Adding SCTP based data channels to the existing webrtc content_browser tests.

R=phoglund@chromium.org, wjia@chromium.org

Review URL: https://codereview.chromium.org/23875039

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@229001 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 108 insertions, 12 deletions

diff --git a/content/browser/media/webrtc_browsertest.cc b/content/browser/media/webrtc_browsertest.cc
index 98e4374..3c73870 100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -235,6 +235,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel.
+// TODO(mallinath) - Remove this test after rtp based data channel is disabled.
 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataOnly) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
@@ -245,6 +246,16 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataOnly) {
   ExpectTitle("OK");
 }
 
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithSctpDataOnly) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("callWithSctpDataOnly();"));
+  ExpectTitle("OK");
+}
+
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithDataAndMedia DISABLED_CallWithDataAndMedia
@@ -254,6 +265,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataOnly) {
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel and audio and video tracks.
+// TODO(mallinath) - Remove this test after rtp based data channel is disabled.
 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndMedia) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
@@ -264,6 +276,25 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndMedia) {
   ExpectTitle("OK");
 }
 
+
+#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
+// Timing out on ARM linux bot: http://crbug.com/238490
+#define MAYBE_CallWithSctpDataAndMedia DISABLED_CallWithSctpDataAndMedia
+#else
+#define MAYBE_CallWithSctpDataAndMedia CallWithSctpDataAndMedia
+#endif
+
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+                       MAYBE_CallWithSctpDataAndMedia) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("callWithSctpDataAndMedia();"));
+  ExpectTitle("OK");
+}
+
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithDataAndLaterAddMedia DISABLED_CallWithDataAndLaterAddMedia
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index 3cc53b2..c2acbdc 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -92,7 +92,7 @@
   function callWithoutMsidAndBundle() {
     createConnections(null);
     gTestWithoutMsidAndBundle = true;
-    navigator.webkitGetUserMedia({audio:true, video:true},
+    navigator.webkitGetUserMedia({audio: true, video: true},
       addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
     waitForVideo('remote-view-1');
     waitForVideo('remote-view-2');
@@ -101,25 +101,42 @@
   // Test only a data channel.
   function callWithDataOnly() {
     createConnections({optional:[{RtpDataChannels: true}]});
-    setupDataChannel();
+    setupDataChannel({reliable: false});
+    negotiate();
+  }
+
+  function callWithSctpDataOnly() {
+    createConnections({optional: [{DtlsSrtpKeyAgreement: true}]});
+    setupSctpDataChannel({reliable: true});
     negotiate();
   }
 
   // Test call with audio, video and a data channel.
   function callWithDataAndMedia() {
     createConnections({optional:[{RtpDataChannels: true}]});
-    setupDataChannel();
-    navigator.webkitGetUserMedia({audio:true, video:true},
+    setupDataChannel({reliable: false});
+    navigator.webkitGetUserMedia({audio: true, video: true},
       addStreamToBothConnectionsAndNegotiate,
       printGetUserMediaError);
     waitForVideo('remote-view-1');
     waitForVideo('remote-view-2');
   }
 
+  function callWithSctpDataAndMedia() {
+    createConnections({optional: [{DtlsSrtpKeyAgreement: true}]});
+    setupSctpDataChannel({reliable: true});
+    navigator.webkitGetUserMedia({audio: true, video: true},
+      addStreamToBothConnectionsAndNegotiate,
+      printGetUserMediaError);
+    waitForVideo('remote-view-1');
+    waitForVideo('remote-view-2');
+  }
+
+
   // Test call with a data channel and later add audio and video.
   function callWithDataAndLaterAddMedia() {
     createConnections({optional:[{RtpDataChannels: true}]});
-    setupDataChannel();
+    setupDataChannel({reliable: false});
     negotiate();
 
     // Set an event handler for when the data channel has been closed.
@@ -128,7 +145,7 @@
       setAllEventsOccuredHandler(function() {
         document.title = 'OK';
       });
-      navigator.webkitGetUserMedia({audio:true, video:true},
+      navigator.webkitGetUserMedia({audio: true, video: true},
         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
       waitForVideo('remote-view-1');
       waitForVideo('remote-view-2');
@@ -138,7 +155,7 @@
   // Test that we can setup call and send DTMF.
   function callAndSendDtmf(tones) {
     createConnections(null);
-    navigator.webkitGetUserMedia({audio:true, video:true},
+    navigator.webkitGetUserMedia({audio: true, video: true},
       addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
     var onCallEstablished = function() {
       // Send DTMF tones.
@@ -165,7 +182,7 @@
   // stream generated by getUserMedia.
   function callWithNewVideoMediaStream() {
     createConnections(null);
-    navigator.webkitGetUserMedia({audio:true, video:true},
+    navigator.webkitGetUserMedia({audio: true, video: true},
         createNewVideoStreamAndAddToBothConnections, printGetUserMediaError);
     waitForVideo('remote-view-1');
     waitForVideo('remote-view-2');
@@ -178,7 +195,7 @@
   // mediastream works.
   function callWithNewVideoMediaStreamLaterSwitchToAudio() {
     createConnections(null);
-    navigator.webkitGetUserMedia({audio:true, video:true},
+    navigator.webkitGetUserMedia({audio: true, video: true},
         createNewVideoStreamAndAddToBothConnections, printGetUserMediaError);
 
     waitForVideo('remote-view-1');
@@ -241,10 +258,10 @@
   //    is sent to |gFirstConnection|.
   // 3. When data is received on |gFirstConnection|, the data
   //    channel is closed. The test passes when the state transition completes.
-  function setupDataChannel() {
+  function setupDataChannel(params) {
     var sendDataString = "send some text on a data channel."
     firstDataChannel = gFirstConnection.createDataChannel(
-        "sendDataChannel", {reliable : false});
+        "sendDataChannel", params);
     expectEquals('connecting', firstDataChannel.readyState);
 
     // When |firstDataChannel| transition to open state, send a text string.
@@ -281,11 +298,59 @@
     }
   }
 
+  // SCTP data channel setup is slightly different then RTP based
+  // channels. Due to a bug in libjingle, we can't send data immediately
+  // after channel becomes open. So for that reason in SCTP,
+  // we are sending data from second channel, when ondatachannel event is
+  // received. So data flow happens 2 -> 1 -> 2.
+  function setupSctpDataChannel(params) {
+    var sendDataString = "send some text on a data channel."
+    firstDataChannel = gFirstConnection.createDataChannel(
+        "sendDataChannel", params);
+    expectEquals('connecting', firstDataChannel.readyState);
+
+    // When |firstDataChannel| transition to open state, send a text string.
+    firstDataChannel.onopen = function() {
+      expectEquals('open', firstDataChannel.readyState);
+    }
+
+    // When |firstDataChannel| receive a message, send message back.
+    // initiate a new offer/answer exchange to complete the closure.
+    firstDataChannel.onmessage = function(event) {
+      expectEquals('open', firstDataChannel.readyState);
+      expectEquals(event.data, sendDataString);
+      firstDataChannel.send(sendDataString);
+    }
+
+
+    // Event handler for when |gSecondConnection| receive a new dataChannel.
+    gSecondConnection.ondatachannel = function (event) {
+      var secondDataChannel = event.channel;
+      secondDataChannel.send(sendDataString);
+
+      // When |secondDataChannel| receive a message, close the channel and
+      // initiate a new offer/answer exchange to complete the closure.
+      secondDataChannel.onmessage = function(event) {
+        expectEquals(event.data, sendDataString);
+        expectEquals('open', secondDataChannel.readyState);
+        secondDataChannel.close();
+        negotiate();
+      }
+
+      // When |secondDataChannel| transition to closed state, the test pass.
+      addExpectedEvent();
+      secondDataChannel.onclose = function() {
+        expectEquals('closed', secondDataChannel.readyState);
+        eventOccured();
+      }
+    }
+  }
+
   // Test call with a stream that has been created by getUserMedia, clone
   // the stream to a cloned stream, send them via the same peer connection.
   function addTwoMediaStreamsToOneConnection() {
     createConnections(null);
-    navigator.webkitGetUserMedia({audio:true, video:true},
+    navigator.webkitGetUserMedia({audio: true, video: true},
         CloneStreamAndAddTwoStreamstoOneConnection, printGetUserMediaError);
   }