diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index 48b3c3abfd8e448a582bd15ccf9f51939a072a63..3cf0fd3583a6b0eb62094842eb00d74e2f322589 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -54,10 +54,8 @@ AudioRtpFactory::~AudioRtpFactory()
stop();
}
-void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
+void AudioRtpFactory::initAudioRtpConfig(SIPCall *ca)
{
- ost::MutexLock m (_audioRtpThreadMutex);
-
assert (ca);
if (_rtpSession != NULL) {
@@ -68,36 +66,35 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
AccountID accountId = Manager::instance().getAccountFromCall (ca->getCallId());
- bool srtpEnabled = false;
- int keyExchangeProtocol = 1;
- bool helloHashEnabled = true;
-
// Check if it is an IP-to-IP call
-
if (accountId == AccountNULL) {
- srtpEnabled = Manager::instance().getConfigBool (IP2IP_PROFILE, SRTP_ENABLE);
- keyExchangeProtocol = Manager::instance().getConfigInt (IP2IP_PROFILE, SRTP_KEY_EXCHANGE);
- _debug ("Ip-to-ip profile selected with key exchange protocol number %d", keyExchangeProtocol);
- helloHashEnabled = Manager::instance().getConfigBool (IP2IP_PROFILE, ZRTP_HELLO_HASH);
+ _srtpEnabled = Manager::instance().getConfigBool (IP2IP_PROFILE, SRTP_ENABLE);
+ _keyExchangeProtocol = Manager::instance().getConfigInt (IP2IP_PROFILE, SRTP_KEY_EXCHANGE);
+ _debug ("Ip-to-ip profile selected with key exchange protocol number %d", _keyExchangeProtocol);
+ _helloHashEnabled = Manager::instance().getConfigBool (IP2IP_PROFILE, ZRTP_HELLO_HASH);
} else {
- srtpEnabled = Manager::instance().getConfigBool (accountId, SRTP_ENABLE);
- keyExchangeProtocol = Manager::instance().getConfigInt (accountId, SRTP_KEY_EXCHANGE);
- _debug ("Registered account %s profile selected with key exchange protocol number %d", accountId.c_str(), keyExchangeProtocol);
- helloHashEnabled = Manager::instance().getConfigBool (accountId, ZRTP_HELLO_HASH);
+ _srtpEnabled = Manager::instance().getConfigBool (accountId, SRTP_ENABLE);
+ _keyExchangeProtocol = Manager::instance().getConfigInt (accountId, SRTP_KEY_EXCHANGE);
+ _debug ("Registered account %s profile selected with key exchange protocol number %d", accountId.c_str(), _keyExchangeProtocol);
+ _helloHashEnabled = Manager::instance().getConfigBool (accountId, ZRTP_HELLO_HASH);
}
+}
- _debug ("Srtp enable: %d ", srtpEnabled);
+void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
+{
+ ost::MutexLock m (_audioRtpThreadMutex);
- if (srtpEnabled) {
+ _debug ("Srtp enable: %d ", _srtpEnabled);
+ if (_srtpEnabled) {
std::string zidFilename (Manager::instance().getConfigString (SIGNALISATION, ZRTP_ZIDFILE));
- switch (keyExchangeProtocol) {
+ switch (_keyExchangeProtocol) {
case Zrtp:
_rtpSession = new AudioZrtpSession (&Manager::instance(), ca, zidFilename);
_rtpSessionType = Zrtp;
- if (helloHashEnabled) {
+ if (_helloHashEnabled) {
// TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
// not even available at that point.
ca->getLocalSDP()->set_zrtp_hash (static_cast<AudioZrtpSession *> (_rtpSession)->getHelloHash());
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
index d965e53f4751c43e1f0e516290b446df87e13f61..cfa14165f1cf0f85960eb775b4b09fd476df7062 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
@@ -61,6 +61,8 @@ namespace sfl {
AudioRtpFactory(SIPCall * ca);
~AudioRtpFactory();
+ void initAudioRtpConfig(SIPCall *ca);
+
/**
* Lazy instantiation method. Create a new RTP session of a given
* type according to the content of the configuration file.
@@ -108,7 +110,22 @@ namespace sfl {
/**
* @param Set internal audio rtp session type (Symmetric, Zrtp, Sdes)
*/
- inline RtpMethod setAudioRtpType(RtpMethod type) { _rtpSessionType = type; }
+ inline void setAudioRtpType(RtpMethod type) { _rtpSessionType = type; }
+
+ /**
+ * Manually set the srtpEnable option (usefull for RTP fallback)
+ */
+ void setSrtpEnabled(bool enable){ _srtpEnabled = enable; }
+
+ /**
+ * Manually set the keyExchangeProtocol parameter (usefull for RTP fallback)
+ */
+ void setKeyExchangeProtocol(int proto){ _keyExchangeProtocol = proto; }
+
+ /**
+ * Manually set the setHelloHashEnabled parameter (usefull for RTP fallback)
+ */
+ void setHelloHashEnabled(bool enable){ _helloHashEnabled = enable; }
/**
* Get the current AudioZrtpSession. Throws an AudioRtpFactoryException
@@ -127,6 +144,18 @@ namespace sfl {
void * _rtpSession;
RtpMethod _rtpSessionType;
ost::Mutex _audioRtpThreadMutex;
+
+ // Field used when initializinga udio 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
+ int _keyExchangeProtocol;
+
+ // Field used when initializinga udio rtp session
+ // May be set manually or from config using initAudioRtpConfig
+ bool _helloHashEnabled;
};
}
#endif // __AUDIO_RTP_FACTORY_H__
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 2fac4d186e1f5239f299fdfdfb6bd73fb31ff5e5..ed81c182f92f36fac8117d3a55cfc43ff2e09bf3 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -757,6 +757,7 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
try {
_debug ("Creating new rtp session in newOutgoingCall");
+ call->getAudioRtp()->initAudioRtpConfig (call);
call->getAudioRtp()->initAudioRtpSession (call);
} catch (...) {
_debug ("Failed to create rtp thread from newOutGoingCall");
@@ -1060,6 +1061,7 @@ SIPVoIPLink::offhold (const CallID& id)
}
try {
+ call->getAudioRtp()->initAudioRtpConfig (call);
call->getAudioRtp()->initAudioRtpSession (call);
} catch (...) {
_debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
@@ -1569,6 +1571,7 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
// Audio Rtp Session must be initialized before creating initial offer in SDP session
// since SDES require crypto attribute.
try {
+ call->getAudioRtp()->initAudioRtpConfig (call);
call->getAudioRtp()->initAudioRtpSession (call);
} catch (...) {
_debug ("! SIP Failure: Unable to create RTP Session in SIPVoIPLink::new_ip_to_ip_call (%s:%d)", __FILE__, __LINE__);
@@ -3265,9 +3268,10 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
// We did not found any crypto context for this media
// @TODO if SRTPONLY, CallFail
- // if RTPFALLBACK, change RTP session
+ // if RTPFALLBACK, change RTP session
+ _debug("Did not found any crypto or negociation failed but Sdes enabled");
call->getAudioRtp()->stop();
- call->getAudioRtp()->setAudioRtpType(sfl::Symmetric);
+ call->getAudioRtp()->setSrtpEnabled(false);
call->getAudioRtp()->initAudioRtpSession(call);
}
@@ -3661,6 +3665,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
call->getLocalSDP()->set_ip_address (addrSdp);
try {
+ call->getAudioRtp()->initAudioRtpConfig (call);
call->getAudioRtp()->initAudioRtpSession (call);
} catch (...) {
_debug ("Failed to create rtp thread from answer");