android 5.0 呼叫流程--呼出,Android 5.0 呼叫流程--SIP呼出

4.2.1.5.0 SIP呼叫MO流程java

SIP MO呼叫流程以下图所示(为了便于看清，分两段截图)：android

3.6.2.1.SIP呼叫App层处理流程session

SIP呼叫在App层的处理流程和普通呼叫流程同样，只是号码uri的参数scheme不同。ide

3.6.2.2.SIP呼叫telecom层处理流程this

在telecom层，SIP呼叫和普通呼叫不一样的地方开始于ConnectionService。spa

在ConnectionService服务端，onCreateOutgoingConnection会被调用到，这个方法被SipConnectionService重写，TelephonyConnectionService是ConnectionService的子类和最终要实例化的类，因此ConnectionService实例的onCreateOutgoingConnection方法在TelephonyConnectionService执行，这个方法代码较多，最终它会调用 placeOutgoingConnection()，须要注意的是其中的phone实例的获取，对于普通的呼叫，其实例是GSMPhone或CDMAPhone，对于IMS呼叫，其实例是IMSPhone，这会决定到后面的调用流程，关于Phone实例的获取，有兴趣能够自行研究。线程

public Connection onCreateOutgoingConnection(server

PhoneAccountHandle connectionManagerPhoneAccount,blog

final ConnectionRequest request) {接口

…

final Phone phone = getPhoneForAccount(request.getAccountHandle(), isEmergencyNumber);

…

final TelephonyConnection connection =

createConnectionFor(phone, null, true /* isOutgoing */);

…

placeOutgoingConnection(connection, phone, request);

}

在ConnectionService端，onCreateOutgoingConnection会被调用到，这个方法被SipConnectionService重写，SipConnectionService是ConnectionService的子类和最终要实例化的类，因此ConnectionService实例的onCreateOutgoingConnection方法在SipConnectionService执行，这个方法里会建立一个SipConnection实例，再调用 createConnectionForProfile()，

createConnectionForProfile则获取到SipPhone实例，而后调用startCallWithPhone，startCallWithPhone代码以下，它经过Phone.dial进行拨号，并返回一个com.android.internal.telephony.Connection类型的链接，

private void placeOutgoingConnection(

TelephonyConnection connection, Phone phone, ConnectionRequest request) {…

originalConnection = phone.dial(number, request.getVideoState());

…

if (originalConnection == null) {…

} else {

connection.setOriginalConnection(originalConnection);

}

}

3.6.2.3.SIP呼叫流程SipPhone层

SipPhone.dail设置当前状态为DIALING，调用SipManager.makeAudioCall进行呼叫，并设置监听器，

void dial() throws SipException {

setState(Call.State.DIALING);

mSipAudioCall = mSipManager.makeAudioCall(mProfile, mPeer, null,

TIMEOUT_MAKE_CALL);

mSipAudioCall.setListener(mAdapter);

}

SipManager的makeAudioCall则新建一个SipAudioCall实例，同时建立一个SipSession实例，并使用SipAudioCall.makeCall进行呼叫，

public SipAudioCall makeAudioCall(SipProfile localProfile,

SipProfile peerProfile, SipAudioCall.Listener listener, int timeout)

throws SipException {

if (!isVoipSupported(mContext)) {

throw new SipException("VOIP API is not supported");

}

SipAudioCall call = new SipAudioCall(mContext, localProfile);

call.setListener(listener);

SipSession s = createSipSession(localProfile, null);

call.makeCall(peerProfile, s, timeout);

return call;

}

后者会调用sipSession.makeCall，这里的处理方式和IMS呼叫比较类似。

3.6.2.4.SIP呼叫流程Sip接口层

SipSession.makeCall则使用了一个AIDL接口ISipSession，ISipSession接口的实现是在SipSessionGroup.java，(frameworks\opt\net\voip\src\java \com\android\server\sip)，

class SipSessionGroup implements SipListener {

…

class SipSessionImpl extends ISipSession.Stub {

…

public void makeCall(SipProfile peerProfile, String sessionDescription,

int timeout) {

doCommandAsync(new MakeCallCommand(peerProfile, sessionDescription,

timeout));

}

…}

}

这里调用doCommandAsync，进而启动一个线程，执行processCommand，再到process，根据SipSession.State，咱们选择READY_TO_CALL，

public boolean process(EventObject evt) throws SipException {

…

switch (mState) {

case SipSession.State.REGISTERING:

case SipSession.State.DEREGISTERING:

processed = registeringToReady(evt);

break;

case SipSession.State.READY_TO_CALL:

processed = readyForCall(evt);

break;

case SipSession.State.INCOMING_CALL:

processed = incomingCall(evt);

break;

case SipSession.State.INCOMING_CALL_ANSWERING:

processed = incomingCallToInCall(evt);

break;

case SipSession.State.OUTGOING_CALL:

case SipSession.State.OUTGOING_CALL_RING_BACK:

processed = outgoingCall(evt);

break;

case SipSession.State.OUTGOING_CALL_CANCELING:

processed = outgoingCallToReady(evt);

break;

case SipSession.State.IN_CALL:

processed = inCall(evt);

break;

case SipSession.State.ENDING_CALL:

processed = endingCall(evt);

break;

default:

processed = false;

}

return (processed || processExceptions(evt));

}

}

}

在readyForCall()方法里面，根据事件类型是MakeCallCommand，咱们判断执行的代码以下，主要是使用mSipHelper.sendInvite向SIP协议栈发起呼叫请求，再进行本地会话管理，最后经过监听器通知状态变化。

private boolean readyForCall(EventObject evt) throws SipException {

// expect MakeCallCommand, RegisterCommand, DEREGISTER

if (evt instanceof MakeCallCommand) {

mState = SipSession.State.OUTGOING_CALL;

MakeCallCommand cmd = (MakeCallCommand) evt;

mPeerProfile = cmd.getPeerProfile();

if (mReferSession != null) {

mSipHelper.sendReferNotify(mReferSession.mDialog,

getResponseString(Response.TRYING));

}

mClientTransaction = mSipHelper.sendInvite(

mLocalProfile, mPeerProfile, cmd.getSessionDescription(),

generateTag(), mReferredBy, mReplaces);

mDialog = mClientTransaction.getDialog();

addSipSession(this);

startSessionTimer(cmd.getTimeout());

mProxy.onCalling(this);

return true;

}

…

后续就是sip协议栈的响应处理，有兴趣能够研究sip协议栈的流程和内容。