doc: adding hls status, bitrate and sip stream documentation

lib/broadcast.js

@@ -41,6 +41,16 @@
 *       information about the playlist index file and media segment files for HLS.
 *     </li>
 *     <li>
+*       <code>hlsStatus</code> (string) &mdash; The possible states for the "hlsStatus" field are the following:
+*       <ul>
+*         <li>'connecting': the opentok server is in the process of starting transcoders. This is the initial state</li>
+*         <li>'ready': the opentok server has succesfully initialised but CDN is not consuming media</li>
+*         <li>'live': the opentok server has succesfully initialised and CDN is consuming media</li>
+*         <li>'ended': the source stream has ended (if DVR is enabled and pre-recorded media is requested then status will transition to 'live')</li>
+*         <li>'error': there is an error in the opentok platform.
+*       </ul>
+*     </li>
+*     <li>
 *       <code>rtmp</code> (Object Array) &mdash; If you specified RTMP stream endpoints,
 *       the object includes an rtmp property. This is an array of objects that include
 *       information on each of the RTMP streams.
@@ -57,6 +67,9 @@
 *   The maximum time allowed for the broadcast, in seconds.
 *   After this time, the broadcast will be stopped automatically, if it is still started.
 *
+* @propert { Number } maxBitRate
+*   Maximum bitrate allowed for broadcast composing
+*
 * @property {String} streamMode
 *   The stream mode for the broadcast. This can be set to one of the the following:
 *
@@ -139,4 +152,3 @@ var Broadcast = function Broadcast(client, json) {
 };
 
 module.exports = Broadcast;
-

lib/opentok.js

@@ -1158,7 +1158,7 @@ OpenTok = function (apiKey, apiSecret, env) {
   * an <code>error</code> object is passed in as the first parameter of the function.
   * Upon success, the function is called with no error object passed in (as the first parameter)
   * and the second parameter is an object with the following properties:
-  * 
+  *
   *   <ul>
   *     <li>
   *       <code>id</code> -- A unique ID identifying the Audio Streamer WebSocket connection.
@@ -1168,7 +1168,7 @@ OpenTok = function (apiKey, apiSecret, env) {
   *       WebSocket connection in the OpenTok session.
   *     </li>
   *   </ul>
-  * 
+  *
   * @method #websocketConnect
   * @memberof OpenTok
   */
@@ -1726,6 +1726,11 @@ OpenTok = function (apiKey, apiSecret, env) {
   *     <a href="https://tokbox.com/developer/guides/moderation/#force_mute">Muting
   *     the audio of streams in a session</a>.
   *   </li>
+  *   <li>
+  *     <code>streams</code> (optional) The stream IDs of the participants'
+  *     which will be subscribed by the SIP participant. If not provided, all
+  *     streams in session will be selected.
+  *   </li>
   * </ul>
   *
   * @return A {@link SipInterconnect} object with the following properties:
@@ -1825,7 +1830,7 @@ OpenTok.prototype.dial = function (sessionId, token, sipUri, options, callback)
  * OpenTok network. If you do not set a location hint, the OpenTok servers will be based on
  * the first client connecting to the session.
  * </li>
- * 
+ *
  *     <li><code>e2ee</code> (Boolean) &mdash;
  * Determines whether to enable <a href="https://tokbox.com/developer/guides/end-to-end-encryption/">
 `end-to-end encryption</a>