Make RTCRtpContributingSource::rtpTimestamp required

webrtc.html

@@ -7344,7 +7344,7 @@ <h2>Methods</h2>
     required DOMHighResTimeStamp timestamp;
     required unsigned long       source;
              double              audioLevel;
-             unsigned long       rtpTimestamp;
+    required unsigned long       rtpTimestamp;
 };</pre>
         <section>
           <h2>Dictionary RTCRtpContributingSource Members</h2>
@@ -7390,8 +7390,8 @@ <h2>Dictionary RTCRtpContributingSource Members</h2>
               127 is converted to 0, and all other values are converted using
               the equation: <code>10^(-rfc_level/20)</code>.</p>
             </dd>
-            <dt><dfn data-idl><code>rtpTimestamp</code></dfn> of type
-                <span class="idlMemberType">unsigned long</span></dt>
+            <dt data-tests="RTCRtpReceiver-getSynchronizationSources.https.html"><dfn data-idl><code>rtpTimestamp</code></dfn> of type
+                <span class="idlMemberType">unsigned long</span>, required</dt>
             <dd>
               <p>The last RTP timestamp, as defined in [[!RFC3559]] Section 5.1,
               of the media played out at <var>timestamp</var>.</p>