do not automacially hang up failed webrtc sessions
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parent
4ec0996dff
commit
61851e5f84
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@ -96,7 +96,12 @@ public class RtpSessionActivity extends XmppActivity implements XmppConnectionSe
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);
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);
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private static final List<RtpEndUserState> STATES_SHOWING_SWITCH_TO_CHAT = Arrays.asList(
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private static final List<RtpEndUserState> STATES_SHOWING_SWITCH_TO_CHAT = Arrays.asList(
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RtpEndUserState.CONNECTING,
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RtpEndUserState.CONNECTING,
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RtpEndUserState.CONNECTED
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RtpEndUserState.CONNECTED,
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RtpEndUserState.RECONNECTING
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);
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private static final List<RtpEndUserState> STATES_CONSIDERED_CONNECTED = Arrays.asList(
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RtpEndUserState.CONNECTED,
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RtpEndUserState.RECONNECTING
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);
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);
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private static final String PROXIMITY_WAKE_LOCK_TAG = "conversations:in-rtp-session";
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private static final String PROXIMITY_WAKE_LOCK_TAG = "conversations:in-rtp-session";
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private static final int REQUEST_ACCEPT_CALL = 0x1111;
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private static final int REQUEST_ACCEPT_CALL = 0x1111;
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@ -502,7 +507,7 @@ public class RtpSessionActivity extends XmppActivity implements XmppConnectionSe
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private boolean isConnected() {
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private boolean isConnected() {
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final JingleRtpConnection connection = this.rtpConnectionReference != null ? this.rtpConnectionReference.get() : null;
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final JingleRtpConnection connection = this.rtpConnectionReference != null ? this.rtpConnectionReference.get() : null;
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return connection != null && connection.getEndUserState() == RtpEndUserState.CONNECTED;
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return connection != null && STATES_CONSIDERED_CONNECTED.contains(connection.getEndUserState());
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}
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}
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private boolean switchToPictureInPicture() {
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private boolean switchToPictureInPicture() {
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@ -661,6 +666,9 @@ public class RtpSessionActivity extends XmppActivity implements XmppConnectionSe
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case CONNECTED:
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case CONNECTED:
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setTitle(R.string.rtp_state_connected);
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setTitle(R.string.rtp_state_connected);
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break;
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break;
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case RECONNECTING:
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setTitle(R.string.rtp_state_reconnecting);
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break;
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case ACCEPTING_CALL:
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case ACCEPTING_CALL:
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setTitle(R.string.rtp_state_accepting_call);
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setTitle(R.string.rtp_state_accepting_call);
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break;
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break;
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@ -803,7 +811,7 @@ public class RtpSessionActivity extends XmppActivity implements XmppConnectionSe
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@SuppressLint("RestrictedApi")
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@SuppressLint("RestrictedApi")
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private void updateInCallButtonConfiguration(final RtpEndUserState state, final Set<Media> media) {
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private void updateInCallButtonConfiguration(final RtpEndUserState state, final Set<Media> media) {
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if (state == RtpEndUserState.CONNECTED && !isPictureInPicture()) {
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if (STATES_CONSIDERED_CONNECTED.contains(state) && !isPictureInPicture()) {
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Preconditions.checkArgument(media.size() > 0, "Media must not be empty");
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Preconditions.checkArgument(media.size() > 0, "Media must not be empty");
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if (media.contains(Media.VIDEO)) {
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if (media.contains(Media.VIDEO)) {
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final JingleRtpConnection rtpConnection = requireRtpConnection();
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final JingleRtpConnection rtpConnection = requireRtpConnection();
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@ -998,7 +1006,7 @@ public class RtpSessionActivity extends XmppActivity implements XmppConnectionSe
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RendererCommon.ScalingType.SCALE_ASPECT_FILL,
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RendererCommon.ScalingType.SCALE_ASPECT_FILL,
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RendererCommon.ScalingType.SCALE_ASPECT_FIT
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RendererCommon.ScalingType.SCALE_ASPECT_FIT
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);
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);
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if (state == RtpEndUserState.CONNECTED) {
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if (STATES_CONSIDERED_CONNECTED.contains(state)) {
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binding.appBarLayout.setVisibility(View.GONE);
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binding.appBarLayout.setVisibility(View.GONE);
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getWindow().addFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN);
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getWindow().addFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN);
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binding.remoteVideoWrapper.setVisibility(View.VISIBLE);
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binding.remoteVideoWrapper.setVisibility(View.VISIBLE);
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@ -1035,24 +1035,7 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
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return RtpEndUserState.CONNECTING;
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return RtpEndUserState.CONNECTING;
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}
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}
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case SESSION_ACCEPTED:
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case SESSION_ACCEPTED:
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//TODO refactor this out into separate method (that uses switch for better readability)
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return getPeerConnectionStateAsEndUserState();
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final PeerConnection.PeerConnectionState state;
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try {
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state = webRTCWrapper.getState();
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} catch (final WebRTCWrapper.PeerConnectionNotInitialized e) {
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//We usually close the WebRTCWrapper *before* transitioning so we might still
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//be in SESSION_ACCEPTED even though the peerConnection has been torn down
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return RtpEndUserState.ENDING_CALL;
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}
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if (state == PeerConnection.PeerConnectionState.CONNECTED) {
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return RtpEndUserState.CONNECTED;
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} else if (state == PeerConnection.PeerConnectionState.NEW || state == PeerConnection.PeerConnectionState.CONNECTING) {
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return RtpEndUserState.CONNECTING;
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} else if (state == PeerConnection.PeerConnectionState.CLOSED) {
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return RtpEndUserState.ENDING_CALL;
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} else {
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return rtpConnectionStarted == 0 ? RtpEndUserState.CONNECTIVITY_ERROR : RtpEndUserState.CONNECTIVITY_LOST_ERROR;
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}
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case REJECTED:
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case REJECTED:
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case REJECTED_RACED:
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case REJECTED_RACED:
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case TERMINATED_DECLINED_OR_BUSY:
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case TERMINATED_DECLINED_OR_BUSY:
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@ -1082,6 +1065,29 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
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throw new IllegalStateException(String.format("%s has no equivalent EndUserState", this.state));
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throw new IllegalStateException(String.format("%s has no equivalent EndUserState", this.state));
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}
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}
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private RtpEndUserState getPeerConnectionStateAsEndUserState() {
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final PeerConnection.PeerConnectionState state;
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try {
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state = webRTCWrapper.getState();
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} catch (final WebRTCWrapper.PeerConnectionNotInitialized e) {
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//We usually close the WebRTCWrapper *before* transitioning so we might still
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//be in SESSION_ACCEPTED even though the peerConnection has been torn down
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return RtpEndUserState.ENDING_CALL;
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}
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switch (state) {
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case CONNECTED:
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return RtpEndUserState.CONNECTED;
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case NEW:
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case CONNECTING:
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return RtpEndUserState.CONNECTING;
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case CLOSED:
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return RtpEndUserState.ENDING_CALL;
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default:
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return rtpConnectionStarted == 0 ? RtpEndUserState.CONNECTIVITY_ERROR : RtpEndUserState.RECONNECTING;
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}
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}
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public Set<Media> getMedia() {
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public Set<Media> getMedia() {
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final State current = getState();
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final State current = getState();
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if (current == State.NULL) {
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if (current == State.NULL) {
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@ -1331,29 +1337,31 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
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@Override
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@Override
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public void onConnectionChange(final PeerConnection.PeerConnectionState oldState, final PeerConnection.PeerConnectionState newState) {
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public void onConnectionChange(final PeerConnection.PeerConnectionState oldState, final PeerConnection.PeerConnectionState newState) {
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Log.d(Config.LOGTAG, id.account.getJid().asBareJid() + ": PeerConnectionState changed: "+oldState+"->" + newState);
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Log.d(Config.LOGTAG, id.account.getJid().asBareJid() + ": PeerConnectionState changed: " + oldState + "->" + newState);
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final boolean neverConnected = this.rtpConnectionStarted == 0;
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if (newState == PeerConnection.PeerConnectionState.CONNECTED && this.rtpConnectionStarted == 0) {
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if (newState == PeerConnection.PeerConnectionState.CONNECTED && this.rtpConnectionStarted == 0) {
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this.rtpConnectionStarted = SystemClock.elapsedRealtime();
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this.rtpConnectionStarted = SystemClock.elapsedRealtime();
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}
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}
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if (newState == PeerConnection.PeerConnectionState.CLOSED && this.rtpConnectionEnded == 0) {
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if (newState == PeerConnection.PeerConnectionState.CLOSED && this.rtpConnectionEnded == 0) {
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this.rtpConnectionEnded = SystemClock.elapsedRealtime();
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this.rtpConnectionEnded = SystemClock.elapsedRealtime();
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}
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}
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//TODO 'failed' means we need to restart ICE
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//
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if (neverConnected && Arrays.asList(PeerConnection.PeerConnectionState.FAILED, PeerConnection.PeerConnectionState.DISCONNECTED).contains(newState)) {
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//TODO 'disconnected' can probably be ignored as "This is a less stringent test than failed
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// and may trigger intermittently and resolve just as spontaneously on less reliable networks,
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// or during temporary disconnections. When the problem resolves, the connection may return
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// to the connected state."
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// Obviously the UI needs to reflect this new state with a 'reconnecting' display or something
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if (Arrays.asList(PeerConnection.PeerConnectionState.FAILED, PeerConnection.PeerConnectionState.DISCONNECTED).contains(newState)) {
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if (isTerminated()) {
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if (isTerminated()) {
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Log.d(Config.LOGTAG, id.account.getJid().asBareJid() + ": not sending session-terminate after connectivity error because session is already in state " + this.state);
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Log.d(Config.LOGTAG, id.account.getJid().asBareJid() + ": not sending session-terminate after connectivity error because session is already in state " + this.state);
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return;
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return;
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}
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}
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new Thread(this::closeWebRTCSessionAfterFailedConnection).start();
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new Thread(this::closeWebRTCSessionAfterFailedConnection).start();
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} else {
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} else if (newState == PeerConnection.PeerConnectionState.FAILED) {
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updateEndUserState();
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Log.d(Config.LOGTAG, "attempting to restart ICE");
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webRTCWrapper.restartIce();
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}
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}
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updateEndUserState();
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}
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@Override
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public void onRenegotiationNeeded() {
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Log.d(Config.LOGTAG, "onRenegotiationNeeded()");
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}
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}
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private void closeWebRTCSessionAfterFailedConnection() {
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private void closeWebRTCSessionAfterFailedConnection() {
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@ -4,6 +4,7 @@ public enum RtpEndUserState {
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INCOMING_CALL, //received a 'propose' message
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INCOMING_CALL, //received a 'propose' message
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CONNECTING, //session-initiate or session-accepted but no webrtc peer connection yet
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CONNECTING, //session-initiate or session-accepted but no webrtc peer connection yet
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CONNECTED, //session-accepted and webrtc peer connection is connected
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CONNECTED, //session-accepted and webrtc peer connection is connected
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RECONNECTING, //session-accepted and webrtc peer connection was connected once but is currently disconnected or failed
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FINDING_DEVICE, //'propose' has been sent out; no 184 ack yet
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FINDING_DEVICE, //'propose' has been sent out; no 184 ack yet
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RINGING, //'propose' has been sent out and it has been 184 acked
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RINGING, //'propose' has been sent out and it has been 184 acked
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ACCEPTING_CALL, //'proceed' message has been sent; but no session-initiate has been received
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ACCEPTING_CALL, //'proceed' message has been sent; but no session-initiate has been received
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@ -100,13 +100,14 @@ public class WebRTCWrapper {
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@Override
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@Override
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public void onConnectionChange(final PeerConnection.PeerConnectionState newState) {
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public void onConnectionChange(final PeerConnection.PeerConnectionState newState) {
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eventCallback.onConnectionChange(currentState, newState);
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final PeerConnection.PeerConnectionState oldState = currentState;
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currentState = newState;
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currentState = newState;
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eventCallback.onConnectionChange(oldState, newState);
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}
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}
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@Override
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@Override
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public void onIceConnectionChange(PeerConnection.IceConnectionState iceConnectionState) {
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public void onIceConnectionChange(PeerConnection.IceConnectionState iceConnectionState) {
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Log.d(EXTENDED_LOGGING_TAG, "onIceConnectionChange(" + iceConnectionState + ")");
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}
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}
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@Override
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@Override
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@ -152,7 +153,10 @@ public class WebRTCWrapper {
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@Override
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@Override
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public void onRenegotiationNeeded() {
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public void onRenegotiationNeeded() {
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Log.d(EXTENDED_LOGGING_TAG,"onRenegotiationNeeded - current state: "+currentState);
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Log.d(EXTENDED_LOGGING_TAG, "onRenegotiationNeeded()");
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if (currentState != null && currentState != PeerConnection.PeerConnectionState.NEW) {
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eventCallback.onRenegotiationNeeded();
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}
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}
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}
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@Override
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@Override
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@ -293,6 +297,10 @@ public class WebRTCWrapper {
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this.peerConnection = peerConnection;
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this.peerConnection = peerConnection;
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}
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}
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void restartIce() {
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requirePeerConnection().restartIce();
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}
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synchronized void close() {
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synchronized void close() {
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final PeerConnection peerConnection = this.peerConnection;
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final PeerConnection peerConnection = this.peerConnection;
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final CapturerChoice capturerChoice = this.capturerChoice;
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final CapturerChoice capturerChoice = this.capturerChoice;
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@ -538,6 +546,8 @@ public class WebRTCWrapper {
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void onConnectionChange(PeerConnection.PeerConnectionState oldState, PeerConnection.PeerConnectionState newState);
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void onConnectionChange(PeerConnection.PeerConnectionState oldState, PeerConnection.PeerConnectionState newState);
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void onAudioDeviceChanged(AppRTCAudioManager.AudioDevice selectedAudioDevice, Set<AppRTCAudioManager.AudioDevice> availableAudioDevices);
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void onAudioDeviceChanged(AppRTCAudioManager.AudioDevice selectedAudioDevice, Set<AppRTCAudioManager.AudioDevice> availableAudioDevices);
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void onRenegotiationNeeded();
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}
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}
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private static abstract class SetSdpObserver implements SdpObserver {
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private static abstract class SetSdpObserver implements SdpObserver {
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@ -904,6 +904,7 @@
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<string name="rtp_state_incoming_video_call">Incoming video call</string>
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<string name="rtp_state_incoming_video_call">Incoming video call</string>
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<string name="rtp_state_connecting">Connecting</string>
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<string name="rtp_state_connecting">Connecting</string>
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<string name="rtp_state_connected">Connected</string>
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<string name="rtp_state_connected">Connected</string>
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<string name="rtp_state_reconnecting">Reconnecting</string>
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<string name="rtp_state_accepting_call">Accepting call</string>
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<string name="rtp_state_accepting_call">Accepting call</string>
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<string name="rtp_state_ending_call">Ending call</string>
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<string name="rtp_state_ending_call">Ending call</string>
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<string name="answer_call">Answer</string>
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<string name="answer_call">Answer</string>
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