EDGE-630 Add voice callId to the generateTransferVerb for debugging

Author: acopeland-bw

src/main/java/com/bandwidth/webrtc/utils/WebRtcTransfer.java

@@ -2,20 +2,20 @@
 package com.bandwidth.webrtc.utils;
 
 public class WebRtcTransfer {
-    public static String generateBxml(String deviceToken) {
-        return WebRtcTransfer.generateBxml(deviceToken, "sip:sipx.webrtc.bandwidth.com:5060");
+    public static String generateBxml(String deviceToken, String voiceCallId) {
+        return WebRtcTransfer.generateBxml(deviceToken, voiceCallId, "sip:sipx.webrtc.bandwidth.com:5060");
     }
 
-    public static String generateBxml(String deviceToken, String sipUri) {
+    public static String generateBxml(String deviceToken, String voiceCallId, String sipUri) {
         return "<?xml version=\"1.0\" encoding=\"UTF-8\" ?>\n"
                 + "<Response>"
-                + generateTransferVerb(deviceToken, sipUri)
+                + generateTransferVerb(deviceToken, voiceCallId, sipUri)
                 + "</Response>\n";
     }
 
-    public static String generateTransferVerb(String deviceToken, String sipUri) {
+    public static String generateTransferVerb(String deviceToken, String voiceCallId, String sipUri) {
         return "<Transfer>\n"
-                + "\t<SipUri uui=\"" + deviceToken + ";encoding=jwt\">" + sipUri + "</SipUri>\n"
+                + "\t<SipUri uui=\"" + voiceCallId + ";encoding=base64," + deviceToken + ";encoding=jwt\">" + sipUri + "</SipUri>\n"
                 + "</Transfer>";
     }
 }