Touch up parcel voice enable/disable.

3 files changed, 261 insertions, 422 deletions

diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index e55d94d128..9b3ec2889b 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -86,7 +86,9 @@ void LLWebRTCImpl::init()
     mPlayoutDevice  = 0;
     mRecordingDevice = 0;
     rtc::InitializeSSL();
-    rtc::LogMessage::LogToDebug(rtc::LS_WARNING);
+    rtc::LogMessage::LogToDebug(rtc::LS_NONE);
+    rtc::LogMessage::SetLogToStderr(true);
+
     mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
     
     mNetworkThread = rtc::Thread::CreateWithSocketServer();
@@ -98,7 +100,7 @@ void LLWebRTCImpl::init()
     mSignalingThread = rtc::Thread::Create();
     mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
     mSignalingThread->Start();
-    
+
     mWorkerThread->PostTask(
                             [this]()
                             {
@@ -466,93 +468,98 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection()
     RTC_DCHECK(!mPeerConnection);
     mAnswerReceived = false;
 
-    webrtc::PeerConnectionInterface::RTCConfiguration config;
-    config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-    webrtc::PeerConnectionInterface::IceServer server;
-    server.uri = "stun:roxie-turn.staging.secondlife.io:3478";
-    config.servers.push_back(server);
-    server.uri = "stun:stun.l.google.com:19302";
-    config.servers.push_back(server);
-    server.uri = "stun:stun1.l.google.com:19302";
-    config.servers.push_back(server);
-    server.uri = "stun:stun2.l.google.com:19302";
-    config.servers.push_back(server);
-    server.uri = "stun:stun3.l.google.com:19302";
-    config.servers.push_back(server);
-    server.uri = "stun:stun4.l.google.com:19302";
-    config.servers.push_back(server);
-    
-    webrtc::PeerConnectionDependencies pc_dependencies(this);
-    auto error_or_peer_connection = mPeerConnectionFactory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
-    if (error_or_peer_connection.ok())
-    {
-        mPeerConnection = std::move(error_or_peer_connection.value());
-    }
-    else
-    {
-        RTC_LOG(LS_ERROR) << __FUNCTION__ << "Error creating peer connection: " << error_or_peer_connection.error().message();
-        return false;
-    }
-    
-    webrtc::DataChannelInit init;
-    init.ordered = true;
-    
-    auto data_channel_or_error = mPeerConnection->CreateDataChannelOrError("SLData", &init);
-    if (data_channel_or_error.ok())
-    {
-        mDataChannel = std::move(data_channel_or_error.value());
-        
-        mDataChannel->RegisterObserver(this);
-    }
-    
-    RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
-    
-    cricket::AudioOptions audioOptions;
-    audioOptions.auto_gain_control = true;
-    audioOptions.echo_cancellation = false;  // incompatible with opus stereo
-    audioOptions.noise_suppression = true;
-    
-    mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
-    rtc::scoped_refptr<webrtc::AudioTrackInterface>  audio_track(
-                                                                 mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
-    audio_track->set_enabled(true);
-    mLocalStream->AddTrack(audio_track);
-    
-    mPeerConnection->AddTrack(audio_track, {"SLStream"});
-    
-    auto senders = mPeerConnection->GetSenders();
-    
-    for (auto &sender : senders)
-    {
-        webrtc::RtpParameters      params;
-        webrtc::RtpCodecParameters codecparam;
-        codecparam.name                       = "opus";
-        codecparam.kind                       = cricket::MEDIA_TYPE_AUDIO;
-        codecparam.clock_rate                 = 48000;
-        codecparam.num_channels               = 2;
-        codecparam.parameters["stereo"]       = "1";
-        codecparam.parameters["sprop-stereo"] = "1";
-        params.codecs.push_back(codecparam);
-        sender->SetParameters(params);
-    }
-    
-    auto receivers = mPeerConnection->GetReceivers();
-    for (auto &receiver : receivers)
-    {
-        webrtc::RtpParameters      params;
-        webrtc::RtpCodecParameters codecparam;
-        codecparam.name                       = "opus";
-        codecparam.kind                       = cricket::MEDIA_TYPE_AUDIO;
-        codecparam.clock_rate                 = 48000;
-        codecparam.num_channels               = 2;
-        codecparam.parameters["stereo"]       = "1";
-        codecparam.parameters["sprop-stereo"] = "1";
-        params.codecs.push_back(codecparam);
-        receiver->SetParameters(params);
-    }
-    
-    webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offerOptions;
-    mPeerConnection->CreateOffer(this, offerOptions);
+    mWebRTCImpl->PostSignalingTask(
+        [this]()
+        {
+            webrtc::PeerConnectionInterface::RTCConfiguration config;
+            config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+            webrtc::PeerConnectionInterface::IceServer server;
+            server.uri = "stun:roxie-turn.staging.secondlife.io:3478";
+            config.servers.push_back(server);
+            server.uri = "stun:stun.l.google.com:19302";
+            config.servers.push_back(server);
+            server.uri = "stun:stun1.l.google.com:19302";
+            config.servers.push_back(server);
+            server.uri = "stun:stun2.l.google.com:19302";
+            config.servers.push_back(server);
+            server.uri = "stun:stun3.l.google.com:19302";
+            config.servers.push_back(server);
+            server.uri = "stun:stun4.l.google.com:19302";
+            config.servers.push_back(server);
+
+            config.set_min_port(60000);
+            config.set_max_port(60100);
+
+            webrtc::PeerConnectionDependencies pc_dependencies(this);
+            auto error_or_peer_connection = mPeerConnectionFactory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
+            if (error_or_peer_connection.ok())
+            {
+                mPeerConnection = std::move(error_or_peer_connection.value());
+            }
+            else
+            {
+                RTC_LOG(LS_ERROR) << __FUNCTION__ << "Error creating peer connection: " << error_or_peer_connection.error().message();
+                return;
+            }
+
+            webrtc::DataChannelInit init;
+            init.ordered = true;
+
+            auto data_channel_or_error = mPeerConnection->CreateDataChannelOrError("SLData", &init);
+            if (data_channel_or_error.ok())
+            {
+                mDataChannel = std::move(data_channel_or_error.value());
+
+                mDataChannel->RegisterObserver(this);
+            }
+
+            cricket::AudioOptions audioOptions;
+            audioOptions.auto_gain_control = true;
+            audioOptions.echo_cancellation = false;  // incompatible with opus stereo
+            audioOptions.noise_suppression = true;
+
+            mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
+            rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+                mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
+            audio_track->set_enabled(true);
+            mLocalStream->AddTrack(audio_track);
+
+            mPeerConnection->AddTrack(audio_track, {"SLStream"});
+
+            auto senders = mPeerConnection->GetSenders();
+
+            for (auto &sender : senders)
+            {
+                webrtc::RtpParameters      params;
+                webrtc::RtpCodecParameters codecparam;
+                codecparam.name                       = "opus";
+                codecparam.kind                       = cricket::MEDIA_TYPE_AUDIO;
+                codecparam.clock_rate                 = 48000;
+                codecparam.num_channels               = 2;
+                codecparam.parameters["stereo"]       = "1";
+                codecparam.parameters["sprop-stereo"] = "1";
+                params.codecs.push_back(codecparam);
+                sender->SetParameters(params);
+            }
+
+            auto receivers = mPeerConnection->GetReceivers();
+            for (auto &receiver : receivers)
+            {
+                webrtc::RtpParameters      params;
+                webrtc::RtpCodecParameters codecparam;
+                codecparam.name                       = "opus";
+                codecparam.kind                       = cricket::MEDIA_TYPE_AUDIO;
+                codecparam.clock_rate                 = 48000;
+                codecparam.num_channels               = 2;
+                codecparam.parameters["stereo"]       = "1";
+                codecparam.parameters["sprop-stereo"] = "1";
+                params.codecs.push_back(codecparam);
+                receiver->SetParameters(params);
+            }
+
+            webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offerOptions;
+            mPeerConnection->CreateOffer(this, offerOptions);
+        });
     
     return true;
 }
@@ -839,7 +846,7 @@ void LLWebRTCPeerConnectionImpl::OnIceCandidate(const webrtc::IceCandidateInterf
     else
     {
         mCachedIceCandidates.push_back(
-                                       webrtc::CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(), candidate->candidate()));
+            webrtc::CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(), candidate->candidate()));
     }
 }
 
@@ -878,21 +885,16 @@ void LLWebRTCPeerConnectionImpl::OnSuccess(webrtc::SessionDescriptionInterface *
             sdp_mangled_stream << sdp_line << "\n";
         }
     }
-    
-    webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str());
-    
-    
-    
-    mPeerConnection->SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>(
-                                                                                              webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str())),
-                                         rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
+
     RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_mangled_stream.str();
-    
-    
+
     for (auto &observer : mSignalingObserverList)
     {
         observer->OnOfferAvailable(sdp_mangled_stream.str());
     }
+
+    mPeerConnection->SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>(webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str())),
+                                         rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
 }
 
 void LLWebRTCPeerConnectionImpl::OnFailure(webrtc::RTCError error) { RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); }
@@ -930,7 +932,6 @@ void LLWebRTCPeerConnectionImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError
 //
 void LLWebRTCPeerConnectionImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error)
 {
-    
 }
 
 //
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
index ec1e6a353d..9a1b3525d7 100644
--- a/indra/newview/llvoicewebrtc.cpp
+++ b/indra/newview/llvoicewebrtc.cpp
@@ -455,17 +455,29 @@ void LLWebRTCVoiceClient::idle(void* user_data)
 // 
 // 
 
-void LLWebRTCVoiceClient::predProcessSessionStates(const LLWebRTCVoiceClient::sessionStatePtr_t& session)
+void LLWebRTCVoiceClient::sessionState::processSessionStates()
 {
-    session->processSessionStates();
+    auto iter = mSessions.begin();
+    while (iter != mSessions.end())
+    {
+        if (!iter->second->processConnectionStates())
+        {
+            iter = mSessions.erase(iter);
+        }
+        else
+        {
+            iter++;
+        }
+    }
 }
 
-void LLWebRTCVoiceClient::sessionState::processSessionStates()
+bool LLWebRTCVoiceClient::sessionState::processConnectionStates()
 {
-    std::map<std::string, connectionPtr_t>::iterator iter;
-    for (iter = mWebRTCConnections.begin(); iter != mWebRTCConnections.end();)
+    std::list<connectionPtr_t>::iterator iter = mWebRTCConnections.begin();
+
+    while (iter != mWebRTCConnections.end())
     {
-        if (!iter->second->connectionStateMachine())
+        if (!iter->get()->connectionStateMachine())
         {
             iter = mWebRTCConnections.erase(iter);
         }
@@ -474,6 +486,7 @@ void LLWebRTCVoiceClient::sessionState::processSessionStates()
             ++iter;
         }
     }
+    return !mWebRTCConnections.empty();
 }
 
 void LLWebRTCVoiceClient::voiceConnectionCoro()
@@ -488,33 +501,39 @@ void LLWebRTCVoiceClient::voiceConnectionCoro()
             llcoro::suspendUntilTimeout(UPDATE_THROTTLE_SECONDS);
             // add session for region or parcel voice.
             LLViewerRegion *regionp = gAgent.getRegion();
-            if (!regionp)
+            if (!regionp || regionp->getRegionID().isNull())
             {
                 continue;
             }
-            if (regionp->getRegionID().isNull())
-            {
-                continue;
-            }
-            if (mVoiceEnabled && (!mAudioSession || mAudioSession->mIsSpatial))
+
+            if (mVoiceEnabled && (!mAudioSession || mAudioSession->isSpatial()) && !mNextAudioSession)
             {
                 // check to see if parcel changed.
                 std::string channelID = regionp->getRegionID().asString();
 
                 LLParcel *parcel = LLViewerParcelMgr::getInstance()->getAgentParcel();
                 S32       parcel_local_id = INVALID_PARCEL_ID;
-                if (parcel && parcel->getLocalID() != INVALID_PARCEL_ID && !parcel->getParcelFlagUseEstateVoiceChannel())
+                if (parcel && parcel->getLocalID() != INVALID_PARCEL_ID)
                 {
-                    parcel_local_id = parcel->getLocalID();
-                    channelID += "-" + std::to_string(parcel->getLocalID());
+                    if (!parcel->getParcelFlagAllowVoice())
+                    {
+                        channelID.clear();
+                    }
+                    else if (!parcel->getParcelFlagUseEstateVoiceChannel())
+                    {
+                        parcel_local_id = parcel->getLocalID();
+                        channelID += "-" + std::to_string(parcel->getLocalID());
+                    }
                 }
-                if (!mAudioSession || channelID != mAudioSession->mChannelID)
+                
+                if ((mNextAudioSession && channelID != mNextAudioSession->mChannelID) ||
+                    (!mAudioSession && !channelID.empty()) || 
+                    (mAudioSession && channelID != mAudioSession->mChannelID))
                 {
                     setSpatialChannel(channelID, "", parcel_local_id);
                 }
             }
-            sessionState::for_each(boost::bind(predProcessSessionStates, _1));
-            reapEmptySessions();
+            sessionState::processSessionStates();
             if (mVoiceEnabled)
             {
                 updatePosition();
@@ -829,7 +848,7 @@ void LLWebRTCVoiceClient::predUpdateOwnVolume(const LLWebRTCVoiceClient::session
 
 void LLWebRTCVoiceClient::predSendData(const LLWebRTCVoiceClient::sessionStatePtr_t& session, const std::string& spatial_data, const std::string& volume_data)
 {
-    if (session->mIsSpatial && !spatial_data.empty())
+    if (session->isSpatial() && !spatial_data.empty())
     {
         session->sendData(spatial_data);
     }
@@ -843,7 +862,7 @@ void LLWebRTCVoiceClient::sessionState::sendData(const std::string &data)
 {
     for (auto& connection : mWebRTCConnections)
     {
-        connection.second->sendData(data);
+        connection->sendData(data);
     }
 }
 
@@ -852,7 +871,7 @@ void LLWebRTCVoiceClient::sessionState::setMuteMic(bool muted)
     mMuted = muted;
     for (auto& connection : mWebRTCConnections)
     {
-        connection.second->setMuteMic(muted);
+        connection->setMuteMic(muted);
     }
 }
 
@@ -861,7 +880,7 @@ void LLWebRTCVoiceClient::sessionState::setMicGain(F32 gain)
     mMicGain = gain;
     for (auto& connection : mWebRTCConnections)
     {
-        connection.second->setMicGain(gain);
+        connection->setMicGain(gain);
     }
 }
 
@@ -870,7 +889,7 @@ void LLWebRTCVoiceClient::sessionState::setSpeakerVolume(F32 volume)
     mSpeakerVolume = volume;
     for (auto& connection : mWebRTCConnections)
     {
-        connection.second->setSpeakerVolume(volume);
+        connection->setSpeakerVolume(volume);
     }
 }
 
@@ -878,52 +897,6 @@ void LLWebRTCVoiceClient::sendLocalAudioUpdates()
 {
 }
 
-
-void LLWebRTCVoiceClient::joinedAudioSession(const sessionStatePtr_t &session)
-{
-	LL_DEBUGS("Voice") << "Joined Audio Session" << LL_ENDL;
-	if(mAudioSession != session)
-	{
-        sessionStatePtr_t oldSession = mAudioSession;
-
-		mAudioSession = session;
-		mAudioSessionChanged = true;
-
-		// The old session may now need to be deleted.
-		reapSession(oldSession);
-	}
-	
-	// This is the session we're joining.
-	if(mIsJoiningSession)
-	{
-        LLSD WebRTCevent(LLSDMap("channel", session->mChannelID)
-                ("session", "joined"));
-
-        mWebRTCPump.post(WebRTCevent);
-		
-		if(!session->mIsChannel)
-		{
-			// this is a p2p session.  Make sure the other end is added as a participant.
-            participantStatePtr_t participant(session->addParticipant(LLUUID(session->mChannelID)));
-			if(participant)
-			{
-				if(participant->mAvatarIDValid)
-				{
-					lookupName(participant->mAvatarID);
-				}
-				else if(!session->mName.empty())
-				{
-					participant->mDisplayName = session->mName;
-					avatarNameResolved(participant->mAvatarID, session->mName);
-				}
-				
-				// TODO: Question: Do we need to set up mAvatarID/mAvatarIDValid here?
-				LL_INFOS("Voice") << "added caller as participant  (" << participant->mAvatarID << ")"<< LL_ENDL;
-			}
-		}
-	}
-}
-
 void LLWebRTCVoiceClient::reapSession(const sessionStatePtr_t &session)
 {
 	if(session)
@@ -950,19 +923,6 @@ void LLWebRTCVoiceClient::reapSession(const sessionStatePtr_t &session)
 }
 
 
-void LLWebRTCVoiceClient::leftAudioSession(const sessionStatePtr_t &session)
-{
-    if (mAudioSession == session)
-    {
-        LLSD WebRTCevent(LLSDMap("channel", session->mChannelID)
-            ("session", "removed"));
-
-        mWebRTCPump.post(WebRTCevent);
-    }
-}
-
-
-
 void LLWebRTCVoiceClient::muteListChanged()
 {
 	// The user's mute list has been updated.  Go through the current participant list and sync it with the mute list.
@@ -1215,149 +1175,47 @@ void LLWebRTCVoiceClient::removeParticipantByID(const std::string &channelID, co
     }
 }
 
-
-// Check for parcel boundary crossing
-bool LLWebRTCVoiceClient::checkParcelChanged(bool update)
-{
-	LLViewerRegion *region = gAgent.getRegion();
-	LLParcel *parcel = LLViewerParcelMgr::getInstance()->getAgentParcel();
-	
-	if(region && parcel)
-	{
-		S32 parcelLocalID = parcel->getLocalID();
-		std::string regionName = region->getName();
-		
-		//			LL_DEBUGS("Voice") << "Region name = \"" << regionName << "\", parcel local ID = " << parcelLocalID << ", cap URI = \"" << capURI << "\"" << LL_ENDL;
-		
-		// The region name starts out empty and gets filled in later.  
-		// Also, the cap gets filled in a short time after the region cross, but a little too late for our purposes.
-		// If either is empty, wait for the next time around.
-		if(!regionName.empty())
-		{
-			if((parcelLocalID != mCurrentParcelLocalID) || (regionName != mCurrentRegionName))
-			{
-				// We have changed parcels.  Initiate a parcel channel lookup.
-				if (update)
-				{
-					mCurrentParcelLocalID = parcelLocalID;
-					mCurrentRegionName = regionName;
-				}
-				return true;
-			}
-		}
-	}
-	return false;
-}
-
-bool LLWebRTCVoiceClient::switchChannel(
-	const std::string channelID,
-	bool spatial,
-	bool no_reconnect,
-	bool is_p2p,
-	std::string hash,
-    S32 parcel_local_id)
+bool LLWebRTCVoiceClient::switchChannel(const std::string          channelID,
+                                        sessionState::ESessionType session_type,
+                                        S32                        parcel_local_id)
 {
-	bool needsSwitch = false;
-
     if (mAudioSession)
     {
-        if (mSessionTerminateRequested)
+        // If we're already in a channel, or if we're joining one, terminate
+        // so we can rejoin with the new session data.
+        sessionTerminate();
+        notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL);
+        deleteSession(mAudioSession);
+    }
+
+    if (channelID.empty())
+    {
+        // Leave any channel we may be in
+        LL_DEBUGS("Voice") << "leaving channel" << LL_ENDL;
+
+        if (mNextAudioSession)
         {
-            // If a terminate has been requested, we need to compare against where the URI we're already headed to.
-            if(mNextAudioSession)
-            {
-                if (mNextAudioSession->mChannelID != channelID)
-                    needsSwitch = true;
-            }
-            else
-            {
-                // mNextAudioSession is null -- this probably means we're on our way back to spatial.
-                if (!channelID.empty())
-                {
-                    // We do want to process a switch in this case.
-                    needsSwitch = true;
-                }
-            }
+            deleteSession(mNextAudioSession);
         }
-        else
+        // If voice was on, turn it off
+        if (LLVoiceClient::getInstance()->getUserPTTState())
         {
-            // Otherwise, compare against the URI we're in now.
-            if(mAudioSession)
-            {
-                if (mAudioSession->mChannelID != channelID)
-                {
-                    needsSwitch = true;
-                }
-            }
-            else
-            {
-                if (!channelID.empty())
-                {
-                    // mAudioSession is null -- it's not clear what case would cause this.
-                    // For now, log it as a warning and see if it ever crops up.
-                    LL_WARNS("Voice") << "No current audio session... Forcing switch" << LL_ENDL;
-                    needsSwitch = true;
-                }
-            }
+            LLVoiceClient::getInstance()->setUserPTTState(false);
         }
+
+        notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED);
     }
     else
     {
-        if (!mNextAudioSession || mNextAudioSession->mChannelID != channelID)
+        if (mNextAudioSession)
         {
-            needsSwitch = true;
+            deleteSession(mNextAudioSession);
         }
+        LL_DEBUGS("Voice") << "switching to channel " << channelID << LL_ENDL;
+        mNextAudioSession             = addSession(channelID, parcel_local_id);
+        mNextAudioSession->mSessionType = session_type;
     }
-
-    if(needsSwitch)
-	{
-        if (mAudioSession)
-        {
-            // If we're already in a channel, or if we're joining one, terminate
-            // so we can rejoin with the new session data.
-            sessionTerminate();
-            notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL);
-            deleteSession(mAudioSession);
-        }
-        
-        if (channelID.empty())
-		{
-			// Leave any channel we may be in
-			LL_DEBUGS("Voice") << "leaving channel" << LL_ENDL;
-
-            if (mNextAudioSession)
-            {
-                mAudioSession->shutdownAllConnections();
-            }
-            sessionStatePtr_t oldSession = mNextAudioSession;
-			mNextAudioSession.reset();
-
-			// The old session may now need to be deleted.
-			reapSession(oldSession);
-
-            // If voice was on, turn it off
-            if (LLVoiceClient::getInstance()->getUserPTTState())
-            {
-                LLVoiceClient::getInstance()->setUserPTTState(false);
-            }
-
-			notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED);
-		}
-		else
-		{
-            if (mNextAudioSession)
-            {
-                deleteSession(mNextAudioSession);
-            }
-            LL_DEBUGS("Voice") << "switching to channel " << channelID << LL_ENDL;
-            mNextAudioSession             = addSession(channelID, parcel_local_id);
-            mNextAudioSession->mIsSpatial = spatial;
-            mNextAudioSession->mReconnect = !no_reconnect;
-            mNextAudioSession->mIsP2P     = is_p2p;
-		}
-	}
-
-    return needsSwitch;
+    return true;	
 }
 
 void LLWebRTCVoiceClient::joinSession(const sessionStatePtr_t &session)
@@ -1376,7 +1234,7 @@ void LLWebRTCVoiceClient::setNonSpatialChannel(
 	const std::string &uri,
 	const std::string &credentials)
 {
-	switchChannel(uri, false, false, false, credentials);
+	switchChannel(uri, sessionState::SESSION_TYPE_P2P);
 }
 
 bool LLWebRTCVoiceClient::setSpatialChannel(
@@ -1386,7 +1244,7 @@ bool LLWebRTCVoiceClient::setSpatialChannel(
 
 	LL_DEBUGS("Voice") << "got spatial channel uri: \"" << uri << "\"" << LL_ENDL;
 	
-	if((mAudioSession && !(mAudioSession->mIsSpatial)) || (mNextAudioSession && !(mNextAudioSession->mIsSpatial)))
+	if((mAudioSession && !mAudioSession->isSpatial()) || (mNextAudioSession && !mNextAudioSession->isSpatial()))
 	{
 		// User is in a non-spatial chat or joining a non-spatial chat.  Don't switch channels.
 		LL_INFOS("Voice") << "in non-spatial chat, not switching channels" << LL_ENDL;
@@ -1394,13 +1252,13 @@ bool LLWebRTCVoiceClient::setSpatialChannel(
 	}
 	else
 	{
-        return switchChannel(uri, true, false, false, "", parcel_local_id);
+        return switchChannel(uri, parcel_local_id == INVALID_PARCEL_ID ? sessionState::SESSION_TYPE_ESTATE : sessionState::SESSION_TYPE_PARCEL, parcel_local_id);
 	}
 }
 
 void LLWebRTCVoiceClient::callUser(const LLUUID &uuid)
 { 
-    switchChannel(uuid.asString(), false, true, true);
+    switchChannel(uuid.asString(), sessionState::SESSION_TYPE_P2P);
 }
 
 void LLWebRTCVoiceClient::endUserIMSession(const LLUUID &uuid)
@@ -1419,9 +1277,7 @@ bool LLWebRTCVoiceClient::answerInvite(std::string &channelID)
     sessionStatePtr_t session(findP2PSession(LLUUID(channelID)));
 	if(session)
 	{
-		session->mIsSpatial = false;
-		session->mReconnect = false;	
-		session->mIsP2P = true;
+        session->mSessionType = sessionState::ESessionType::SESSION_TYPE_P2P;
 
 		joinSession(session);
 		return true;
@@ -1604,7 +1460,7 @@ bool LLWebRTCVoiceClient::inSpatialChannel(void)
 	
 	if(mAudioSession)
     {
-		result = mAudioSession->mIsSpatial;
+		result = mAudioSession->isSpatial();
     }
     
 	return result;
@@ -2061,34 +1917,12 @@ BOOL LLWebRTCVoiceClient::getAreaVoiceDisabled()
 	return mAreaVoiceDisabled;
 }
 
-void LLWebRTCVoiceClient::reapEmptySessions()
-{
-    sessionState::reapEmptySessions();
-
-    // mAudioSession or mNextAudioSession was reaped,
-    // so reset them.
-    if (mAudioSession && !sessionState::hasSession(mAudioSession->mChannelID))
-    {
-        mAudioSession.reset();
-    }
-    if (mNextAudioSession && !sessionState::hasSession(mNextAudioSession->mChannelID))
-    {
-        mNextAudioSession.reset();
-    }
-}
-
 //------------------------------------------------------------------------
 std::map<std::string, LLWebRTCVoiceClient::sessionState::ptr_t> LLWebRTCVoiceClient::sessionState::mSessions;
 
 
 LLWebRTCVoiceClient::sessionState::sessionState() :
     mErrorStatusCode(0),
-    mIsChannel(false),
-    mIsSpatial(false),
-    mIsP2P(false),
-    mIncoming(false),
-    mVoiceActive(false),
-    mReconnect(false),
     mVolumeDirty(false),
     mMuteDirty(false),
     mParticipantsChanged(false)
@@ -2102,8 +1936,7 @@ LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::crea
 
     sessionState::ptr_t session(new sessionState());
     session->mChannelID = channelID;
-    connectionPtr_t connection = connectionPtr_t(new LLVoiceWebRTCConnection(region_id, parcelLocalID, channelID));
-    session->mWebRTCConnections[channelID] = connection;
+    session->mWebRTCConnections.emplace_back(new LLVoiceWebRTCConnection(region_id, parcelLocalID, channelID));
     
     session->mPrimaryConnectionID          = channelID;
 
@@ -2202,7 +2035,7 @@ void LLWebRTCVoiceClient::sessionState::for_eachPredicate(const std::pair<std::s
 LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::findP2PSession(const LLUUID &agent_id)
 {
     sessionStatePtr_t result = sessionState::matchSessionByChannelID(agent_id.asString());
-    if (result && result->mIsP2P)
+    if (result && result->mSessionType == sessionState::SESSION_TYPE_P2P)
     {
         return result;
     }
@@ -2217,7 +2050,7 @@ void LLWebRTCVoiceClient::sessionState::shutdownAllConnections()
 { 
     for (auto &&connection : mWebRTCConnections)
     {
-        connection.second->shutDown();
+        connection->shutDown();
     }
 }
 
@@ -2277,14 +2110,16 @@ void LLWebRTCVoiceClient::deleteSession(const sessionStatePtr_t &session)
 	verifySessionState();
     session->shutdownAllConnections();
 	// If this is the current audio session, clean up the pointer which will soon be dangling.
-	if(mAudioSession == session)
+    bool deleteAudioSession = mAudioSession == session;
+    bool deleteNextAudioSession = mNextAudioSession == session;
+    if (deleteAudioSession)
 	{
 		mAudioSession.reset();
 		mAudioSessionChanged = true;
 	}
 
 	// ditto for the next audio session
-	if(mNextAudioSession == session)
+    if (deleteNextAudioSession)
 	{
 		mNextAudioSession.reset();
 	}
@@ -2696,73 +2531,68 @@ void LLVoiceWebRTCConnection::OnPeerShutDown()
 
 void LLVoiceWebRTCConnection::processIceUpdates()
 {
-    if (LLWebRTCVoiceClient::isShuttingDown())
-    {
-        return;
-    }
-    if (mShutDown)
-    {
-        return;
-    }
-    LLViewerRegion *regionp = LLWorld::instance().getRegionFromID(mRegionID);
-    if (!regionp || !regionp->capabilitiesReceived())
+    if (mShutDown || LLWebRTCVoiceClient::isShuttingDown())
     {
-        LL_DEBUGS("Voice") << "no capabilities for ice gathering; waiting " << LL_ENDL;
         return;
     }
 
-    std::string url = regionp->getCapability("VoiceSignalingRequest");
-    if (url.empty())
-    {
-        return;
-    }
-
-    LL_DEBUGS("Voice") << "region ready to complete voice signaling; url=" << url << LL_ENDL;
-
-    LLCore::HttpRequest::policy_t               httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
-    LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(new LLCoreHttpUtil::HttpCoroutineAdapter("voiceAccountProvision", httpPolicy));
-    LLCore::HttpRequest::ptr_t                  httpRequest(new LLCore::HttpRequest);
-    LLCore::HttpOptions::ptr_t                  httpOpts = LLCore::HttpOptions::ptr_t(new LLCore::HttpOptions);
-
     bool iceCompleted = false;
     LLSD body;
     {
         if (!mTrickling)
         {
-            if (mIceCandidates.size())
+            if (!mIceCandidates.empty() || mIceCompleted)
             {
-                LLSD candidates = LLSD::emptyArray();
-                for (auto &ice_candidate : mIceCandidates)
+                LLViewerRegion *regionp = LLWorld::instance().getRegionFromID(mRegionID);
+                if (!regionp || !regionp->capabilitiesReceived())
                 {
-                LLSD body_candidate;
-                body_candidate["sdpMid"]        = ice_candidate.sdp_mid;
-                body_candidate["sdpMLineIndex"] = ice_candidate.mline_index;
-                body_candidate["candidate"]     = ice_candidate.candidate;
-                candidates.append(body_candidate);
+                    LL_DEBUGS("Voice") << "no capabilities for ice gathering; waiting " << LL_ENDL;
+                    return;
                 }
-                body["candidates"] = candidates;
-                mIceCandidates.clear();
-            }
-            else if (mIceCompleted)
-            {
-                LLSD body_candidate;
-                body_candidate["completed"] = true;
-                body["candidate"]           = body_candidate;
-                iceCompleted                = mIceCompleted;
-                mIceCompleted               = false;
-            }
-            else
-            {
-                return;
+
+                std::string url = regionp->getCapability("VoiceSignalingRequest");
+                if (url.empty())
+                {
+                    return;
+                }
+
+                LL_DEBUGS("Voice") << "region ready to complete voice signaling; url=" << url << LL_ENDL;
+                if (!mIceCandidates.empty()) {
+                    LLSD candidates = LLSD::emptyArray();
+                    for (auto &ice_candidate : mIceCandidates)
+                    {
+                        LLSD body_candidate;
+                        body_candidate["sdpMid"]        = ice_candidate.sdp_mid;
+                        body_candidate["sdpMLineIndex"] = ice_candidate.mline_index;
+                        body_candidate["candidate"]     = ice_candidate.candidate;
+                        candidates.append(body_candidate);
+                    }
+                    body["candidates"] = candidates;
+                    mIceCandidates.clear();
+                }
+                else if (mIceCompleted)
+                {
+                    LLSD body_candidate;
+                    body_candidate["completed"] = true;
+                    body["candidate"]           = body_candidate;
+                    iceCompleted                = mIceCompleted;
+                    mIceCompleted               = false;
+                }
+
+                LLCore::HttpRequest::policy_t               httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+                LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(
+                    new LLCoreHttpUtil::HttpCoroutineAdapter("voiceAccountProvision", httpPolicy));
+                LLCore::HttpRequest::ptr_t httpRequest(new LLCore::HttpRequest);
+                LLCore::HttpOptions::ptr_t httpOpts = LLCore::HttpOptions::ptr_t(new LLCore::HttpOptions);
+                LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+                    url,
+                    LLCore::HttpRequest::DEFAULT_POLICY_ID,
+                    body,
+                    boost::bind(&LLVoiceWebRTCConnection::onIceUpdateComplete, this, iceCompleted, _1),
+                    boost::bind(&LLVoiceWebRTCConnection::onIceUpdateError, this, 3, url, body, iceCompleted, _1));
+                mOutstandingRequests++;
+                mTrickling = true;
             }
-            LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
-                url,
-                LLCore::HttpRequest::DEFAULT_POLICY_ID,
-                body,
-                boost::bind(&LLVoiceWebRTCConnection::onIceUpdateComplete, this, iceCompleted, _1),
-                boost::bind(&LLVoiceWebRTCConnection::onIceUpdateError, this, 3, url, body, iceCompleted, _1));
-            mOutstandingRequests++;
-            mTrickling = true;
         }
     }
 }
@@ -2862,7 +2692,7 @@ void LLVoiceWebRTCConnection::OnVoiceConnectionRequestFailure(std::string url, i
 }
 
 bool LLVoiceWebRTCConnection::connectionStateMachine()
-{   
+{  
     processIceUpdates();
 
     switch (getVoiceConnectionState())
@@ -2872,6 +2702,7 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
             if (mShutDown)
             {
                 setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+                break;
             }
             mTrickling    = false;
             mIceCompleted = false;
diff --git a/indra/newview/llvoicewebrtc.h b/indra/newview/llvoicewebrtc.h
index f311f241dc..1e8ecdf5c2 100644
--- a/indra/newview/llvoicewebrtc.h
+++ b/indra/newview/llvoicewebrtc.h
@@ -330,7 +330,9 @@ public:
 		bool isCallBackPossible();
 		bool isTextIMPossible();
 
-        void processSessionStates();
+		static void processSessionStates();
+
+        bool processConnectionStates();
 
 		void OnConnectionEstablished(const std::string &channelID);
         void OnConnectionFailure(const std::string &channelID);
@@ -347,6 +349,15 @@ public:
 
 		bool isEmpty() { return mWebRTCConnections.empty(); }
 
+		bool isSpatial() { return mSessionType == SESSION_TYPE_ESTATE || mSessionType == SESSION_TYPE_PARCEL; }
+
+    typedef enum e_session_type
+        {
+            SESSION_TYPE_ESTATE                = 1,
+            SESSION_TYPE_PARCEL,
+            SESSION_TYPE_P2P
+        } ESessionType;
+
 		std::string mHandle;
 		std::string mGroupHandle;
 		std::string mChannelID;
@@ -362,9 +373,8 @@ public:
 		LLUUID		mIMSessionID;
 		LLUUID		mCallerID;
 		int			mErrorStatusCode;
-		bool		mIsChannel;	// True for both group and spatial channels (false for p2p, PSTN)
-		bool		mIsSpatial;	// True for spatial channels
-		bool		mIsP2P;
+        ESessionType mSessionType;
+
 		bool		mIncoming;
 		bool		mVoiceActive;
 		bool		mReconnect;	// Whether we should try to reconnect to this session if it's dropped
@@ -381,16 +391,17 @@ public:
 		LLUUID		mVoiceFontID;
 
         static void VerifySessions();
-        static bool hasSession(const std::string &sessionID) { return mSessions.find(sessionID) != mSessions.end(); }
+        static bool hasSession(const std::string &sessionID) 
+		{ return mSessions.find(sessionID) != mSessions.end(); }
 
     private:
 
-		std::map<std::string, connectionPtr_t> mWebRTCConnections;
-        std::string   					       mPrimaryConnectionID;
+		std::list<connectionPtr_t> mWebRTCConnections;
+        std::string   			   mPrimaryConnectionID;
+        static std::map<std::string, ptr_t> mSessions;  // canonical list of outstanding sessions.
 
         sessionState();
 
-        static std::map<std::string, ptr_t> mSessions;  // canonical list of outstanding sessions.
 
         static void for_eachPredicate(const std::pair<std::string, LLWebRTCVoiceClient::sessionState::wptr_t> &a, sessionFunc_t func);
 
@@ -406,7 +417,6 @@ public:
 	// Private Member Functions
 	//////////////////////////////////////////////////////
 
-    static void predProcessSessionStates(const LLWebRTCVoiceClient::sessionStatePtr_t &session);
     static void predOnConnectionEstablished(const LLWebRTCVoiceClient::sessionStatePtr_t &session, std::string channelID);
     static void predOnConnectionFailure(const LLWebRTCVoiceClient::sessionStatePtr_t &session, std::string channelID);
     static void predSendData(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const std::string& spatial_data, const std::string& volume_data);
@@ -415,7 +425,6 @@ public:
     static void predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume);
     static void predSetSpeakerVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume);
     static void predShutdownSession(const LLWebRTCVoiceClient::sessionStatePtr_t &session);
-    void reapEmptySessions();
 
 	//////////////////////////////
 	/// @name TVC/Server management and communication
@@ -476,9 +485,6 @@ public:
 
 	void verifySessionState(void);
 
-    void joinedAudioSession(const sessionStatePtr_t &session);
-    void leftAudioSession(const sessionStatePtr_t &session);
-
 	// This is called in several places where the session _may_ need to be deleted.
 	// It contains logic for whether to delete the session or keep it around.
     void reapSession(const sessionStatePtr_t &session);	
@@ -596,12 +602,8 @@ private:
 	bool mIsInitialized;
 	bool mShutdownComplete;
 	
-	bool checkParcelChanged(bool update = false);
     bool switchChannel(const std::string channelID,
-                       bool         spatial      = true,
-                       bool         no_reconnect = false,
-                       bool         is_p2p       = false,
-                       std::string  hash         = "",
+                       sessionState::ESessionType sessionType,
 		               S32          parcel_local_id = INVALID_PARCEL_ID);
     void joinSession(const sessionStatePtr_t &session);
 	
@@ -808,6 +810,10 @@ protected:
     }
     EVoiceConnectionState getVoiceConnectionState()
     {
+		if (mVoiceStateMutex.isLocked())
+		{
+            LL_WARNS("Voice") << "LOCKED." << LL_ENDL;
+		}
         LLMutexLock lock(&mVoiceStateMutex);
         return mVoiceConnectionState;
     }
@@ -839,6 +845,7 @@ protected:
     llwebrtc::LLWebRTCDataInterface  *mWebRTCDataInterface;
 };
 
+#define VOICE_ELAPSED LLVoiceTimer(__FUNCTION__);
 
 #endif //LL_WebRTC_VOICE_CLIENT_H