Change summary
src/main/java/eu/siacs/conversations/xmpp/jingle/JingleRtpConnection.java | 10
1 file changed, 7 insertions(+), 3 deletions(-)
Detailed changes
@@ -119,7 +119,6 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
private RtpContentMap responderRtpContentMap;
private long rtpConnectionStarted = 0; //time of 'connected'
-
JingleRtpConnection(JingleConnectionManager jingleConnectionManager, Id id, Jid initiator) {
super(jingleConnectionManager, id, initiator);
final Conversation conversation = jingleConnectionManager.getXmppConnectionService().findOrCreateConversation(
@@ -847,6 +846,7 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
}
private void setupWebRTC(final Set<Media> media, final List<PeerConnection.IceServer> iceServers) throws WebRTCWrapper.InitializationException {
+ //TODO ensure registered with connection manager
final AppRTCAudioManager.SpeakerPhonePreference speakerPhonePreference;
if (media.contains(Media.VIDEO)) {
speakerPhonePreference = AppRTCAudioManager.SpeakerPhonePreference.SPEAKER;
@@ -950,13 +950,17 @@ public class JingleRtpConnection extends AbstractJingleConnection implements Web
Log.d(Config.LOGTAG, id.account.getJid().asBareJid() + ": not sending session-terminate after connectivity error because session is already in state " + this.state);
return;
}
- //we need to call close
- sendSessionTerminate(Reason.CONNECTIVITY_ERROR);
+ new Thread(this::closeWebRTCSessionAfterFailedConnection).start();
} else {
updateEndUserState();
}
}
+ private void closeWebRTCSessionAfterFailedConnection() {
+ this.webRTCWrapper.close();
+ sendSessionTerminate(Reason.CONNECTIVITY_ERROR);
+ }
+
public AppRTCAudioManager getAudioManager() {
return webRTCWrapper.getAudioManager();
}