Cannot start hangouts video call because of an error
Reported by
p...@myitcv.org.uk,
Aug 9 2017
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Issue descriptionUserAgent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_12_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/62.0.3178.0 Safari/537.36 Example URL: See below Steps to reproduce the problem: 1. We can reproduce the problem with any of the named hangouts channels we use as well as a brand new session (i.e. random URL) 2. Attempt to join the hangout with video turned either on/off 3. Click the green "Join" button - you appear to join successfully momentarily before the screen shifts show the error: "Couldn't start the video call because of an error". At this point the call is effectively ended because of the error. Screen recording of problem attached. Exactly the same process works just fine in Chrome Beta. What is the expected behavior? Should be able to join the call successfully What went wrong? As above Does it occur on multiple sites: N/A Is it a problem with a plugin? N/A Did this work before? Yes Works just fine with Chrome Beta Does this work in other browsers? Yes Chrome version: 62.0.3178.0 Channel: dev OS Version: OS X 10.12.6 Flash Version:
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Aug 9 2017
Confirmed as occurring on windows as well: Windows 10 build 1703
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Aug 11 2017
Tested on reported version Dev #62.0.3178.0 on Mac 10.12.6, Windows 7, Ubuntu 14.04 and able to reproduce the issue. However, issue is not reproduced on latest Canary and Stable. Using the reverse bisect providing the bisect results, Good Build: 62.0.3179.0 (Revision: 492477). Bad Build: 62.0.3178.0 (Revision: 492239). You are probably looking for a change made after 492340 (known good), but no later than 492341 (first known bad). CHANGELOG URL: The script might not always return single CL as suspect as some perf builds might get missing due to failure. https://chromium.googlesource.com/chromium/src/+log/6fae0475a09388e17c07ba6fed383bb1e0eead2a..2320aa28460a4fd3da9e803b1fd4130958115aa7 Review ON: https://chromium-review.googlesource.com/603848 @maxmorin -- Could you please look into the issue, and merge the changes made to Dev. Thanks.
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Aug 11 2017
The following revision refers to this bug: https://chromium.googlesource.com/chromium/src.git/+/9d8da709a1c467142a153d87b10ac6f18f36955f commit 9d8da709a1c467142a153d87b10ac6f18f36955f Author: Max Morin <maxmorin@chromium.org> Date: Fri Aug 11 09:25:49 2017 [3178 merge] Roll WebRTC 19232:19254 (19 commits) Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/262f12e..375c817 $ git log 262f12e..375c817 --date=short --no-merges --format=%ad %ae %s 2017-08-07 brandtr@webrtc.org Minor improvements to VideoProcessor and corresponding test. 2017-08-07 brandtr@webrtc.org Rename WEBRTC_VIDEOPROCESSOR_H264_TESTS define to WEBRTC_USE_H264. 2017-08-07 asapersson@webrtc.org Remove unused members in MediaOptimization. 2017-08-04 deadbeef@webrtc.org Make Port (and subclasses) fully "Network"-based, instead of IP-based. 2017-08-04 kthelgason@webrtc.org Destroy compression session instead of reset it on release. 2017-08-04 philipel@webrtc.org Reland of Fix off-by-one bugs in video_coding::PacketBuffer when the buffer is filled with a single frame. (patchset #1 id:1 of https://codereview.chromium.org/2990183002/ ) 2017-08-04 eladalon@webrtc.org Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ ) 2017-08-04 srte@webrtc.org Renamed fields in common_types.h/RtcpStatistics. 2017-08-04 mflodman@webrtc.org Remove video_coding/codecs/OWNERS. 2017-08-04 magjed@webrtc.org ObjC: Support non-native frames in encoder 2017-08-03 deadbeef@webrtc.org Ignore "b=AS:-1" instead of treating as a hard error. 2017-08-03 zstein@webrtc.org JNI wrapper for PeerConnection::SetBitrate. 2017-08-03 mbonadei@webrtc.org Removing unused declared arg 2017-08-03 deadbeef@webrtc.org Relanding: Break peerconnection_jni.cc into multiple files, in "pc" directory. 2017-08-03 zhihuang@webrtc.org Revert of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #15 id:280001 of https://codereview.webrtc.org/2968693002/ ) 2017-08-03 mflodman@webrtc.org Rename ViEEncoder to VideoStreamEncoder 2017-08-03 eladalon@webrtc.org Mark ~DirectTransport with "override." 2017-08-03 mbonadei@webrtc.org Tracking mock_process_thread with a GN target 2017-08-03 eladalon@webrtc.org SSRC and RSID may only refer to one sink each in RtpDemuxer TBR=maxmorin@chromium.org CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng (cherry picked from commit 2320aa28460a4fd3da9e803b1fd4130958115aa7) Bug: 753687 Change-Id: Ia6abf9078a49ff2c70f05ea08b5e69ee3f20b9e1 Reviewed-on: https://chromium-review.googlesource.com/603848 Reviewed-by: Max Morin <maxmorin@chromium.org> Commit-Queue: Max Morin <maxmorin@chromium.org> Cr-Original-Commit-Position: refs/heads/master@{#492341} Reviewed-on: https://chromium-review.googlesource.com/611986 Cr-Commit-Position: refs/branch-heads/3178@{#3} Cr-Branched-From: cdd15784955039742fe9a8235581922d41b82d78-refs/heads/master@{#492239} [modify] https://crrev.com/9d8da709a1c467142a153d87b10ac6f18f36955f/DEPS
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Aug 11 2017
Apparently, that's not how it works for DEPS changes. Will try again.
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Aug 11 2017
Ok, I made a CL for merging to the dev branch, but it requires manual review. Bug will be updated when it lands.
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Aug 28 2017
Never merged to dev, but I assume this problem doesn't reproduce now? Reopen if it does. |
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Comment 1 by erikc...@chromium.org
, Aug 9 2017