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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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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/RTCSctpTransport-constructor.html | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.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/RTCSctpTransport-constructor.html')
-rw-r--r-- | testing/web-platform/tests/webrtc/RTCSctpTransport-constructor.html | 125 |
1 files changed, 125 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/RTCSctpTransport-constructor.html b/testing/web-platform/tests/webrtc/RTCSctpTransport-constructor.html new file mode 100644 index 0000000000..484967f76b --- /dev/null +++ b/testing/web-platform/tests/webrtc/RTCSctpTransport-constructor.html @@ -0,0 +1,125 @@ +<!doctype html> +<meta charset="utf-8"> +<title>RTCSctpTransport constructor</title> +<script src="/resources/testharness.js"></script> +<script src="/resources/testharnessreport.js"></script> +<script src="RTCPeerConnection-helper.js"></script> +<script> +'use strict'; + +// Test is based on the following revision: +// https://rawgit.com/w3c/webrtc-pc/1cc5bfc3ff18741033d804c4a71f7891242fb5b3/webrtc.html + +// The following helper functions are called from RTCPeerConnection-helper.js: +// generateDataChannelOffer() +// generateAnswer() + +/* + 6.1. + + partial interface RTCPeerConnection { + readonly attribute RTCSctpTransport? sctp; + ... + }; + + 6.1.1. + + interface RTCSctpTransport { + readonly attribute RTCDtlsTransport transport; + readonly attribute RTCSctpTransportState state; + readonly attribute unrestricted double maxMessageSize; + attribute EventHandler onstatechange; + }; + + 4.4.1.1. Constructor + 9. Let connection have an [[SctpTransport]] internal slot, initialized to null. + + 4.4.1.6. Set the RTCSessionSessionDescription + 2.2.6. If description is of type "answer" or "pranswer", then run the + following steps: + 1. If description initiates the establishment of a new SCTP association, as defined in + [SCTP-SDP], Sections 10.3 and 10.4, create an RTCSctpTransport with an initial state + of "connecting" and assign the result to the [[SctpTransport]] slot. + */ + +promise_test(async (t) => { + const pc1 = new RTCPeerConnection(); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + t.add_cleanup(() => pc2.close()); + + assert_equals(pc1.sctp, null, 'RTCSctpTransport must be null'); + + const offer = await generateAudioReceiveOnlyOffer(pc1); + await Promise.all([pc1.setLocalDescription(offer), pc2.setRemoteDescription(offer)]); + const answer = await pc2.createAnswer(); + await pc1.setRemoteDescription(answer); + + assert_equals(pc1.sctp, null, 'RTCSctpTransport must remain null'); +}, 'setRemoteDescription() with answer not containing data media should not initialize pc.sctp'); + +promise_test(async (t) => { + const pc1 = new RTCPeerConnection(); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + t.add_cleanup(() => pc2.close()); + + assert_equals(pc1.sctp, null, 'RTCSctpTransport must be null'); + + const offer = await generateAudioReceiveOnlyOffer(pc2); + await Promise.all([pc2.setLocalDescription(offer), pc1.setRemoteDescription(offer)]); + const answer = await pc1.createAnswer(); + await pc1.setLocalDescription(answer); + + assert_equals(pc1.sctp, null, 'RTCSctpTransport must remain null'); +}, 'setLocalDescription() with answer not containing data media should not initialize pc.sctp'); + +function validateSctpTransport(sctp) { + assert_not_equals(sctp, null, 'RTCSctpTransport must be available'); + + assert_true(sctp instanceof RTCSctpTransport, + 'Expect pc.sctp to be instance of RTCSctpTransport'); + + assert_true(sctp.transport instanceof RTCDtlsTransport, + 'Expect sctp.transport to be instance of RTCDtlsTransport'); + + assert_equals(sctp.state, 'connecting', 'RTCSctpTransport should be in the connecting state'); + + // Note: Yes, Number.POSITIVE_INFINITY is also a 'number' + assert_equals(typeof sctp.maxMessageSize, 'number', + 'Expect sctp.maxMessageSize to be a number'); +} + +promise_test(async (t) => { + const pc1 = new RTCPeerConnection(); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + t.add_cleanup(() => pc2.close()); + + assert_equals(pc1.sctp, null, 'RTCSctpTransport must be null'); + + const offer = await generateDataChannelOffer(pc1); + await Promise.all([pc1.setLocalDescription(offer), pc2.setRemoteDescription(offer)]); + const answer = await pc2.createAnswer(); + await pc1.setRemoteDescription(answer); + + validateSctpTransport(pc1.sctp); +}, 'setRemoteDescription() with answer containing data media should initialize pc.sctp'); + +promise_test(async (t) => { + const pc1 = new RTCPeerConnection(); + const pc2 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); + t.add_cleanup(() => pc2.close()); + + assert_equals(pc1.sctp, null, 'RTCSctpTransport must be null'); + + const offer = await generateDataChannelOffer(pc2); + await Promise.all([pc2.setLocalDescription(offer), pc1.setRemoteDescription(offer)]); + const answer = await pc1.createAnswer(); + await pc1.setLocalDescription(answer); + + validateSctpTransport(pc1.sctp); +}, 'setLocalDescription() with answer containing data media should initialize pc.sctp'); + +</script> |