ensure webrtc connection gets closed after connection failure
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7dfd47a5c4
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@ -119,7 +119,6 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
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private RtpContentMap responderRtpContentMap;
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private long rtpConnectionStarted = 0; //time of 'connected'
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JingleRtpConnection(JingleConnectionManager jingleConnectionManager, Id id, Jid initiator) {
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super(jingleConnectionManager, id, initiator);
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final Conversation conversation = jingleConnectionManager.getXmppConnectionService().findOrCreateConversation(
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@ -847,6 +846,7 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
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}
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private void setupWebRTC(final Set<Media> media, final List<PeerConnection.IceServer> iceServers) throws WebRTCWrapper.InitializationException {
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//TODO ensure registered with connection manager
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final AppRTCAudioManager.SpeakerPhonePreference speakerPhonePreference;
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if (media.contains(Media.VIDEO)) {
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speakerPhonePreference = AppRTCAudioManager.SpeakerPhonePreference.SPEAKER;
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@ -950,13 +950,17 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
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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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}
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//we need to call close
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sendSessionTerminate(Reason.CONNECTIVITY_ERROR);
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new Thread(this::closeWebRTCSessionAfterFailedConnection).start();
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} else {
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updateEndUserState();
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}
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}
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private void closeWebRTCSessionAfterFailedConnection() {
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this.webRTCWrapper.close();
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sendSessionTerminate(Reason.CONNECTIVITY_ERROR);
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}
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public AppRTCAudioManager getAudioManager() {
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return webRTCWrapper.getAudioManager();
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}
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