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Revision 86fb98f2c3ec4e05bc529f19e41465da62052c3a authored by Lukasz Anforowicz on 02 April 2018, 23:48:36 UTC, committed by Chromium WPT Sync on 02 April 2018, 23:48:36 UTC 
Make CORB MIME type classification consistent with the web specs.
 
https://tools.ietf.org/html/rfc7303 says that if "new media type is
introduced for an XML-based format, the name of the media type SHOULD
end with '+xml'".

https://tools.ietf.org/html/rfc6839 covers '+xml' and '+json' suffixes.

https://mimesniff.spec.whatwg.org/#xml-mime-type says "An XML MIME type
is any MIME type whose subtype ends in '+xml' or whose essence is
'text/xml' or 'application/xml'. [RFC7303]".

https://mimesniff.spec.whatwg.org/#json-mime-type says "A JSON MIME type
is any MIME type whose subtype ends in '+json' or whose essence is
'application/json' or 'text/json'."

There are no occurences of "application/xml+", "text/xml+",
"application/json+", "text/json+" or "text/x-json" in the specs above
and on various lists of MIME types like:
-
https://developer.mozilla.org/en-US/docs/Web/HTTP/Basics_of_HTTP/MIME_types/Complete_list_of_MIME_types
- https://en.wikipedia.org/wiki/Media_type
- https://www.freeformatter.com/mime-types-list.html
- https://www.sitepoint.com/mime-types-complete-list/

Bug: 826756
Cq-Include-Trybots: master.tryserver.chromium.linux:linux_mojo
Change-Id: Ied30f9728bd4f082bb620fea150f342457ea4833
Reviewed-on: https://chromium-review.googlesource.com/985211
Commit-Queue: Łukasz Anforowicz <lukasza@chromium.org>
Reviewed-by: Nick Carter <nick@chromium.org>
Cr-Commit-Position: refs/heads/master@{#547565}
1 parent 94b7255
Raw File
no-media-call.html 
<!doctype html>
<!--
This test uses the legacy callback API with no media, and thus does not require fake media devices.
-->

<html>
<head>
  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
  <title>RTCPeerConnection No-Media Connection Test</title>
</head>
<body>
  <div id="log"></div>
  <h2>iceConnectionState info</h2>
  <div id="stateinfo">
  </div>

  <!-- These files are in place when executing on W3C. -->
  <script src="/resources/testharness.js"></script>
  <script src="/resources/testharnessreport.js"></script>
  <script type="text/javascript">
  var test = async_test('Can set up a basic WebRTC call with no data.');

  var gFirstConnection = null;
  var gSecondConnection = null;

  var onOfferCreated = test.step_func(function(offer) {
    gFirstConnection.setLocalDescription(offer, ignoreSuccess,
                                         failed('setLocalDescription first'));

    // This would normally go across the application's signaling solution.
    // In our case, the "signaling" is to call this function.
    receiveCall(offer.sdp);
  });

  function receiveCall(offerSdp) {

    var parsedOffer = new RTCSessionDescription({ type: 'offer',
                                                  sdp: offerSdp });
    // These functions use the legacy interface extensions to RTCPeerConnection.
    gSecondConnection.setRemoteDescription(parsedOffer,
      function() {
        gSecondConnection.createAnswer(onAnswerCreated,
                                       failed('createAnswer'));
      },
      failed('setRemoteDescription second'));
  };

  var onAnswerCreated = test.step_func(function(answer) {
    gSecondConnection.setLocalDescription(answer, ignoreSuccess,
                                          failed('setLocalDescription second'));

    // Similarly, this would go over the application's signaling solution.
    handleAnswer(answer.sdp);
  });

  function handleAnswer(answerSdp) {
    var parsedAnswer = new RTCSessionDescription({ type: 'answer',
                                                   sdp: answerSdp });
    gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess,
                                          failed('setRemoteDescription first'));
  };

  var onIceCandidateToFirst = test.step_func(function(event) {
    // If event.candidate is null = no more candidates.
    if (event.candidate) {
      gSecondConnection.addIceCandidate(event.candidate);
    }
  });

  var onIceCandidateToSecond = test.step_func(function(event) {
    if (event.candidate) {
      gFirstConnection.addIceCandidate(event.candidate);
    }
  });

  var onIceConnectionStateChange = test.step_func(function(event) {
    assert_equals(event.type, 'iceconnectionstatechange');
    assert_not_equals(gFirstConnection.iceConnectionState, "failed", "iceConnectionState of first connection");
    assert_not_equals(gSecondConnection.iceConnectionState, "failed", "iceConnectionState of second connection");
    var stateinfo = document.getElementById('stateinfo');
    stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
                        + '<br>Second: ' + gSecondConnection.iceConnectionState;
    // Note: All these combinations are legal states indicating that the
    // call has connected. All browsers should end up in completed/completed,
    // but as of this moment, we've chosen to terminate the test early.
    // TODO: Revise test to ensure completed/completed is reached.
    if (gFirstConnection.iceConnectionState == 'connected' &&
        gSecondConnection.iceConnectionState == 'connected') {
      test.done()
    }
    if (gFirstConnection.iceConnectionState == 'connected' &&
        gSecondConnection.iceConnectionState == 'completed') {
      test.done()
    }
    if (gFirstConnection.iceConnectionState == 'completed' &&
        gSecondConnection.iceConnectionState == 'connected') {
      test.done()
    }
    if (gFirstConnection.iceConnectionState == 'completed' &&
        gSecondConnection.iceConnectionState == 'completed') {
      test.done()
    }
  });

  // Returns a suitable error callback.
  function failed(function_name) {
    return test.step_func(function() {
      assert_unreached('WebRTC called error callback for ' + function_name);
    });
  }

  // Returns a suitable do-nothing.
  function ignoreSuccess(function_name) {
  }

  // This function starts the test.
  test.step(function() {
    gFirstConnection = new RTCPeerConnection(null);
    gFirstConnection.onicecandidate = onIceCandidateToFirst;
    gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;

    gSecondConnection = new RTCPeerConnection(null);
    gSecondConnection.onicecandidate = onIceCandidateToSecond;
    gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;

    // The offerToReceiveVideo is necessary and sufficient to make
    // an actual connection.
    gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
        {offerToReceiveVideo: true});
  });
</script>

</body>
</html>
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