test_peerConnection_rtcp_rsize.html

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<head>

  <script type="application/javascript" src="pc.js"></script>

  <script type="application/javascript" src="stats.js"></script>

  <script type="application/javascript" src="sdpUtils.js"></script>

</head>

<body>

<pre id="test">

<script type="application/javascript">

  createHTML({

    bug: "1279153",

    title: "rtcp-rsize",

    visible: true

});

  // 0) Use webrtc-sdp

  // 1) ADD RTCP-RISZE to all video m-sections

  // 2) Check for RTCP-RSIZE in ANSWER

  // 3) Wait for media to flow

  // 4) Wait for RTCP stats

  runNetworkTest(async function (options) {

    const test = new PeerConnectionTest(options);

    let mSectionsAltered = 0;

    test.chain.insertAfter("PC_LOCAL_CREATE_OFFER", [

      function PC_LOCAL_ADD_RTCP_RSIZE(test) {

        const lines = test.originalOffer.sdp.split("\r\n");

        info(`SDP before rtcp-rsize: ${lines.join('\n')}`);

        // Insert an rtcp-rsize for each m section

        const rsizeAdded = lines.flatMap(line => {

          if (line.startsWith("m=video")) {

            mSectionsAltered = mSectionsAltered + 1;

            return [line, "a=rtcp-rsize"];

          return [line];

});

        test.originalOffer.sdp = rsizeAdded.join("\r\n");

        info(`SDP with rtcp-rsize: ${rsizeAdded.join("\n")}`);

        is(mSectionsAltered, 1, "We only altered 1 msection")

    }]);

    // Check that the rtcp-rsize makes into the answer

    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", [

      function PC_LOCAL_CHECK_RTCP_RSIZE(test) {

        const msections = sdputils.getMSections(test.pcLocal._pc.currentRemoteDescription.sdp);

        var alteredMSectionsFound = 0;

        for (msection of msections) {

          if (msection.startsWith("m=video")) {

            ok(msection.includes("\r\na=rtcp-rsize\r\n"), "video m-section includes RTCP-RSIZE");

            alteredMSectionsFound = alteredMSectionsFound + 1;

          } else {

            ok(!msection.includes("\r\na=rtcp-rsize\r\n"), "audio m-section does not include RTCP-RSIZE");

        is(alteredMSectionsFound, mSectionsAltered, "correct number of msections found");

]);

    // Make sure that we are still getting RTCP stats

    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",

      async function PC_LOCAL_AND_REMOTE_CHECK_FOR_RTCP_STATS(test) {

        await Promise.all([

          waitForSyncedRtcp(test.pcLocal._pc),

          waitForSyncedRtcp(test.pcRemote._pc),

]);

        // The work is done by waitForSyncedRtcp which will throw if

        // RTCP stats are not received.

        info("RTCP stats received!");

},

);

    test.setMediaConstraints([{audio: true}, {video: true}], []);

    await test.run();

});

</script>

</pre>

</body>

</html>