/**
* BigBlueButton open source conferencing system - http://www.bigbluebutton.org/
*
* Copyright (c) 2012 BigBlueButton Inc. and by respective authors (see below).
*
* This program is free software; you can redistribute it and/or modify it under the
* terms of the GNU Lesser General Public License as published by the Free Software
* Foundation; either version 3.0 of the License, or (at your option) any later
* version.
*
* BigBlueButton is distributed in the hope that it will be useful, but WITHOUT ANY
* WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A
* PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License along
* with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
*
*/
package org.bigbluebutton.voiceconf.sip;
import java.util.Collection;
import java.util.Iterator;
import org.zoolu.sip.provider.*;
import org.zoolu.net.SocketAddress;
import org.slf4j.Logger;
import org.bigbluebutton.voiceconf.messaging.IMessagingService;
import org.bigbluebutton.voiceconf.red5.CallStreamFactory;
import org.bigbluebutton.voiceconf.red5.ClientConnectionManager;
import org.red5.logging.Red5LoggerFactory;
import org.red5.server.api.scope.IScope;
import org.red5.server.api.stream.IBroadcastStream;
/**
* Class that is a peer to the sip server. This class will maintain
* all calls to it's peer server.
* @author Richard Alam
*
*/
public class SipPeer implements SipRegisterAgentListener {
private static Logger log = Red5LoggerFactory.getLogger(SipPeer.class, "sip");
private ClientConnectionManager clientConnManager;
private CallStreamFactory callStreamFactory;
private CallManager callManager = new CallManager();
private IMessagingService messagingService;
private SipProvider sipProvider;
private String clientRtpIp;
private SipRegisterAgent registerAgent;
private final String id;
private final AudioConferenceProvider audioconfProvider;
private boolean registered = false;
private SipPeerProfile registeredProfile;
public SipPeer(String id, String sipClientRtpIp, String host, int sipPort,
int startAudioPort, int stopAudioPort, IMessagingService messagingService) {
this.id = id;
this.clientRtpIp = sipClientRtpIp;
this.messagingService = messagingService;
audioconfProvider = new AudioConferenceProvider(host, sipPort, startAudioPort, stopAudioPort);
initSipProvider(host, sipPort);
}
private void initSipProvider(String host, int sipPort) {
sipProvider = new SipProvider(host, sipPort);
sipProvider.setOutboundProxy(new SocketAddress(host));
sipProvider.addSipProviderListener(new OptionMethodListener());
}
public void register(String username, String password) {
log.debug( "SIPUser register" );
createRegisterUserProfile(username, password);
if (sipProvider != null) {
registerAgent = new SipRegisterAgent(sipProvider, registeredProfile.fromUrl,
registeredProfile.contactUrl, registeredProfile.username,
registeredProfile.realm, registeredProfile.passwd);
registerAgent.addListener(this);
registerAgent.register(registeredProfile.expires, registeredProfile.expires/2, registeredProfile.keepaliveTime);
}
}
private void createRegisterUserProfile(String username, String password) {
registeredProfile = new SipPeerProfile();
registeredProfile.audioPort = audioconfProvider.getStartAudioPort();
String fromURL = "\"" + username + "\" <sip:" + username + "@" + audioconfProvider.getHost() + ">";
registeredProfile.username = username;
registeredProfile.passwd = password;
registeredProfile.realm = audioconfProvider.getHost();
registeredProfile.fromUrl = fromURL;
registeredProfile.contactUrl = "sip:" + username + "@" + sipProvider.getViaAddress();
if (sipProvider.getPort() != SipStack.default_port) {
registeredProfile.contactUrl += ":" + sipProvider.getPort();
}
registeredProfile.keepaliveTime=8000;
registeredProfile.acceptTime=0;
registeredProfile.hangupTime=20;
log.debug( "SIPUser register : {}", fromURL );
log.debug( "SIPUser register : {}", registeredProfile.contactUrl );
}
public void call(String clientId, String callerName, String destination) {
if (!registered) {
/*
* If we failed to register with FreeSWITCH, reject all calls right away.
* This way the user will know that there is a problem as quickly as possible.
* If we pass the call, it take more that 30seconds for the call to timeout
* (in case FS is offline) and the user will be kept wondering why the call
* isn't going through.
*/
log.warn("We are not registered to FreeSWITCH. However, we will allow {} to call {}.", callerName, destination);
// return;
}
CallAgent ca = createCallAgent(clientId);
ca.call(callerName, destination);
}
public void connectToGlobalStream(String clientId, String callerIdName, String destination) {
CallAgent ca = createCallAgent(clientId);
ca.connectToGlobalStream(clientId, callerIdName, destination);
}
private CallAgent createCallAgent(String clientId) {
SipPeerProfile callerProfile = SipPeerProfile.copy(registeredProfile);
CallAgent ca = new CallAgent(this.clientRtpIp, sipProvider, callerProfile, audioconfProvider, clientId, messagingService);
ca.setClientConnectionManager(clientConnManager);
ca.setCallStreamFactory(callStreamFactory);
callManager.add(ca);
return ca;
}
public void close() {
log.debug("SIPUser close1");
try {
unregister();
} catch(Exception e) {
log.error("close: Exception:>\n" + e);
}
log.debug("Stopping SipProvider");
sipProvider.halt();
}
public void hangup(String clientId) {
log.debug( "SIPUser hangup" );
CallAgent ca = callManager.remove(clientId);
if (ca != null) {
if (ca.isListeningToGlobal()) {
log.debug("User is in listen only mode.");
String destination = ca.getDestination();
ListenOnlyUser lou = GlobalCall.removeUser(clientId, destination);
if (lou != null) {
log.info("User has disconnected from global audio, user [{}] voiceConf {}", lou.callerIdName, lou.voiceConf);
messagingService.userDisconnectedFromGlobalAudio(lou.voiceConf, lou.callerIdName);
}
ca.hangup();
boolean roomRemoved = GlobalCall.removeRoomIfUnused(destination);
log.debug("Should the global connection be removed? {}", roomRemoved? "yes": "no");
if (roomRemoved) {
log.debug("Hanging up the global audio call {}", destination);
CallAgent caGlobal = callManager.remove(destination);
caGlobal.hangup();
}
} else {
ca.hangup();
}
}
}
public void unregister() {
log.debug( "SIPUser unregister" );
Collection<CallAgent> calls = callManager.getAll();
for (Iterator<CallAgent> iter = calls.iterator(); iter.hasNext();) {
CallAgent ca = (CallAgent) iter.next();
ca.hangup();
}
if (registerAgent != null) {
registerAgent.unregister();
registerAgent = null;
}
}
public void startTalkStream(String clientId, IBroadcastStream broadcastStream, IScope scope) {
CallAgent ca = callManager.get(clientId);
if (ca != null) {
ca.startTalkStream(broadcastStream, scope);
}
}
public void stopTalkStream(String clientId, IBroadcastStream broadcastStream, IScope scope) {
CallAgent ca = callManager.get(clientId);
if (ca != null) {
ca.stopTalkStream(broadcastStream, scope);
} else {
log.info("Can't stop talk stream as stream may have already been stopped.");
}
}
@Override
public void onRegistrationFailure(String result) {
log.error("Failed to register with Sip Server.");
registered = false;
}
@Override
public void onRegistrationSuccess(String result) {
log.info("Successfully registered with Sip Server.");
registered = true;
}
@Override
public void onUnregistedSuccess() {
log.info("Successfully unregistered with Sip Server");
registered = false;
}
public void setCallStreamFactory(CallStreamFactory csf) {
callStreamFactory = csf;
}
public void setClientConnectionManager(ClientConnectionManager ccm) {
clientConnManager = ccm;
}
}