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Starred by 2 users

Issue metadata

Status: Duplicate
Merged: issue 501318
Owner: ----
Closed: May 2016
Cc:
Components:
EstimatedDays: ----
NextAction: ----
OS: ----
Pri: 3
Type: Bug



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WebRTC is not working on dragon board 410c(arm64)+debian

Reported by chamo...@lge.com, May 3 2016

Issue description

Chromium Version       : 49.0.2623 <Copy from: 'about:version'>
URLs (if applicable) : http://www.96boards.org/forums/topic/dragon-board-410c-chromium-update-for-webrtc/
Other browsers tested:
  Add OK or FAIL, along with the version, after other browsers where you
have tested this issue:
     Safari:
    Firefox:
         IE:

What steps will reproduce the problem?
(1) Enviroment : Dragon board 410c(arm64) + linaro debian 8.4
(2) In WebRTC, Remote stream is not working. (Camera interface is available but just black screen)
(3)

What is the expected result?
Can detect camera interface using getusermedia().
Remote streaming is working well, and we can chat video call.

What happens instead?
Local streaming is working well, but remote is not.

Please provide any additional information below. Attach a screenshot if
possible.

 

Comment 1 Deleted

Comment 2 by chamo...@lge.com, May 4 2016

Is there any problem on linaro Debian(arm64) with using SRTP for WebRTC?
I’ve installed latest srtp packges. (https://packages.debian.org/source/jessie/srtp)
But has same issue with below chromium logs.

[51:62:0504/071416:ERROR:srtpfilter.cc(721)] Failed to init SRTP, err=5
[51:62:0504/071416:ERROR:srtpfilter.cc(721)] Failed to init SRTP, err=2
[51:62:0504/071416:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071417:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071418:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071419:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071419:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071420:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071421:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071422:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071423:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071423:ERROR:stunport.cc(90)] Binding request timed out from 192.168.43.13:44834 (wlan0)
[51:62:0504/071424:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071425:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071426:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071426:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071427:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071427:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071428:ERROR:channel.cc(579)] Can’t send outgoing RTCP packet when SRTP is inactive and crypto is required
[51:62:0504/071433:ERROR:stunport.cc(90)] Binding request timed out from 192.168.43.13:44834 (wlan0)
[51:62:0504/071442:ERROR:stunport.cc(90)] Binding request timed out from 192.168.43.13:44834 (wlan0)
Labels: TE-Hardware-Dependency
Labels: Needs-Feedback
Could you provide net-internals log? The instructions can be found here: https://sites.google.com/a/chromium.org/dev/for-testers/providing-network-details

Comment 5 by chamo...@lge.com, May 5 2016

If you use ‘chromium –no-sandbox’, remote streaming is temporarily working.


and same issue is already reported. Here is a bug-fix for it.

https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=770659
https://bugzilla.mozilla.org/attachment.cgi?id=822066&action=diff

Comment 6 by chamo...@lge.com, May 5 2016

Is there no problem applying the fix patch?
Project Member

Comment 7 by sheriffbot@chromium.org, May 5 2016

Labels: -Needs-Feedback Needs-Review
Owner: kapishnikov@chromium.org
Thank you for providing more feedback. Adding requester "kapishnikov@chromium.org" for another review and adding "Needs-Review" label for tracking.

For more details visit https://www.chromium.org/issue-tracking/autotriage - Your friendly Sheriffbot
Cc: kapishnikov@chromium.org
Components: Blink>WebRTC>Network
Mergedinto: 501318
Owner: ----
Status: Duplicate (was: Unconfirmed)
This is a WebRTC issue, not a net stack one. It should go to WebRTC folks.

That said, per comment #5, this is  issue #501318 . This is not in a build that we support, so, in general, you need to file bugs with Debian, not us. Debian builds Chromium in an unsupported configuration which makes it impossible for us to resolve a lot of issues. We also only support the latest version which is 50, not 49.

Conveniently, in 50, I took away the unsupported build knob that Debian was incorrectly setting anyway. You should ask Debian to (a) update to 50 as being out-of-date means you are missing important security fixes and (b) stop setting use_system_libsrtp because it doesn't work and, in 50 and later, was removed altogether.
Sadly WebRTC is still broken in Chromium 50 on Debian (ffmpeg issue this time), see:

https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=823312

What I can suggest to the OP is to install the backported version for Jessie, it should work there.

Oh fun. :-/ That sounds like something you all would have to get Debian to fix. We ship a bundled ffmpeg which I'm guessing Debian is unbundling. This tends not to work well for libraries inside a sandbox.

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