[#1722] Uncoupled initAudioRtpConfig from initAudioRtpSession

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());

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__

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__);
@@ -3266,8 +3269,9 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
         // @TODO if SRTPONLY, CallFail
 
         // 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");