diff options
Diffstat (limited to 'testing/web-platform/meta/webrtc/protocol')
29 files changed, 294 insertions, 0 deletions
diff --git a/testing/web-platform/meta/webrtc/protocol/RTCPeerConnection-payloadTypes.html.ini b/testing/web-platform/meta/webrtc/protocol/RTCPeerConnection-payloadTypes.html.ini new file mode 100644 index 0000000000..f63530850c --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/RTCPeerConnection-payloadTypes.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-payloadTypes.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/__dir__.ini b/testing/web-platform/meta/webrtc/protocol/__dir__.ini new file mode 100644 index 0000000000..c6a51b9705 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/__dir__.ini @@ -0,0 +1,2 @@ +lsan-allowed: [Create, CreateNullDecoderModule, IPC::Channel::Channel, MakeRefPtr, MakeUnique, NewPage, NewSegment, PLDHashTable::Add, PLDHashTable::ChangeTable, PLDHashTable::MakeEntryHandle, Realloc, RevocableStore::RevocableStore, allocate, already_AddRefed, maybe_pod_malloc, mozilla::FFmpegDecoderModule, mozilla::KnowsCompositorVideo::TryCreateForIdentifier, mozilla::detail::UniqueSelector, mozilla::ipc::IProtocol::ActorConnected, mozilla::ipc::MessageChannel::Open, mozilla::layers::BufferTextureData::CreateInternal, mozilla::layers::ImageContainer::CreatePlanarYCbCrImage, mozilla::layers::ImageContainer::EnsureRecycleAllocatorForRDD, mozilla::layers::TextureClient::CreateIPDLActor, mozilla::layers::TextureClientRecycleAllocator::CreateOrRecycle, mozilla::layers::VideoBridgeChild::Open, sctp_add_vtag_to_timewait, sctp_hashinit_flags] +leak-threshold: [default:3020800, rdd:51200] diff --git a/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini new file mode 100644 index 0000000000..9c79e38d51 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini @@ -0,0 +1,13 @@ +[bundle.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [not negotiating BUNDLE creates two separate ice and dtls transports] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [bundles on the first transport and closes the second] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [max-bundle with an offer without bundle only negotiates the first m-line] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini new file mode 100644 index 0000000000..c29b1c9c1f --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini @@ -0,0 +1,19 @@ +[candidate-exchange.https.html] + expected: + if (os == "linux") and not debug and fission: [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] + [Adding only caller -> callee candidates gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Adding only callee -> caller candidates gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Explicit offer/answer exchange gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Adding callee -> caller candidates from end-of-candidates gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini new file mode 100644 index 0000000000..51d47dc6ce --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini @@ -0,0 +1,34 @@ +[crypto-suite.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [srtpCipher is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [srtpCipher is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsGroup is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsGroup is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [dtlsCipher is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [dtlsCipher is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsVersion is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsVersion is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini new file mode 100644 index 0000000000..e5a1ec8db9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini @@ -0,0 +1,6 @@ +[dtls-fingerprint-validation.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: TIMEOUT + [Connection fails if one side provides a wrong DTLS fingerprint] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: TIMEOUT diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini new file mode 100644 index 0000000000..b399895d1b --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini @@ -0,0 +1,14 @@ +[dtls-setup.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [PC with setup=actpass should have a dtlsRole of client] + expected: FAIL + + [PC with setup=active should have a dtlsRole of server] + expected: FAIL + + [PC with setup=passive should have a dtlsRole of client] + expected: FAIL + + [dtlsRole is `unknown` before negotiation of the DTLS handshake] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini new file mode 100644 index 0000000000..1dbd03b13c --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini @@ -0,0 +1,57 @@ +[h264-profile-levels.https.html] + [Level 1 H264 video is appropriately constrained] + expected: FAIL + + [Level 2 H264 video is appropriately constrained] + expected: FAIL + + [Level 3 H264 video is appropriately constrained] + expected: FAIL + + [Level 4 H264 video is appropriately constrained] + expected: FAIL + + [Level 5 H264 video is appropriately constrained] + expected: FAIL + + [Level 6 H264 video is appropriately constrained] + expected: FAIL + + [Level 1.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 1.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 1.3 H264 video is appropriately constrained] + expected: FAIL + + [Level 2.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 2.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 3.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 3.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 4.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 4.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 5.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 5.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 6.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 6.2 H264 video is appropriately constrained] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini new file mode 100644 index 0000000000..840f90f7e7 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini @@ -0,0 +1,3 @@ +[handover-datachannel.html] + [Handover with datachannel reinitiated from new callee completes] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/handover.html.ini b/testing/web-platform/meta/webrtc/protocol/handover.html.ini new file mode 100644 index 0000000000..3f168c5b49 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/handover.html.ini @@ -0,0 +1,8 @@ +[handover.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Negotiation of handover initiated at callee works] + expected: FAIL + + [Negotiation of handover initiated at caller works] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini new file mode 100644 index 0000000000..8801b5c5f1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini @@ -0,0 +1,6 @@ +[ice-state.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [PC should enter disconnected state when a failing candidate is sent] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1557053 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini new file mode 100644 index 0000000000..09560fcfec --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini @@ -0,0 +1,10 @@ +[ice-ufragpwd.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription with a ice-ufrag containing a non-ice-char fails] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1617686 + expected: FAIL + + [setRemoteDescription with a ice-pwd containing a non-ice-char fails] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1617686 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/jsep-initial-offer.https.html.ini b/testing/web-platform/meta/webrtc/protocol/jsep-initial-offer.https.html.ini new file mode 100644 index 0000000000..19cdc7b1e2 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/jsep-initial-offer.https.html.ini @@ -0,0 +1,3 @@ +[jsep-initial-offer.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/missing-fields.html.ini b/testing/web-platform/meta/webrtc/protocol/missing-fields.html.ini new file mode 100644 index 0000000000..4a96a2feea --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/missing-fields.html.ini @@ -0,0 +1,3 @@ +[missing-fields.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/msid-generate.html.ini b/testing/web-platform/meta/webrtc/protocol/msid-generate.html.ini new file mode 100644 index 0000000000..099a2d0723 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/msid-generate.html.ini @@ -0,0 +1,29 @@ +[msid-generate.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [AddTrack with a stream produces MSID with a stream ID] + expected: FAIL + + [AddTrack with two streams produces two MSID lines] + expected: FAIL + + [AddTrack with the stream twice produces single MSID with a stream ID] + expected: FAIL + + [AddTransceiver with a stream produces MSID with a stream ID] + expected: FAIL + + [AddTransceiver with two streams produces two MSID lines] + expected: FAIL + + [AddTransceiver with the stream twice produces single MSID with a stream ID] + expected: FAIL + + [SetStreams with a stream produces MSID with a stream ID] + expected: FAIL + + [SetStreams with two streams produces two MSID lines] + expected: FAIL + + [SetStreams with the stream twice produces single MSID with a stream ID] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/msid-parse.html.ini b/testing/web-platform/meta/webrtc/protocol/msid-parse.html.ini new file mode 100644 index 0000000000..a5976d5758 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/msid-parse.html.ini @@ -0,0 +1,3 @@ +[msid-parse.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini new file mode 100644 index 0000000000..4b058cbe13 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini @@ -0,0 +1,4 @@ +[rtp-clockrate.html] + expected: TIMEOUT + [video rtp timestamps increase by approximately 90000 per second] + expected: TIMEOUT diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini new file mode 100644 index 0000000000..b42afbaaa6 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini @@ -0,0 +1,11 @@ +[rtp-demuxing.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + if (os == "mac") and not debug: [OK, TIMEOUT] + [Can demux two video tracks with different payload types on a bundled connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531460 + expected: FAIL + + [Can demux two video tracks with the same payload type on an unbundled connection] + expected: + if (os == "mac") and not debug: [PASS, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini new file mode 100644 index 0000000000..5c77f6e741 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini @@ -0,0 +1,11 @@ +[rtp-extension-support.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [RTP header extension urn:3gpp:video-orientation is present in offer] + expected: FAIL + + [RTP header extension urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id is present in offer] + expected: FAIL + + [RTP header extension urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id is present in offer] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini new file mode 100644 index 0000000000..277eb7a7f6 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini @@ -0,0 +1,5 @@ +[rtp-payloadtypes.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription with a codec in the range 64-95 throws an InvalidAccessError] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/rtx-codecs.https.html.ini b/testing/web-platform/meta/webrtc/protocol/rtx-codecs.https.html.ini new file mode 100644 index 0000000000..634ed07ca7 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtx-codecs.https.html.ini @@ -0,0 +1,3 @@ +[rtx-codecs.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/sctp-format.html.ini b/testing/web-platform/meta/webrtc/protocol/sctp-format.html.ini new file mode 100644 index 0000000000..d6d1b12461 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/sctp-format.html.ini @@ -0,0 +1,4 @@ +[sctp-format.html] + max-asserts: 3 + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/sdes-dont-dont-dont.html.ini b/testing/web-platform/meta/webrtc/protocol/sdes-dont-dont-dont.html.ini new file mode 100644 index 0000000000..87ec9989b9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/sdes-dont-dont-dont.html.ini @@ -0,0 +1,9 @@ +[sdes-dont-dont-dont.html] + expected: + if (os == "win") and debug and (processor == "x86_64") and not swgl: OK + if (os == "android") and debug and not fission: OK + if (os == "android") and debug and fission: [OK, TIMEOUT] + if (os == "win") and not debug: OK + if os == "mac": OK + [OK, ERROR] + max-asserts: 3 diff --git a/testing/web-platform/meta/webrtc/protocol/simulcast-answer.html.ini b/testing/web-platform/meta/webrtc/protocol/simulcast-answer.html.ini new file mode 100644 index 0000000000..a29b91f67c --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/simulcast-answer.html.ini @@ -0,0 +1,6 @@ +[simulcast-answer.html] + max-asserts: 3 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Using the ~rid SDP syntax in a remote offer does not control the local encodings active flag] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini b/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini new file mode 100644 index 0000000000..1422fe0bc8 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini @@ -0,0 +1,3 @@ +[simulcast-offer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/split.https.html.ini b/testing/web-platform/meta/webrtc/protocol/split.https.html.ini new file mode 100644 index 0000000000..4c5f3695ca --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/split.https.html.ini @@ -0,0 +1,3 @@ +[split.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1643050 + diff --git a/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini new file mode 100644 index 0000000000..2e54d190b9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini @@ -0,0 +1,5 @@ +[unknown-mediatypes.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Unknown media types are rejected with the port set to 0] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini new file mode 100644 index 0000000000..c14dffd01b --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini @@ -0,0 +1,11 @@ +[video-codecs.https.html] + max-asserts: 3 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [H.264 and VP8 should be supported in initial offer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534688 + expected: FAIL + + [H.264 and VP8 should be negotiated after handshake] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534687 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini new file mode 100644 index 0000000000..e59f7d68cf --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini @@ -0,0 +1,6 @@ +[vp8-fmtp.html] + expected: + if (os == "win") and debug: [OK, TIMEOUT] + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription parses max-fr and max-fs fmtp parameters] + expected: FAIL |