diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.cpp b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
index 5ce37ce1f53c08fc342d4b60daff3beefc3173dd..4b4b3c5c0e1ad676e37e0e1dde0d30e0038184d9 100644
--- a/daemon/src/audio/audiortp/audio_rtp_factory.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
@@ -43,9 +43,9 @@
namespace sfl {
AudioRtpFactory::AudioRtpFactory(SIPCall *ca) : rtpSession_(NULL),
- audioRtpThreadMutex_(), srtpEnabled_(false),
- keyExchangeProtocol_(Symmetric), helloHashEnabled_(false),
- remoteContext_(NULL), localContext_(NULL), ca_(ca)
+ audioRtpThreadMutex_(), srtpEnabled_(false), helloHashEnabled_(false),
+ remoteContext_(NULL), localContext_(NULL), ca_(ca),
+ keyExchangeProtocol_(NONE)
{}
AudioRtpFactory::~AudioRtpFactory()
@@ -53,7 +53,7 @@ AudioRtpFactory::~AudioRtpFactory()
delete rtpSession_;
}
-void AudioRtpFactory::initAudioRtpConfig()
+void AudioRtpFactory::initConfig()
{
if (rtpSession_ != NULL)
stop();
@@ -65,23 +65,23 @@ void AudioRtpFactory::initAudioRtpConfig()
if (account) {
srtpEnabled_ = account->getSrtpEnabled();
std::string key(account->getSrtpKeyExchange());
-
- if (key == "sdes")
- keyExchangeProtocol_ = Sdes;
- else if (key == "zrtp")
- keyExchangeProtocol_ = Zrtp;
- else
- keyExchangeProtocol_ = Symmetric;
-
+ if (srtpEnabled_) {
+ if (key == "sdes")
+ keyExchangeProtocol_ = SDES;
+ else if (key == "zrtp")
+ keyExchangeProtocol_ = ZRTP;
+ } else {
+ keyExchangeProtocol_ = NONE;
+ }
helloHashEnabled_ = account->getZrtpHelloHash();
} else {
srtpEnabled_ = false;
- keyExchangeProtocol_ = Symmetric;
+ keyExchangeProtocol_ = NONE;
helloHashEnabled_ = false;
}
}
-void AudioRtpFactory::initAudioSymmetricRtpSession()
+void AudioRtpFactory::initSession()
{
ost::MutexLock m(audioRtpThreadMutex_);
@@ -90,7 +90,7 @@ void AudioRtpFactory::initAudioSymmetricRtpSession()
switch (keyExchangeProtocol_) {
- case Zrtp:
+ case ZRTP:
rtpSession_ = new AudioZrtpSession(ca_, zidFilename);
// TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
// not even available at that point.
@@ -98,7 +98,7 @@ void AudioRtpFactory::initAudioSymmetricRtpSession()
ca_->getLocalSDP()->setZrtpHash(static_cast<AudioZrtpSession *>(rtpSession_)->getHelloHash());
break;
- case Sdes:
+ case SDES:
rtpSession_ = new AudioSrtpSession(ca_);
break;
@@ -114,12 +114,11 @@ void AudioRtpFactory::start(AudioCodec* audiocodec)
if (rtpSession_ == NULL)
throw AudioRtpFactoryException("AudioRtpFactory: Error: RTP session was null when trying to start audio thread");
- if (rtpSession_->getAudioRtpType() == Sdes)
- if (localContext_ and remoteContext_)
- static_cast<AudioSrtpSession *>(rtpSession_)->restoreCryptoContext(localContext_, remoteContext_);
+ if (keyExchangeProtocol_ == SDES and localContext_ and remoteContext_)
+ static_cast<AudioSrtpSession *>(rtpSession_)->restoreCryptoContext(localContext_, remoteContext_);
if (rtpSession_->startRtpThread(audiocodec) != 0)
- throw AudioRtpFactoryException("AudioRtpFactory: Error: Failed to start AudioZrtpSession thread");
+ throw AudioRtpFactoryException("AudioRtpFactory: Error: Failed to start AudioRtpSession thread");
}
void AudioRtpFactory::stop()
@@ -129,7 +128,7 @@ void AudioRtpFactory::stop()
if (rtpSession_ == NULL)
return;
- if (rtpSession_->getAudioRtpType() == Sdes) {
+ if (keyExchangeProtocol_ == SDES) {
localContext_ = static_cast<AudioSrtpSession*>(rtpSession_)->localCryptoCtx_;
remoteContext_ = static_cast<AudioSrtpSession*>(rtpSession_)->remoteCryptoCtx_;
}
@@ -162,7 +161,7 @@ void AudioRtpFactory::updateDestinationIpAddress()
sfl::AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
{
- if (rtpSession_->getAudioRtpType() == Zrtp)
+ if (keyExchangeProtocol_ == ZRTP)
return static_cast<AudioZrtpSession *>(rtpSession_);
else
throw AudioRtpFactoryException("RTP: Error: rtpSession_ is NULL in getAudioZrtpSession");
@@ -170,15 +169,16 @@ sfl::AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
void sfl::AudioRtpFactory::initLocalCryptoInfo()
{
- if (rtpSession_ && rtpSession_->getAudioRtpType() == Sdes) {
- static_cast<AudioSrtpSession *>(rtpSession_)->initLocalCryptoInfo();
- ca_->getLocalSDP()->setLocalSdpCrypto(static_cast<AudioSrtpSession *>(rtpSession_)->getLocalCryptoInfo());
+ if (rtpSession_ && keyExchangeProtocol_ == SDES) {
+ AudioSrtpSession *srtpSession = static_cast<AudioSrtpSession*>(rtpSession_);
+ srtpSession->initLocalCryptoInfo();
+ ca_->getLocalSDP()->setLocalSdpCrypto(srtpSession->getLocalCryptoInfo());
}
}
void AudioRtpFactory::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
{
- if (rtpSession_ && rtpSession_->getAudioRtpType() == Sdes)
+ if (rtpSession_ and keyExchangeProtocol_ == SDES)
static_cast<AudioSrtpSession *>(rtpSession_)->setRemoteCryptoInfo(nego);
else
throw AudioRtpFactoryException("RTP: Error: rtpSession_ is NULL in setRemoteCryptoInfo");
diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.h b/daemon/src/audio/audiortp/audio_rtp_factory.h
index 9e8ca514ceb67216b48bb6408de2f1eb88d0f89d..f39d6519d33f4895694fe27ecaffd8ad4a820ba1 100644
--- a/daemon/src/audio/audiortp/audio_rtp_factory.h
+++ b/daemon/src/audio/audiortp/audio_rtp_factory.h
@@ -62,15 +62,15 @@ class AudioRtpFactory {
AudioRtpFactory(SIPCall *ca);
~AudioRtpFactory();
- void initAudioRtpConfig();
+ void initConfig();
/**
* Lazy instantiation method. Create a new RTP session of a given
* type according to the content of the configuration file.
* @param ca A pointer on a SIP call
- * @return A new AudioSymmetricRtpSession object
+ * @return A new AudioRtpSession object
*/
- void initAudioSymmetricRtpSession();
+ void initSession();
/**
* Start the audio rtp thread of the type specified in the configuration
@@ -103,7 +103,7 @@ class AudioRtpFactory {
void updateDestinationIpAddress();
bool isSdesEnabled() const {
- return srtpEnabled_ and keyExchangeProtocol_ == sfl::Sdes;
+ return srtpEnabled_ and keyExchangeProtocol_ == SDES;
}
/**
@@ -140,17 +140,14 @@ class AudioRtpFactory {
private:
NON_COPYABLE(AudioRtpFactory);
+ enum KeyExchangeProtocol { NONE, SDES, ZRTP };
AudioRtpSession *rtpSession_;
ost::Mutex audioRtpThreadMutex_;
- // Field used when initializinga udio rtp session
+ // Field used when initializing audio rtp session
// May be set manually or from config using initAudioRtpConfig
bool srtpEnabled_;
- // Field used when initializinga udio rtp session
- // May be set manually or from config using initAudioRtpConfig
- RtpMethod keyExchangeProtocol_;
-
// Field used when initializinga udio rtp session
// May be set manually or from config using initAudioRtpConfig
bool helloHashEnabled_;
@@ -162,6 +159,7 @@ class AudioRtpFactory {
ost::CryptoContext *localContext_;
SIPCall *ca_;
+ KeyExchangeProtocol keyExchangeProtocol_;
};
}
#endif // __AUDIO_RTP_FACTORY_H__
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
index 2a6bea42e7e2f404cbe47751b1a2c1e3fb857a22..0df3c691dae301ebd7118866453910a22e3f3f03 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_session.cpp
@@ -43,17 +43,16 @@
#include "manager.h"
namespace sfl {
-AudioRtpSession::AudioRtpSession(SIPCall * sipcall, RtpMethod type, ost::RTPDataQueue *queue, ost::Thread *thread) :
+AudioRtpSession::AudioRtpSession(SIPCall * sipcall, ost::RTPDataQueue *queue, ost::Thread *thread) :
AudioRtpRecordHandler(sipcall)
, ca_(sipcall)
- , type_(type)
- , remote_ip_()
- , remote_port_(0)
, timestamp_(0)
, timestampIncrement_(0)
- , timestampCount_(0)
- , isStarted_(false)
, queue_(queue)
+ , isStarted_(false)
+ , remote_ip_()
+ , remote_port_(0)
+ , timestampCount_(0)
, thread_(thread)
{
assert(ca_);
@@ -96,26 +95,28 @@ void AudioRtpSession::setSessionMedia(AudioCodec *audioCodec)
else
timestampIncrement_ = frameSize;
- DEBUG("AudioRptSession: Codec payload: %d", payloadType);
- DEBUG("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate);
- DEBUG("AudioSymmetricRtpSession: Codec frame size: %d", frameSize);
- DEBUG("AudioSymmetricRtpSession: RTP timestamp increment: %d", timestampIncrement_);
+ DEBUG("AudioRtpSession: Codec payload: %d", payloadType);
+ DEBUG("AudioRtpSession: Codec sampling rate: %d", smplRate);
+ DEBUG("AudioRtpSession: Codec frame size: %d", frameSize);
+ DEBUG("AudioRtpSession: RTP timestamp increment: %d", timestampIncrement_);
if (payloadType == g722PayloadType) {
- DEBUG("AudioSymmetricRtpSession: Setting G722 payload format");
+ DEBUG("AudioRtpSession: Setting G722 payload format");
queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
} else {
if (dynamic) {
- DEBUG("AudioSymmetricRtpSession: Setting dynamic payload format");
+ DEBUG("AudioRtpSession: Setting dynamic payload format");
queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
} else {
- DEBUG("AudioSymmetricRtpSession: Setting static payload format");
+ DEBUG("AudioRtpSession: Setting static payload format");
queue_->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
}
}
+}
- if (type_ != Zrtp)
- ca_->setRecordingSmplRate(getCodecSampleRate());
+void AudioRtpSession::incrementTimestampForDTMF()
+{
+ timestamp_ += timestampIncrement_;
}
void AudioRtpSession::sendDtmfEvent()
@@ -131,7 +132,7 @@ void AudioRtpSession::sendDtmfEvent()
DEBUG("AudioRtpSession: Send RTP Dtmf (%d)", payload.event);
- timestamp_ += (type_ == Zrtp) ? 160 : timestampIncrement_;
+ incrementTimestampForDTMF();
// discard equivalent size of audio
processDataEncode();
@@ -166,22 +167,18 @@ void AudioRtpSession::receiveSpeakerData()
}
-
void AudioRtpSession::sendMicData()
{
int compSize = processDataEncode();
// if no data return
- if (!compSize)
+ if (compSize == 0)
return;
// Increment timestamp for outgoing packet
timestamp_ += timestampIncrement_;
- if (type_ == Zrtp)
- queue_->putData(timestamp_, getMicDataEncoded(), compSize);
-
- // putData put the data on RTP queue, sendImmediate bypass this queue
+ // putData puts the data on RTP queue, sendImmediate bypass this queue
queue_->sendImmediate(timestamp_, getMicDataEncoded(), compSize);
}
@@ -238,7 +235,7 @@ int AudioRtpSession::startRtpThread(AudioCodec* audiocodec)
if (isStarted_)
return 0;
- DEBUG("AudioSymmetricRtpSession: Starting main thread");
+ DEBUG("AudioRtpSession: Starting main thread");
isStarted_ = true;
setSessionTimeouts();
@@ -247,14 +244,7 @@ int AudioRtpSession::startRtpThread(AudioCodec* audiocodec)
initNoiseSuppress();
queue_->enableStack();
- int ret = thread_->start();
-
- if (type_ == Zrtp)
- return ret;
-
- AudioSymmetricRtpSession *self = dynamic_cast<AudioSymmetricRtpSession*>(this);
- assert(self);
- return self->startSymmetricRtpThread();
+ return thread_->start();
}
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.h b/daemon/src/audio/audiortp/audio_rtp_session.h
index 4f8723b234f0897490cadd65faa237518d02bbfc..f52f6472ef4cadc3fa7364db3556cef6c76e97db 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_rtp_session.h
@@ -46,29 +46,18 @@ namespace sfl {
class AudioCodec;
-// Possible kind of rtp session
-typedef enum RtpMethod {
- Symmetric,
- Zrtp,
- Sdes
-} RtpMethod;
-
-
class AudioRtpSession : public AudioRtpRecordHandler {
public:
/**
* Constructor
* @param sipcall The pointer on the SIP call
*/
- AudioRtpSession(SIPCall* sipcall, RtpMethod type, ost::RTPDataQueue *queue, ost::Thread *thread);
+ AudioRtpSession(SIPCall* sipcall, ost::RTPDataQueue *queue, ost::Thread *thread);
virtual ~AudioRtpSession();
- RtpMethod getAudioRtpType() {
- return type_;
- }
void updateSessionMedia(AudioCodec *audioCodec);
- int startRtpThread(AudioCodec*);
+ virtual int startRtpThread(AudioCodec*);
/**
* Used mostly when receiving a reinvite
@@ -76,6 +65,11 @@ class AudioRtpSession : public AudioRtpRecordHandler {
void updateDestinationIpAddress();
protected:
+ /**
+ * Set the audio codec for this RTP session
+ */
+ virtual void setSessionMedia(AudioCodec *codec) = 0;
+
bool onRTPPacketRecv(ost::IncomingRTPPkt&);
@@ -90,19 +84,26 @@ class AudioRtpSession : public AudioRtpRecordHandler {
/**
* Send encoded data to peer
*/
- void sendMicData();
+ virtual void sendMicData();
SIPCall *ca_;
- RtpMethod type_;
-
- private:
- NON_COPYABLE(AudioRtpSession);
+ /**
+ * Timestamp for this session
+ */
+ int timestamp_;
/**
- * Set the audio codec for this RTP session
+ * Timestamp incrementation value based on codec period length (framesize)
+ * except for G722 which require a 8 kHz incrementation.
*/
- void setSessionMedia(AudioCodec*);
+ int timestampIncrement_;
+
+ ost::RTPDataQueue *queue_;
+
+ bool isStarted_;
+ private:
+ NON_COPYABLE(AudioRtpSession);
/**
* Set RTP Sockets send/receive timeouts
@@ -119,6 +120,11 @@ class AudioRtpSession : public AudioRtpRecordHandler {
*/
void receiveSpeakerData();
+ /**
+ * Increment timestamp for DTMF event
+ */
+ virtual void incrementTimestampForDTMF();
+
// Main destination address for this rtp session.
// Stored in case or reINVITE, which may require to forget
// this destination and update a new one.
@@ -129,26 +135,11 @@ class AudioRtpSession : public AudioRtpRecordHandler {
// this destination and update a new one
unsigned short remote_port_;
- /**
- * Timestamp for this session
- */
- int timestamp_;
-
- /**
- * Timestamp incrementation value based on codec period length (framesize)
- * except for G722 which require a 8 kHz incrementation.
- */
- int timestampIncrement_;
-
/**
* Timestamp reset frequency specified in number of packet sent
*/
short timestampCount_;
- bool isStarted_;
-
- ost::RTPDataQueue *queue_;
-
ost::Thread *thread_;
};
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.cpp b/daemon/src/audio/audiortp/audio_srtp_session.cpp
index ed8a637fc07953ce9a04e3879bab053b62084f5a..60e173ba62a63da2528abd7b638a51f789885c52 100644
--- a/daemon/src/audio/audiortp/audio_srtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_srtp_session.cpp
@@ -59,14 +59,7 @@ AudioSrtpSession::AudioSrtpSession(SIPCall * sipcall) :
remoteMasterSalt_(),
remoteMasterSaltLength_(0),
remoteOfferIsSet_(false)
-{
- type_ = Sdes;
-}
-
-AudioSrtpSession::~AudioSrtpSession()
-{
- DEBUG("AudioSrtp: Destroy audio srtp session");
-}
+{}
void AudioSrtpSession::initLocalCryptoInfo()
{
@@ -326,4 +319,5 @@ char* AudioSrtpSession::decodeBase64(unsigned char *input, int length)
return buffer;
}
+
}
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.h b/daemon/src/audio/audiortp/audio_srtp_session.h
index f641e8eaa55b21065e7efcee35eb87f3543f93eb..02429d7b193d63a8777f7dfc441bdb1130f2a200 100644
--- a/daemon/src/audio/audiortp/audio_srtp_session.h
+++ b/daemon/src/audio/audiortp/audio_srtp_session.h
@@ -75,8 +75,6 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
*/
AudioSrtpSession(SIPCall * sipcall);
- ~AudioSrtpSession();
-
/**
* Used to get sdp crypto header to be included in sdp session. This
* method must be called befor setRemoteCryptoInfo in case of an
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
index cc5d9ba37e068ad94c070a739cde6d6079b5116d..5169ce97eb44be937033f2eb918b460e823c4d88 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
@@ -44,8 +44,7 @@ namespace sfl {
AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall * sipcall) :
ost::TimerPort()
, ost::SymmetricRTPSession(ost::InetHostAddress(sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort())
- , AudioRtpSession(sipcall, Symmetric, this, this)
- , echoCanceller()
+ , AudioRtpSession(sipcall, this, this)
, rtpThread_(new AudioRtpThread(this))
{
DEBUG("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
@@ -54,12 +53,14 @@ AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall * sipcall) :
AudioSymmetricRtpSession::~AudioSymmetricRtpSession()
{
- rtpThread_->running = false;
+ rtpThread_->running_ = false;
delete rtpThread_;
}
-AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread(AudioSymmetricRtpSession *session) : running(true), rtpSession(session)
-{}
+AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread(AudioSymmetricRtpSession *session) : running_(true), rtpSession_(session)
+{
+ assert(rtpSession_);
+}
void AudioSymmetricRtpSession::AudioRtpThread::run()
{
@@ -69,12 +70,12 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
DEBUG("AudioRtpThread: Entering Audio rtp thread main loop");
- while (running) {
+ while (running_) {
// Send session
- if (rtpSession->DtmfPending())
- rtpSession->sendDtmfEvent();
+ if (rtpSession_->DtmfPending())
+ rtpSession_->sendDtmfEvent();
else
- rtpSession->sendMicData();
+ rtpSession_->sendMicData();
Thread::sleep(TimerPort::getTimer());
@@ -84,4 +85,20 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
DEBUG("AudioRtpThread: Leaving audio rtp thread loop");
}
+void AudioSymmetricRtpSession::setSessionMedia(AudioCodec *audioCodec)
+{
+ AudioRtpSession::setSessionMedia(audioCodec);
+ ca_->setRecordingSmplRate(getCodecSampleRate());
+}
+
+int AudioSymmetricRtpSession::startRtpThread(AudioCodec* audiocodec)
+{
+ DEBUG("AudioSymmetricRtpSession: Starting main thread");
+ if (isStarted_)
+ return 0;
+
+ AudioRtpSession::startRtpThread(audiocodec);
+ return startSymmetricRtpThread();
+}
+
}
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
index 256e55ae69f53ab36f349dbd7bb478e4f311ffd7..96270012d262ecedeacbbf632a75a9160417f85a 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
@@ -47,7 +47,6 @@ using std::ptrdiff_t;
#pragma GCC diagnostic ignored "-Weffc++"
#include <ccrtp/rtp.h>
#include <ccrtp/iqueue.h>
-#include <cc++/numbers.h> // ost::Time
class SIPCall;
@@ -60,7 +59,6 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
* @param sipcall The pointer on the SIP call
*/
AudioSymmetricRtpSession(SIPCall* sipcall);
-
~AudioSymmetricRtpSession();
virtual bool onRTPPacketRecv(ost::IncomingRTPPkt& pkt) {
@@ -68,7 +66,6 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
}
int startSymmetricRtpThread() {
- assert(rtpThread_);
return rtpThread_->start();
}
@@ -78,19 +75,18 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
class AudioRtpThread : public ost::Thread, public ost::TimerPort {
public:
AudioRtpThread(AudioSymmetricRtpSession *session);
- ~AudioRtpThread(){}
virtual void run();
- bool running;
+ bool running_;
private:
NON_COPYABLE(AudioRtpThread);
- AudioSymmetricRtpSession *rtpSession;
+ AudioSymmetricRtpSession *rtpSession_;
};
- SpeexEchoCancel echoCanceller;
+ void setSessionMedia(AudioCodec *codec);
+ int startRtpThread(AudioCodec* audiocodec);
- private:
AudioRtpThread *rtpThread_;
};
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.cpp b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
index ebdb8f839e421b1494341b8a19bf846f9c6218c2..81903a9aae11eb9b2a5878d68064b3d15419798e 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
@@ -50,7 +50,7 @@
namespace sfl {
AudioZrtpSession::AudioZrtpSession(SIPCall * sipcall, const std::string& zidFilename) :
- AudioRtpSession(sipcall, Zrtp, this, this),
+ AudioRtpSession(sipcall, this, this),
ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue>(ost::InetHostAddress(sipcall->getLocalIp().c_str()),
sipcall->getLocalAudioPort(),
0,
@@ -117,6 +117,24 @@ void AudioZrtpSession::initializeZid()
return;
}
+void AudioZrtpSession::sendMicData()
+{
+ int compSize = processDataEncode();
+
+ // if no data return
+ if (compSize == 0)
+ return;
+
+ // Increment timestamp for outgoing packet
+ timestamp_ += timestampIncrement_;
+
+ // this step is only needed for ZRTP
+ queue_->putData(timestamp_, getMicDataEncoded(), compSize);
+
+ // putData puts the data on RTP queue, sendImmediate bypasses this queue
+ queue_->sendImmediate(timestamp_, getMicDataEncoded(), compSize);
+}
+
void AudioZrtpSession::run()
{
// Set recording sampling rate
@@ -165,4 +183,17 @@ void AudioZrtpSession::run()
DEBUG("AudioZrtpSession: Left main loop for call %s", ca_->getCallId().c_str());
}
+
+
+void AudioZrtpSession::incrementTimestampForDTMF()
+{
+ timestamp_ += 160;
+}
+
+
+void AudioZrtpSession::setSessionMedia(AudioCodec *audioCodec)
+{
+ AudioRtpSession::setSessionMedia(audioCodec);
+}
+
}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.h b/daemon/src/audio/audiortp/audio_zrtp_session.h
index 3a19a5e3fd148e2e9090965a0b058bb56296407f..00d9dad819d875b9b8854bcaf2c164a5ea214e7d 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.h
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.h
@@ -43,6 +43,7 @@ using std::ptrdiff_t;
#include <cc++/numbers.h> // OST::Time
class SIPCall;
+class AudioCodec;
namespace sfl {
@@ -68,8 +69,11 @@ class AudioZrtpSession : public AudioRtpSession, protected ost::Thread, public o
}
private:
+ void sendMicData();
void initializeZid();
std::string zidFilename_;
+ void incrementTimestampForDTMF();
+ void setSessionMedia(AudioCodec *codec);
};
}
diff --git a/daemon/src/sip/sdes_negotiator.h b/daemon/src/sip/sdes_negotiator.h
index 7fb4213555d670af6bd32f68073584107a886c09..416f940b011eecd97d00e196a44e07cb146e688f 100644
--- a/daemon/src/sip/sdes_negotiator.h
+++ b/daemon/src/sip/sdes_negotiator.h
@@ -80,7 +80,7 @@ struct CryptoSuiteDefinition {
* List of accepted Crypto-Suites
* as defined in RFC4568 (6.2)
*/
-const CryptoSuiteDefinition CryptoSuites[3] = {
+static const CryptoSuiteDefinition CryptoSuites[] = {
{ (char*) "AES_CM_128_HMAC_SHA1_80", 128, 112, 48, 31, AESCounterMode, 128, HMACSHA1, 80, 80, 160, 160 },
{ (char*) "AES_CM_128_HMAC_SHA1_32", 128, 112, 48, 31, AESCounterMode, 128, HMACSHA1, 32, 80, 160, 160 },
{ (char*) "F8_128_HMAC_SHA1_80", 128, 112, 48, 31, AESF8Mode, 128, HMACSHA1, 80, 80, 160, 160 }
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index a6bf9c935e84cd10fa4b0b137d9a0f568f6eb34f..8a482f76fd7e1d5ecc75bf02cae0801e0240ad87 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -322,8 +322,8 @@ pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
call->getLocalSDP()->setLocalIP(addrSdp);
- call->getAudioRtp().initAudioRtpConfig();
- call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initConfig();
+ call->getAudioRtp().initSession();
if (rdata->msg_info.msg->body) {
char sdpbuffer[1000];
@@ -340,7 +340,7 @@ pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
CryptoOffer crypto_offer;
crypto_offer.push_back(std::string(sdpoffer.substr(start, (sdpoffer.size() - start) - 1)));
- std::vector<sfl::CryptoSuiteDefinition>localCapabilities;
+ std::vector<sfl::CryptoSuiteDefinition> localCapabilities;
for (int i = 0; i < 3; i++)
localCapabilities.push_back(sfl::CryptoSuites[i]);
@@ -699,8 +699,8 @@ Call *SIPVoIPLink::newOutgoingCall(const std::string& id, const std::string& toU
}
try {
- call->getAudioRtp().initAudioRtpConfig();
- call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initConfig();
+ call->getAudioRtp().initSession();
call->getAudioRtp().initLocalCryptoInfo();
call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
} catch (...) {
@@ -834,8 +834,8 @@ SIPVoIPLink::offhold(const std::string& id)
if (audiocodec == NULL)
throw VoipLinkException("Could not instantiate codec");
- call->getAudioRtp().initAudioRtpConfig();
- call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initConfig();
+ call->getAudioRtp().initSession();
call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
} catch (const SdpException &e) {
ERROR("UserAgent: Exception: %s", e.what());
@@ -1168,8 +1168,8 @@ bool SIPVoIPLink::SIPNewIpToIpCall(const std::string& id, const std::string& to)
// Audio Rtp Session must be initialized before creating initial offer in SDP session
// since SDES require crypto attribute.
- call->getAudioRtp().initAudioRtpConfig();
- call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initConfig();
+ call->getAudioRtp().initSession();
call->getAudioRtp().initLocalCryptoInfo();
call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
@@ -1416,7 +1416,7 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
std::string accountID = Manager::instance().getAccountFromCall(call->getCallId());
if (dynamic_cast<SIPAccount*>(Manager::instance().getAccount(accountID))->getSrtpFallback())
- call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initSession();
}
if (!sdpSession)