diff --git a/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini b/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini
index 98eed7ef6589..bcc65d46e257 100644
--- a/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini
+++ b/testing/web-platform/meta/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html.ini
@@ -2,7 +2,6 @@
   expected:
     if fission and (os == "linux") and not debug: [OK, TIMEOUT]
     if fission and (os == "win") and not debug: [OK, TIMEOUT]
-    TIMEOUT
   [[audio\] getSynchronizationSources() should contain senderCaptureTimeOffset if absolute capture time RTP header extension is negotiated]
     expected:
       if fission and (os == "linux") and not debug: [FAIL, NOTRUN]
@@ -13,13 +12,13 @@
     expected:
       if fission and (os == "win") and not debug: [FAIL, NOTRUN]
       if fission and (os == "linux") and not debug: [FAIL, NOTRUN]
-      NOTRUN
+      FAIL
 
   [Audio and video RTCRtpSynchronizationSource.senderCaptureTimeOffset must be zero]
     expected:
       if fission and (os == "win") and not debug: [FAIL, NOTRUN]
       if fission and (os == "linux") and not debug: [FAIL, NOTRUN]
-      NOTRUN
+      FAIL
 
   [[audio\] getSynchronizationSources() should not contain senderCaptureTimeOffset if absolute capture time RTP header extension is not offered]
     expected:
@@ -30,13 +29,11 @@
     expected:
       if fission and (os == "win") and not debug: [PASS, NOTRUN]
       if fission and (os == "linux") and not debug: [PASS, NOTRUN]
-      NOTRUN
 
   [[video\] getSynchronizationSources() should not contain senderCaptureTimeOffset if absolute capture time RTP header extension is not offered]
     expected:
-      if fission and (os == "linux") and not debug: [TIMEOUT, PASS, NOTRUN]
+      if fission and (os == "linux") and not debug: [PASS, NOTRUN]
       if fission and (os == "win") and not debug: [PASS, NOTRUN]
-      TIMEOUT
 
   [[audio\] getSynchronizationSources() should not contain senderCaptureTimeOffset if absolute capture time RTP header extension is offered, but not answered]
     expected: