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Home Page: https://webrtc.github.io/samples
License: BSD 3-Clause "New" or "Revised" License
WebRTC Web demos and samples
Home Page: https://webrtc.github.io/samples
License: BSD 3-Clause "New" or "Revised" License
Moving webrtc issue https://code.google.com/p/webrtc/issues/detail?id=2018 to github.
Steps to reproduce:
Result: When viewing the test page in a mobile browser, the JavaScript debugging log does not appear. This normally appears on the left side of the page when viewing on desktop.
Expected: JavaScript debug information should appear for testing purposes.
What steps will reproduce the problem?
What is the expected result?
should be visible
Sam do you want to grab this one?
While accessing the page http://googlechrome.github.io/webrtc/samples/web/content/constraints/
and checking the details by connecting, it would be more useful if there are more of the informative texts(or explanatory texts) available for the constraints used and on what details and where to be checked
Currently, adapter.js uses navigator.mozGetUserMedia to detect the browser. As adapter.js expands to support more environments, we should make the feature detection and injection more granular, so that, for example, RTCPeerConnection can be handled even in environments (e.g. Firefox Add-ons, node.js) that do not have navigator.getUserMedia.
time to connect
Currently in progress and is served at kaptenjansson.github.io/webrtc/samples/web/content/manual-test/peer2peer/.
Sending a single message over channel takes a lot less time than sending two messages, for some reason.
Hi Sir,
I am trying to implement video conferencing between two peers using WebRTC and found below URL:
http://html5videoguide.net/presentations/WebDirCode2012/websocket/webrtc.html
Above works fine when button Start Video & Stop Video buttons are clicked & it displays my machines video. but when I click on connect button to connect with the some other machine, who is also at the same URL then nothing happens. Do I need to incorporate IP address of my machine or remote machine in the code for connecting through Web.rtc ? Do I need node.js running ? Please help.
Thanks in advance,
Awadesh Badaya
Right now, the callee initiates the PeerConnection, which means during call setup, we need to wait for the caller to warm up the stack on their side. We should have the caller start the PeerConnection and let it idle until the callee connects.
Queue cron tasks to clean up rooms that have no occupants, or both occupants are not connected (and aren't experiencing a spurious disconnection).
https://cloud.google.com/appengine/docs/python/config/cron
Currently, adapter.js makes no attempt to unify the getStats behavior between different browsers. Now that the standard seems to be settling on a stats format, adapter.js should attempt to provide it.
This work item does not include munging the names of any stats keys. The goal is only to provide a unified, browser-independent access method.
All of the links in the README, which are also the links that can be found at http://googlechrome.github.io/webrtc/, do not work. They lead to a 404 error page like this: http://googlechrome.github.io/webrtc/samples/web/content/getusermedia/
I would like to be able to automatically push master to AppRTC prod, instead of having to use appcfg.
See https://developers.google.com/cloud/devtools/repo/push-to-deploy.
Make a loopback apprtc call and then press hang up. You see below errors in console.
Uncaught TypeError: Cannot read property 'iceConnectionState' of null main.js:465
Uncaught TypeError: Cannot read property 'signalingState' of null main.js:460
http://googlechrome.github.io/webrtc/samples/web/content/constraints/
Nice new look of all demo pages!
I would suggest calling MediaStreamTrack::Stop on the video track before requesting a new gUM.
Currently, the new constraints are not applied since Chrome can't handle multiple video tracks with different constraints. (This will hopefully be fixed in M36.)
Unclear if there is a Datastore wrapper than can do this automatically or if we need to do it by hand.
Fork constraints-and-stats.html to include googNoiseReduction, googNoiseSuppression, googEchoCancellation and googHighpassFilter as well.
The testrtc page should ask for video+audio (and only once) so the user don't have to allow multiple getUserMedia requests (when the page is served over HTTPS).
Because of http://crbug.com/374263, we can't use recent Mac chromium builds to bisect issues on apprtc. In particular, during the gUM call, the adapter.js line that tries to parse the version id fails, which, in turn, causes the gUM call to fail.
One suggestion by Niklas is to change adapter.js to assume latest version when the info is not available.
Thoughts?
OS: Ubuntu 14.04.1 LTS
Setup: use iptables to block nearly all outbound tcp (except 80 and 443) and udp (except dns and dhcp).
Repro steps:
Issue:
Test seems to hang at the "Data channel throughput" test. Perhaps it's missing a timeout?
Full test output below:
[ RUN ] MicrophoneTest
[ OK ] getUserMedia succeeded.
[ OK ] Audio track exists with label=Default
[ OK ] Audio num channels=1
[ OK ] Audio sample rate=44100
[ OK ] Audio power=-36.1199102794985
[ ------ ]
[ RUN ] UdpConnectivityTest
[ FAILED ] Failed to gather specified candidates
[ ------ ]
[ RUN ] TcpConnectivityTest
[ FAILED ] Failed to gather specified candidates
[ ------ ]
[ RUN ] HasIpv6Test
[ OK ] Gathered candidate with type: host address: 2620:0:1000:1600:a2e:5fff:fe0d:28c0
[ ------ ]
[ RUN ] CamResolutionsTest
[ INFO ] Supported resolution: (160x120)
[ INFO ] Supported resolution: (320x180)
[ INFO ] Supported resolution: (320x240)
[ INFO ] Supported resolution: (640x360)
[ INFO ] Supported resolution: (640x480)
[ INFO ] Resolution NOT supported: (768x576)
[ INFO ] Resolution NOT supported: (1024x576)
[ FAILED ] Camera does not support a mandatory resolution, (1280x720)
[ INFO ] Resolution NOT supported: (1280x768)
[ INFO ] Resolution NOT supported: (1280x800)
[ INFO ] Resolution NOT supported: (1920x1080)
[ INFO ] Resolution NOT supported: (1920x1200)
[ INFO ] Resolution NOT supported: (3840x2160)
[ INFO ] Resolution NOT supported: (4096x2160)
[ ------ ]
[ RUN ] CamCaptureTest
[ OK ] getUserMedia succeeded.
[ OK ] Video track exists with label = Microsoft® LifeCam VX-2000 (045e:0761)
[ INFO ] Checking if your camera is delivering frames for five seconds...
[ OK ] Expected: 640 === 640: Width OK
[ OK ] Expected: 480 === 480: Height OK
[ OK ] Expected: false === false: Camera is delivering frames
[ OK ] Expected: 0 === 0: Camera is sending non-black frames.
[ ------ ]
[ RUN ] Data channel throughput
Created issue for tracking remaining work for issue 2187 (https://code.google.com/p/webrtc/issues/detail?id=2187).
Created issue to track webrtc issue 2875 (https://code.google.com/p/webrtc/issues/detail?id=2875)
Create a short guide for developers working on this repo.
Anything else?
I noticed that after a lot of attempts this demo still not working on chrome but it's working fine with firefox, and don't understand why!
In chrome i just see a black screen.
(Tried both in local and in remote official demo)
Chrome is picky. We have implemented one of the first drafts on the constraints spec for resolutions. The minimum value for framerate should be set as minFrameRate.
Firefox supports getVideoTracks() and getAudioTracks() methods as of version 23.
https://developer.mozilla.org/en-US/docs/Web/API/MediaStream
Spawned from chromium bug: https://code.google.com/p/chromium/issues/detail?id=378205
Moving issue 1380 (https://code.google.com/p/webrtc/issues/detail?id=1380) to github.
AppRTC code should be updated to handle an ICE restart, e.g. should recover when someone switches networks.
See issue 2128 (https://code.google.com/p/webrtc/issues/detail?id=2128) for details.
As per #73 discussion.
I just ran into an issue where FF nightly (33.0a1 2014-06-15) refused to work with the datachannel sample. It turned out that it failed because the success/error callbacks to setLocalDescription are missing (they're not optional per spec).
See e.g. https://github.com/GoogleChrome/webrtc/blob/master/samples/web/content/datachannel/js/main.js#L119
This seems to be used quite often, do a
grep -r setLocalDescription * | grep ')'
in the content directory.
I'll see if I can provide a patch.
Moving issue from webrtc tracker to github.
Quoth https://github.com/GoogleChrome/webrtc/blob/master/samples/web/content/apprtc/apprtc.py#L395
debug = self.request.get('debug')
if debug == 'loopback':
# Set dtls to false as DTLS does not work for loopback.
dtls = 'false'
Indeed, failing to disable DTLS causes errors like:
Error(transport.cc:101): Answerer must use either active or passive value for setup attribute.
Since SDES is eventually going to disappear in favor of DTLS we should make sure loopback calls work with DTLS as well. The answer is probably to set up a second PeerConnection on the same page.
Once the JS version is updated the same should be done for the Android & iOS sample apps.
The testrtc page needs to tear down the use of devices properly. Right now it hangs on to the video device after all tests have been executed.
Ronghua knows the details on e2e. RTT can be obtained from connection.googRtt
Copied from https://code.google.com/p/chromium/issues/detail?id=419452.
Browser: Chrome 39.0.2167.5
Output from the troubleshooting page at http://test.webrtc.org:
[ RUN ] MicrophoneTest
[ OK ] getUserMedia succeeded.
[ OK ] Audio track exists with label=Default
[ OK ] Audio num channels=1
[ OK ] Audio sample rate=48000
[ OK ] Audio power=-25.60951604743385
[ ------ ]
[ RUN ] UdpConnectivityTest
[ OK ] Gathered candidate with type: relay address: 23.251.136.126
[ ------ ]
[ RUN ] TcpConnectivityTest
[ OK ] Gathered candidate with type: relay address: 192.158.30.23
[ ------ ]
[ RUN ] HasIpv6Test
[ FAILED ] Failed to gather specified candidates
[ ------ ]
[ RUN ] CamResolutionsTest
The second time I ran the test, I got a javascript error here:
theResolution[0] + 'x' + theResolution[1] + ')');
I propose moving all manual WebRTC manual test pages in the chromium repo to github for more exposure and make it easier for people to contribute.
Main page currently used for manual testing is peerconnection.html which we should rename (including sub pages and javascript). It also uses adapter.js for polyfill and stylesheet.css for page layout.
http://src.chromium.org/chrome/trunk/src/chrome/test/data/webrtc/manual/
To do prior moving:
To do after moving:
Add a new "it" parameter that is passed down and used as the iceTransports member in the RTCConfiguration object. For now, since iceTransports doesn't do anything, specifying it=relay should cause all non-relay candidates to be discarded in the oncandidate callback, forcing the use of the relay.
What steps will reproduce the problem?
What is the expected result?
Be able to establish a call and send DTMF tones.
What do you see instead?
The page at googlechrome.github.io says:
This demo requires the RTCPeerConnection method createDTMFSender() which is not support by this browser.
can't download the code in zip
adapter.js is being currently used in a lot of pages, but this require to use a (probably outdated) local copy, or use some tricks like RawGit. A better solution would be to publish it on Bower, that's as easy as add a bower.json file and register it's location, and it will be updated automatically.
Instead of having many getusermedia-xxxx samples, let's create a new subfolder called 'getusermedia' and move unprefixed versions underneath that dir. Same for 'peerconnection' and 'datachannel'.
'apprtc' and 'manual-test' can remain under 'content'.
Expected Result:
The alignment should not change
Actual Result:
The alignment changes as seen in the screen shot. This happens only when the details are entered and clicked on 'Connect'
Created issue to track webrtc issue 2876 (https://code.google.com/p/webrtc/issues/detail?id=2876).
Blocked on #61
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