I couldn't find the code for the page https://webrtc.org/getting-started/media-devices in this repo, but in the code block for "Listening for devices changes" instead of the following:

// Listen for changes to media devices and update the list accordingly
navigator.mediaDevices.addEventListener('devicechange', event => {
    const newCameraList = getConnectedDevices('video');
    updateCameraList(newCameraList);
});

it should be:

// Listen for changes to media devices and update the list accordingly
navigator.mediaDevices.addEventListener('devicechange', async event => {
    const newCameraList = await getConnectedDevices('videoinput');
    updateCameraList(newCameraList);
});

(Added async and await, and replaced 'video' with 'videoinput'.)

Most likely in https://github.com/webrtc/webrtc-org#build-and-run-the-site-locally

As per bugs.chromium.org/p/webrtc/issues/detail?id=5560:

Reported by [email protected], Feb 19

At https://webrtc.org/architecture/ and https://webrtc.org/faq/ (and possibly others) I noticed that VP9 hadn't been mentioned. Looking at https://developers.google.com/web/updates/2016/01/vp9-webrtc it seems like WebRTC has VP9 support. This is a request to update the documentation with regards to VP9.

[email protected], Today

VP9 is included without a flag since M48 and is not the default video codec yet. And the website should be updated accordingly.

I compiled webrtc code and generated webrtc.lib and peerconnection_client.exe and peerconnection_server.exe,
but I have a strange error when I run peerconnection_client.exe program in Windows x86 Debug configuration on Visual Studio :

Error message:

C:\Program Files(x86)\Microsoft Visual Studio\2019\Professinal\VC\msvc\14.25.28610\include\deque
line 279

Expression:cannot increment value-initialized deque iterator

//////////////////

_Deque_const_iterator& operator++() {
#if _ITERATOR_DEBUG_LEVEL != 0
const auto _Mycont = static_cast<const _Mydeque*>(this->_Getcont());
_STL_VERIFY(_Mycont, "cannot increment value-initialized deque iterator");
here----> _STL_VERIFY(this->_Myoff < _Mycont->_Myoff + _Mycont->_Mysize, "cannot increment deque iterator past end");**
#endif // _ITERATOR_DEBUG_LEVEL != 0

    ++_Myoff;
    return *this;

}

///////////

With the same configuration (except is_debug=false), the relase version runs well
`

I found that https://webrtc.org/reference/getting-started/ is still being served and shows outdated information (the page doesn't exist in the github repo).

@louquillio do we need to add a step to the deployment procedure to ensure cleanup when pages are removed/moved.

Related: I created pull request #41 for cleaning up that directory entirely.

OS: Win10 Pro 64-bit
VS: VS2017,

Build webrtc library for Windows x86.
when i set is_clang = true,
peerconnection_client.exe works well.

but set is_clang = false then
peerconnection_client.exe will crash.
command:
gn gen out/x86/release_clang_flase --ide=vs2017 --args="is_debug=false rtc_include_tests=false target_cpu=\"x86\" symbol_level=0 gtest_enable_absl_printers=false is_clang=false"

two or three seconds later after starting the video chat, peerconnection_client.exe crashed.
errors:
peerconnection_client.exe!_vp8_fast_quantize_b_ssse3�() C peerconnection_client.exe!_vp8_quantize_mby�() C peerconnection_client.exe!_vp8_encode_intra16x16mby�() C peerconnection_client.exe!_vp8cx_encode_inter_macroblock�() C peerconnection_client.exe!_vp8_store_drop_frame_info�() C peerconnection_client.exe!_vp8_encode_frame�() C peerconnection_client.exe!_av1_write_uleb_obu_size�() C peerconnection_client.exe!_vp8_get_compressed_data�() C peerconnection_client.exe!webrtc::Vp8FrameConfig::Updates(enum webrtc::Vp8FrameConfig::Buffer) C peerconnection_client.exe!_vpx_codec_encode�() C peerconnection_client.exe!webrtc::LibvpxInterface::CreateEncoder(void) C++ peerconnection_client.exe!webrtc::LibvpxVp8Encoder::Encode(class webrtc::VideoFrame const &,class std::vector<enum webrtc::VideoFrameType,class std::allocator<enum webrtc::VideoFrameType> > const *) C++ peerconnection_client.exe!webrtc::VideoStreamEncoder::EncodeVideoFrame(class webrtc::VideoFrame const &,__int64) C++ peerconnection_client.exe!webrtc::VideoStreamEncoder::MaybeEncodeVideoFrame(class webrtc::VideoFrame const &,__int64) C++ peerconnection_client.exe!webrtc::VideoStreamEncoder::EncoderRateSettings::operator!=(struct webrtc::VideoStreamEncoder::EncoderRateSettings const &) C++ peerconnection_client.exe!webrtc::VideoStreamEncoder::ReleaseEncoder(void) C++ peerconnection_client.exe!webrtc::CreateTaskQueueWinFactory(void) C++ peerconnection_client.exe!webrtc::CreateTaskQueueWinFactory(void) C++ peerconnection_client.exe!rtc::PlatformThread::Run(void) C++ peerconnection_client.exe!rtc::PlatformThread::StartThread(void *) C++ kernel32.dll!76f86359() Unkown

It seems that after setting is_clang = false, the vp8 encoding is wrong.
How to solve it?

Since the contents in src/webrtc was moved to src/, some instructions on the site are no longer valid.
On https://webrtc.org/native-code/android/ I found two issues:

"For instructions on how to build and run, see webrtc/examples/androidapp/README."
should just say "...see examples/androidapp/README."

and under the section for Android Studio the argument should be
--target "//examples:AppRTCMobile" instead of --target "//webrtc/examples:AppRTCMobile"

Write instructions for building the site and serving locally, dependencies, etc.

VS2017 failed to link to webrtc.lib.
I used this command:
gn gen out / Release_x86 --ide = vs2017 --args = "is_debug = false target_cpu = " x86 \ "is_clang = false"
to generate webrtc.lib for windows, the generated peerconnection_client.exe and peerconnection_server.exe can work well.
But I built VS2017 project and cannot link webrtc.lib，
The error is:
fatal error LNK1127: library is corrupt

Commit e6a9067 was intended to address code block syntax-highlighting in the updated GitHub Pages environment, which is phasing in Jekyll 3 and a requirement to use the Rouge highlighter. This works as intended in my local clone and my GitHub fork (example):

http://www.quillio.com/webrtc-org/native-code/android/

But it's not working in the production site:

https://webrtc.org/native-code/android/

My guess is that the production site is being generated with different versions of various gems than GH Pages now uses. @crhym3, could you take a look? The most convenient way to update my local env was to uninstall all gems and then reinstall the github-pages gem. Trying to update gems piecemeal sent me into dependency hell.

Thanks!

The link to the external renderer API points to https://chromium.googlesource.com/external/webrtc/+/master/webrtc/video_engine/include/vie_render.h, which doesn't exist.

Line 130 under How can I implement my own renderer or add my own hooks in the WebRTC Platform in https://github.com/webrtc/webrtc-org/blob/gh-pages/faq/index.md

I can't find vie_render.h or the folder video_engine at all.

Thanks to @MirkoBonadei suggestion, I ran html-proofer and it appears webrtc.org has quite a few broken links. Here are the commands I used:

# build the website:
bundle exec jekyll build --config _config.yml,_config_prod.yml

# check broken links:
gem install html-proofer
htmlproofer ./_site --only-4xx --allow-hash-href --alt-ignore '/.*/'

And here's the output:

Running ["ScriptCheck", "LinkCheck", "ImageCheck"] on ["./_site"] on *.html... 

Checking 379 external links...
Ran on 75 files!

- ./_site/blog/2012/03/05/source-code-to-apprtc.appspot.com-example-app-available.html
  *  External link https://chromium.googlesource.com/external/webrtc/+/master/talk/examples/android/src/org/appspot/apprtc failed: 404 No error
- ./_site/blog/2012/03/14/peerconnection-is-now-deprecatedpeerconnection.html
  *  External link http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep failed: 404 No error
- ./_site/debug/index.html
  *  External link https://github.com/louquillio/webrtc-site failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/10 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/11 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/12 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/13 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/14 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/15 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/16 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/17 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/2 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/4 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/5 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/6 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/7 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/8 failed: 404 No error
  *  External link https://github.com/louquillio/webrtc-site/issues/9 failed: 404 No error
  *  internally linking to /archive/, which does not exist (line 344)
     <a href="/archive/">archive</a>
- ./_site/faq/index.html
  *  External link https://chromium.googlesource.com/external/webrtc/+/master/webrtc/video_engine/include/vie_render.h failed: 404 No error
- ./_site/license/ilbc-freeware/ilbc-extra-documentation/index.html
  *  External link http://www.ilbcfreeware.org/documentation/ietf55-avt-duric.pdf failed: 404 No error
  *  External link http://www.ilbcfreeware.org/documentation/ietf56-avt-duric.pdf failed: 404 No error
- ./_site/native-code/android/index.html
  *  External link https://chromium.googlesource.com/external/webrtc/+/master/webrtc/api/java/README failed: 404 No error
- ./_site/native-code/native-apis/index.html
  *  External link https://chromium.googlesource.com/external/webrtc/+/master/webrtc/api/peerconnection.h failed: 404 No error
- ./_site/start/index.html
  *  External link http://auduno.github.io/clmtrackr/face_mask.html failed: 404 No error
  *  External link http://bit.ly/learnwebrtc failed: 404 No error
  *  External link http://hu.tt failed: 404 No error
  *  External link https://plus.sandbox.google.com/113817074606039822053/posts failed: 404 No error
- ./_site/testing/conformance/index.html
  *  External link http://build.chromium.org/p/chromium.webkit/builders/WebKit%20Linux%20%28dbg%29 failed: 404 No error
- ./_site/web-apis/index.html
  *  External link http://docs.webplatform.org/wiki/apis/webrtc failed: 404 No error
htmlproofer 3.6.0 | Error:  HTML-Proofer found 31 failures!

/cc @louquillio @samdutton

The page at /status/ ("Status of Implementation") seems
to be an orphan, and has no nav link in the old or new sites.
What's the current disposition of this content?

http://webrtc.github.io/webrtc-org/bugs/ should in my IMHO contain links to Firefox and Opera bug trackers as well. The argument is that we do list a link for Chrome and we list said browsers on the front page.

Lots of people are wondering how big a checkout is and some even cancel the checkout process after 5GB thinking something is wrong.

It will be nice to have the latest 2017 Video https://www.youtube.com/watch?v=PEXnbTyygi4 in the Homepage !

Several pages are out of date when compared with the old site at https://sites.google.com/site/webrtc/.

@louquillio did you have a good way of syncing them, or is it just manual work? If so we should at least load-balance the work to get it synced up quickly.
I recall you kept a record on what changes were synced back when the new site was being developed. Maybe you can create a list of which changes needs to be synced?

Formatting of code spans and blocks looks crummy -- owing partly to imperfect local replication of GH's Jekyll environment and syntax highlighter.

Also, padding-left on pre + code is cascading improperly. First line of a code block shows superfluous indent.

With this library, there is the frame limitation hardcoded of 10ms, 20ms, 30ms where as the minimum possible is only 10ms. Is there any way to make it like 1ms / 2ms / 3ms / 4ms / 5ms / 6ms 7ms / 8ms / 9ms or so flexible or at least any reduced small possible frame? because this frame size has issues. Please help.

"ICE" is an organization that forces mexican immigrants outside the US thus might be triggering to persons of hispanic descent. It'd be much better if we renamed ICE to "ISE".

E.g. http://webrtc.org/bugs/:

There are two bug trackers relevant to WebRTC:

<//crbug.com> - for Chrome.

<//code.google.com/p/webrtc/issues/list> - for WebRTC native code.

When I am executing the send() - method on my dataChannel too often in a short period of time the message buffer flows over. I tried to fix this by stoping bthe execution on the onBufferedAmountLow event. But when the event is called the buffer is already full.

Is this a known problem or is there any workaround?

lots of people redirect how to use webRTC on android / ios to here
but sadly, there is 404 error.
does webRTC stop support native app?
or just link is changed?

Thanks.

While Chromebook ChromeOS Chrome version updated to new stable version 83.0.4103.97 from 81.xxx or lower，the sender auido got from chrome extension webrtc sound so many noise at the receiver peer on android or win or other system, just ChromeOS has this problem, others not.
And if a new tab page hold the microphone stream got by navigator.mediaDevices.getUserMedia({audio: true, video: false}) api，the noise problem disappear，and happen again while release the microphone stream or close the tab page.
Any sugestion?

the chromium issue link:
https://bugs.chromium.org/p/chromium/issues/detail?id=1094279

it looks like webrtc.org was updated. However, the new content looks like its missing a lot of things, including content updates in the last 12 months (in particular #229). Changelogs are missing, so are the protocol descriptions for various header extensions. The unified plan description/timeline page is AWOL too.

What is the plan?

Thankfully this page still gets deployed to gh-pages ;-)

As reported by [email protected]; the code review server link in https://webrtc.org/contributing/ currently points to http://webrtc-codereview.appspot.com/, but should be https://codereview.webrtc.org.

And in https://webrtc.org/native-code/, the bug tracker link should be changed from https://code.google.com/p/webrtc/issues/list to https://bugs.chromium.org/p/webrtc/issues/list?

There are also several stale URLs to Chromium build instructions.

Htmlproofer was disabled because of a tricky self-link at the rtp-hdrext page. It should be re-enabled back.
Currently there are some broken links preventing from doing that:

- ./_site/start/index.html
  *  External link https://protothon.com/blog/2012/mar/27/red-square-puzzle-group failed: 404 No error
htmlproofer 3.7.2 | Error:  HTML-Proofer found 1 failure!

When clicking the WebRTC logo I would expect to go back to the landing page but the URL is always the current one rather than real baseurl (should webrtc.org).

I took a quick look and could not figure out what's wrong due to not knowing the templating framework used. However I can take a proper look at it after the holidays.

https://webrtc.org/reference/getting-started/prerequisite-sw/ gives a 404.

It looks like the path is correct in https://github.com/webrtc/webrtc-org/blame/gh-pages/native-code/development/index.md#L360.

@louquillio do you know what's going on?

Dead Links

http://vidtok.com
http://dropple.me
http://cdn.peer5.com/pilots/kaltura/index.html

At Page

https://webrtc.org/start/

This Could be removed.

https://html5rocks.com/en/tutorials/webrtc/infrastructure/ result in 404, i.e. it is not served over HTTPS.

@samdutton do you know if there are any plans on serving HTML5rocks over HTTPS anytime soon?

Formatting of code spans and blocks looks crummy -- owing partly to imperfect local replication of GH's Jekyll environment and syntax highlighter.

Also, padding-left on pre + code is cascading improperly. First line of a code block shows superfluous indent.

Hi dev team,

I have implemented an Android app that transfer data between 2 devices. But the DataChannel object is not stable. After some message, the DC object is null. I used observer to hook CLOSING and CLOSED event and recreate another DC with other label. But the DC still null after that. Please help me

Here is my code:

public void onDataChannelDisconnected(String peer_id)
{
Controller.this.rtcClient.getPeer(peer_id).CreateDataChannel();
}

public void CreateDataChannel()
{
DataChannel.Init init = new DataChannel.Init();
init.negotiated = false;
init.ordered = true;
init.maxRetransmits = 30;
this.dc = pc.createDataChannel(UUID.randomUUID().toString(), init);
}

It will be nice to add KITE, a test engine designed to test WebRTC interoperability across browsers - into webrtc/start - Developer tools.

as of the change announced here
the information on the native code development page
is outdated, no? @anatolid: the email text looks pretty good.

In function OnSentPacket(SentPacket msg) of "pcc_network_controller.cc" file. There is codes
if (last_received_packets_.size() > 0)
sending_time = last_received_packets_.back().receive_time -
last_received_packets_.front().receive_time;
DataRate receiving_rate = bandwidth_estimate_;
if (sending_time > TimeDelta::Zero())
receiving_rate = received_size / sending_time;
bandwidth_estimate_ =
std::min(bandwidth_estimate_ * 0.5, receiving_rate);
But the unit of sending_time is ms(millisecond), the unit of received_size is bytes. So the unit of receiving_rate is B/ms，is not bps, is it right? but the unit of bandwidth_estimate_ is bps. So what does it mean? The unit is different?

Another question:
In function ComputeDelayGradient(double delay_gradient_threshold) of file "monitor_interval.cc", why the unit of delay is us(microsecond)? And by analyzing the source code of us(), I found this function does nothing but call function ToValue(), so the result is still ms(millisecond)? I am confused. Can anyone answer to me? Thanks a lot!

Is there any reason why do not want to redirect to HTTPS for webrtc.org? I guess an DNS ISP redirect is best.

@crhym3 @samdutton @louquillio WDYT?

https://webrtc.org/bugs/security/ says
(None so far)

https://bugs.chromium.org/p/project-zero/issues/list?can=1&q=webrtc&colspec=ID+Status+Restrict+Reported+Vendor+Product+Finder+Summary&cells=ids shows seven.

Add link to Unified Plan migration guide

Creating this bug mostly for myself and also /cc @MirkoBonadei because we had a discussion about this in #116. I know it's been a crazy long time but I'm getting to it now. Better late than never.