It will resemble the same sample App for iOS SDK and will have just a Dial Button and a Hang up button and try to register on startup. It will be contacting our Demo Restcomm instance.

Add Support for WebRTC Video.

If the INVITE from the application doesn't result in a call(user busy, not picked, congestion etc) then the RTP stream that is created before sending the INVITE should be destroyed.

When terminated at local side, everything is ok

When call is disconnected phone's state is not being set to normal
audio.setMode(AudioManager.RINGER_MODE_NORMAL); in MainActivity
is not setting the mode correctly

This issue tracks the interface & stubs for all involved java objects so that project is error-free. Underneath existing implementation will be integrated as a separate issue.

Currently investigated in the Restcomm side:

RestComm/Restcomm-Connect#298

And make available in GitHub pages

I think we should wire the SipUA event to RCDeviceListener.onDisconnected() that will affect the pending connection.

Currently, the low level, JAIN library is part of the example 'JAIN', as a separate package. What we need is for the existing JAIN library to be turned into a separate module and pull it outside example code so that it is easy to use and introduce yet another module that will hold the high level android client API (which will be using the JAIN module). Example directory, except from 'JAIN' example that exists currently will also get two more projects HelloWorld and Messenger, as discussed

Add a accept/Reject screen

Modify the SDK so that the API to place a call allows to take a List of parameters as Key,Value pair. Those would be added as custom SIP Headers in the initial INVITE sent by the SDK to Restcomm so that RestComm can pass those headers to the application on RVD as described in https://www.twilio.com/docs/sip/sip-twiml.

Custom SIP headers are converted to HTTP headers when Restcomm makes requests to an RVD application and are finally available as RVD variables.

For example, the following sip header:

X-MyApp-Language

will be available as an RVD variable named:

x_myapp_language

Mind the name mangling between the initial SIP header and the final RVD variable name:

"All alpha characters are converted to lowercase and all non-alphanumeric characters are converted to underscore "_".

That used to be enabled in the Application level but it's more efficient if its moved to the module level

This means adding

• callbacks for connection and device entities
• error codes/texts definitions in a central place

Which shouldn't happen -this is normally conducted in the Proxy side not the UA

Currently we get notifications only when the App is on the foreground, but we need to be getting notifications when in the home screen or in another App

i am developing a softphone using restcomm-android-sdk, everything is fine except when i disconnect the call using Hangup() then both callee and caller phone doesnot disconnected

The side effect is that when the BYE is constructed it is takes by Restcomm to be another SIP entity and Restcomm tries to forward it instead of replying with a 200 OK.

Keep in mind that for some reason INVITE works

Apparently the Via branch of the first INVITE is used instead of the second that comes with Authentication info, which is the one that succeeds.

I'm getting this exception when specifying hostname as the proxy IP. Is this expected?

Place them more centrally so we don't need to duplicate them to use them in another sample project. Use a similar directory scheme to the iOS SDK

I think the crash occurs when the reply to the CANCEL is received

1. In SipStackAndroid>processResponse() add a case for INVITE
2. Read the SDP packet and fire SipEvent with type CALL
3. In MainActivity catch the event, update UI and create rtp stream to connect with the call

Keep in mind that we will implement that the Android way, using the default settings facilities provided

Be Able to use the android WebRTC compliant media capabilities to make calls to WebRTC Clients in the browser.

i am upgrading previous Sdk with new one, Null pointer Exception at DeviceImpl.GetInstance().Call in new uploaded sdk
java.lang.NullPointerException
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.ActivityThread.performLaunchActivity(ActivityThread.java:2343)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.ActivityThread.handleLaunchActivity(ActivityThread.java:2395)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.ActivityThread.access$600(ActivityThread.java:162)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.ActivityThread$H.handleMessage(ActivityThread.java:1364)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.os.Handler.dispatchMessage(Handler.java:107)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.os.Looper.loop(Looper.java:194)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.ActivityThread.main(ActivityThread.java:5371)
06-18 21:37:48.052: E/AndroidRuntime(11588): at java.lang.reflect.Method.invokeNative(Native Method)
06-18 21:37:48.052: E/AndroidRuntime(11588): at java.lang.reflect.Method.invoke(Method.java:525)
06-18 21:37:48.052: E/AndroidRuntime(11588): at com.android.internal.os.ZygoteInit$MethodAndArgsCaller.run(ZygoteInit.java:833)
06-18 21:37:48.052: E/AndroidRuntime(11588): at com.android.internal.os.ZygoteInit.main(ZygoteInit.java:600)
06-18 21:37:48.052: E/AndroidRuntime(11588): at dalvik.system.NativeStart.main(Native Method)
06-18 21:37:48.052: E/AndroidRuntime(11588): Caused by: java.lang.NullPointerException
06-18 21:37:48.052: E/AndroidRuntime(11588): at com.android.internal.os.LoggingPrintStream.println(LoggingPrintStream.java:298)
06-18 21:37:48.052: E/AndroidRuntime(11588): at org.mobicents.restcomm.android.sipua.impl.sipmessages.Invite.MakeRequest(Invite.java:129)
06-18 21:37:48.052: E/AndroidRuntime(11588): at org.mobicents.restcomm.android.sipua.impl.SipManager.Call(SipManager.java:735)
06-18 21:37:48.052: E/AndroidRuntime(11588): at org.mobicents.restcomm.android.sipua.impl.DeviceImpl.Call(DeviceImpl.java:183)
06-18 21:37:48.052: E/AndroidRuntime(11588): at me.cs.oicalls.CallOut.onCreate(CallOut.java:78)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.Activity.performCreate(Activity.java:5122)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.Instrumentation.callActivityOnCreate(Instrumentation.java:1084)
06-18 21:37:48.052: E/AndroidRuntime(11588): at android.app.ActivityThread.performLaunchActivity(ActivityThread.java:2307)
06-18 21:37:48.052: E/AndroidRuntime(11588): ... 11 more

Scenario:

• Call from Android Messenger -> Linksys phone through Restcomm
• Everything is ok, media flows both directions
• Hangup from Messenger
• Retry the same call, but this time now outgoing media from Messenger only incoming

On the 3rd call some times it works some times it doesn't

Currently, we can only tell of an incoming call through the Android Studio console

Currently there are a couple of places where the sipmessagetest MainActivity is referenced

First call works fine. Here's the exception:

06-17 18:26:18.015 4003-4327/com.telestax.restcomm_helloworld I/System.out﹕ Dispatching event:CALL_CONNECTED
06-17 18:26:18.015 4003-4327/com.telestax.restcomm_helloworld I/System.out﹕ Sip Event fired
06-17 18:26:18.015 4003-4327/com.telestax.restcomm_helloworld W/dalvikvm﹕ threadid=18: thread exiting with uncaught exception (group=0x41b71db8)
06-17 18:26:18.025 4003-4327/com.telestax.restcomm_helloworld E/AndroidRuntime﹕ FATAL EXCEPTION: Thread-1518
Process: com.telestax.restcomm_helloworld, PID: 4003
java.lang.IllegalStateException: Busy
at android.net.rtp.RtpStream.associate(RtpStream.java:155)
at org.mobicents.restcomm.android.sipua.impl.SoundManager.setupAudio(SoundManager.java:59)
at org.mobicents.restcomm.android.sipua.impl.DeviceImpl.onSipMessage(DeviceImpl.java:82)
at org.mobicents.restcomm.android.sipua.impl.SipManager.dispatchSipEvent(SipManager.java:178)
at org.mobicents.restcomm.android.sipua.impl.SipManager.access$300(SipManager.java:65)
at org.mobicents.restcomm.android.sipua.impl.SipManager$2.run(SipManager.java:675)
06-17 18:26:18.075 4003-4021/com.telestax.restcomm_helloworld I/System.out﹕ ACK

Currently no such functionality is apparent in IDevice and ISipManager interfaces

Low level API (SipUA) has them covered, but they need to be propagated all the way to the Application

That will have the additional UI elements to support messaging (much like messenger in iOS SDK)

Extract the rtp port from SDP when 200OK is received from the calle.

Would it be possible to add DTMF support? I need this for my application :) I'll be willing to spend some time on this, if someone can give me directions..

For that listener interfaces will be introduced:

• SipUAConnectionListener: to notify an RCConnection of an update
• SipUADeviceListener: to notify an RCDevice of an update

Need to implement it both on SipUA and properly expose it in Android Client API

Create an Android Library that support compatibility of the Twilio Android SDK https://www.twilio.com/docs/client/android and extends it with Messaging.