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<!DOCTYPE HTML>
<html>
<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>
|
