S_48KHZ_MAX : src_data.output_frames_gen;
#ifdef DATAFORMAT_IS_FLOAT
......@@ -576,6 +632,8 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
#endif
}
audiolayer->putMain(toAudioLayer, nbSample * sizeof(SFLDataFormat));
} else {
_debug("IAX: incoming audio, but no sound card open");
}
break;
......@@ -607,26 +665,114 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
}
/**
* Handle the registration process
*/
void
IAXVoIPLink::iaxHandleRegReply(iax_event* event)
{
//unregister
if (event->etype == IAX_EVENT_REGREJ) {
/* Authentication failed! */
iax_destroy(_regSession);
_regSession = NULL;
Manager::instance().registrationFailed(getAccountID());
} else if (event->etype == IAX_EVENT_REGACK) {
}
else if (event->etype == IAX_EVENT_REGACK) {
/* Authentication succeeded */
Manager::instance().registrationSucceed(getAccountID());
}
}
IAXCall*
IAXVoIPLink::getIAXCall(const CallID& id)
void
IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
{
Call* call = getCall(id);
if (call) {
return dynamic_cast<IAXCall*>(call);
IAXCall* call = NULL;
CallID id;
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_CONNECT:
// We've got an incoming call! Yikes!
_debug("> IAX_EVENT_CONNECT (receive)\n");
id = Manager::instance().getNewCallID();
call = new IAXCall(id, Call::Incoming);
if (!call) {
_debug("! IAX Failure: unable to create an incoming call");
return;
}
return 0;
// 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);
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 */
// 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;
iax_accept(event->session, format);
iax_ring_announce(event->session);
addCall(call);
} else {
// reject call, unable to add it
iax_reject(event->session, "Error ringing user.");
delete call; call = NULL;
}
break;
case IAX_EVENT_HANGUP:
// Remote peer hung up
call = iaxFindCallBySession(event->session);
id = call->getCallId();
Manager::instance().peerHungupCall(id);
removeCall(id);
break;
case IAX_EVENT_TIMEOUT: // timeout for an unknown session
break;
default:
_debug("Unknown event type: %d\n", event->etype);
}
}
src/iaxvoiplink.h
+ 41
− 3
View file @ a1f6c4ac
......@@ -22,6 +22,7 @@
#include "voiplink.h"
#include <iax/iax-client.h>
#include "global.h"
#include <samplerate.h>
class EventThread;
......@@ -49,6 +50,15 @@ public:
void terminate (void);
bool setRegister (void);
/**
* Destroy registration session
*
* @todo Send an IAX_COMMAND_REGREL to force unregistration upstream.
* Urgency: low
*
* @return bool If we're registered upstream
*/
bool setUnregister (void);
Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
......@@ -101,26 +111,45 @@ private:
*/
void iaxHandleRegReply(iax_event* event);
/**
* Handle IAX pre-call setup-related events
* @param event An iax_event pointer
*/
void iaxHandlePrecallEvent(iax_event* event);
/**
* Send an outgoing call invite to iax
* @param call An IAXCall pointer
*/
bool iaxOutgoingInvite(IAXCall* Call);
bool iaxOutgoingInvite(IAXCall* call);
/**
* Convert CodecMap to IAX format using IAX constants
* @return `format` ready to go into iax_* calls
*/
int iaxCodecMapToFormat(IAXCall* call);
/** Threading object */
EventThread* _evThread;
/** registration session : 0 if not register */
struct iax_session* _regSession;
/** IAX Host */
std::string _host;
/** IAX User */
std::string _user;
/** IAX Password */
std::string _pass;
/** IAX full name */
std::string _fullName;
ost::Mutex _mutexIAX;
// extra pointer / not dynamic yet
AudioCodec* audiocodec;
AudioLayer* audiolayer;
/** When we receive data, we decode it inside this buffer */
......@@ -140,6 +169,15 @@ private:
/** Buffer for 8000hz samples for mic conversion */
int16* _intBuffer8000;
/** libsamplerate converter for incoming voice */
SRC_STATE* _src_state_spkr;
/** libsamplerate converter for outgoing voice */
SRC_STATE* _src_state_mic;
/** libsamplerate error */
int _src_err;
/** Current IAX call pointer, used for sending, change when starting audio, switching */
IAXCall* _currentCall;
};
......
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