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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
| commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
| tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /testing/web-platform/tests/webrtc/protocol/rtp-extension-support.html | |
| parent | Initial commit. (diff) | |
| download | firefox-upstream.tar.xz firefox-upstream.zip | |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc/protocol/rtp-extension-support.html')
| -rw-r--r-- | testing/web-platform/tests/webrtc/protocol/rtp-extension-support.html | 74 |
1 files changed, 74 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/protocol/rtp-extension-support.html b/testing/web-platform/tests/webrtc/protocol/rtp-extension-support.html new file mode 100644 index 0000000000..bf6954ed13 --- /dev/null +++ b/testing/web-platform/tests/webrtc/protocol/rtp-extension-support.html @@ -0,0 +1,74 @@ +<!doctype html> +<meta charset=utf-8> +<title>RTCPeerConnection RTP extensions</title> +<script src="/resources/testharness.js"></script> +<script src="/resources/testharnessreport.js"></script> +<script src="../third_party/sdp/sdp.js"></script> +<script> +'use strict'; + +async function setup() { + const pc1 = new RTCPeerConnection(); + pc1.addTransceiver('audio'); + pc1.addTransceiver('video'); + const offer = await pc1.createOffer(); + pc1.close(); + return offer.sdp; +} + +// Extensions that MUST be supported +const mandatoryExtensions = [ + // Directly referenced in WebRTC RTP usage + 'urn:ietf:params:rtp-hdrext:ssrc-audio-level', // RFC 8834 5.2.2 + 'urn:ietf:params:rtp-hdrext:sdes:mid', // RFC 8834 5.2.4 + 'urn:3gpp:video-orientation', // RFC 8834 5.2.5 + // Required for support of simulcast with RID + 'urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id', // RFC 8852 4.3 + 'urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id', // RFC 8852 4.4 +]; + +// For further testing: +// - Add test for rapid synchronization - RFC 8834 5.2.1 +// - Add test for encrypted header extensions (RFC 6904) +// - Separate tests for extensions in audio and video sections + +for (const extension of mandatoryExtensions) { + promise_test(async t => { + const sdp = await setup(); + const extensions = SDPUtils.matchPrefix(sdp, 'a=extmap:') + .map(SDPUtils.parseExtmap); + assert_true(!!extensions.find(ext => ext.uri === extension)); + }, `RTP header extension ${extension} is present in offer`); +} + +// Test for illegal remote behavior: Reassignment of hdrext ID +// in a subsequent offer/answer cycle. +promise_test(async t => { + const pc1 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc2.close()); + + pc1.addTransceiver('audio'); + await pc1.setLocalDescription(); + await pc2.setRemoteDescription(pc1.localDescription); + await pc2.setLocalDescription(); + await pc1.setRemoteDescription(pc2.localDescription); + // Do a second offer/answer cycle. + await pc1.setLocalDescription(); + await pc2.setRemoteDescription(pc1.localDescription); + const answer = await pc2.createAnswer(); + + // Swap the extension number of the two required extensions + answer.sdp = answer.sdp.replace('urn:ietf:params:rtp-hdrext:ssrc-audio-level', + 'xyzzy') + .replace('urn:ietf:params:rtp-hdrext:sdes:mid', + 'urn:ietf:params:rtp-hdrext:ssrc-audio-level') + .replace('xyzzy', + 'urn:ietf:params:rtp-hdrext:sdes:mid'); + + return promise_rejects_dom(t, 'InvalidAccessError', + pc1.setRemoteDescription(answer)); +}, 'RTP header extension reassignment causes failure'); + +</script> |