Running into some issues using webrtc with an sfu. Occasionally we run into problems where the video track on an incoming stream is muted: true and readystate: "muted". Those are read only properties so I know I am not setting them anywhere. Are those properties determined by the browser (chrome) or are they explicitly set on the stream/track by the sfu? Any help is welcome, but I'd love to see some documentation somewhere about this. For reference, the sfu we are using is Jitsi Video Bridge.
Related
I have a simple setup for Desktop Capturing using html5 libraries.
This includes a simple webpage and a chrome-extension. I am using
Extension to get the sourceId
Using the sourceId I call navigator.mediaDevices.getUserMedia to get the MediaStream
This MediaStream is then fed into an instance of MediaRecorder for recording.
This setup works most of the times, but a few times I see that requestData() on MediaRecorder instance returns blob with empty data consistently. I am clueless as to what can cause a running setup to start misbehaving sometimes.
Some weird behaviour that I noticed in the bad state:
When I try to close/refresh the window it doesn't respond.
The MediaStreamTrack object in Step 2) above is 'live' but as soon as I go to Step 3) it becomes 'muted'.
There's no pattern to it, sometimes it even happens when I request for the MediaStreams the very first time(which rules out the possibility that there could be some dangling resources eating up the contexts)
Is there anything that I am doing wrong and am unaware of? Any help/pointers would be highly appreciated!
I made simple WebRTC application and I think that I understand WebRTC framework. I want to add new feature to my app which allows only for one user share audio/video while the second user does not share video and audio.
How can I do this? What will be difference of standard mechanism?
For this there's no difference in the standard mechanism. When specifying the constraints to getUserMedia() you would specify either {video: false} or {audio: true}. I think you need at least audio, video or data to create an SDP offer. If not, then simply create an SDP answer without attaching any streams.
After connected you can simply disable the audio/video track of the stream using getAudioTracks()/getVideoTracks() of MediaStream. Each track has the enabled property which can be set to true or false.
Do not get MediaStream and do not call peerConnection.addStream() for the user whom you don't want to share his video/audio.
There is one condition if you do this.
You have to add this constraints to peerConnection.createOffer
Firefox:
{ offerToReceiveVideo: true, offerToReceiveAudio: true }
Chrome:
{mandatory: { OfferToReceiveAudio: true, OfferToReceiveVideo: true }}
You should give this constraints while creating offer because by default it will offer to receive only what stream you add.
To decide dynamically who will only send media and only receive media you can use SDP attribute a=sendonly, a=recvonly for corresponding media to signal this or negotiate this between two party.
I am using SimpleWebRTC to create a video chat room application.
One of the requirements is, a peer machine that has no microphone and webcam, should atleast be able to hear and see the video of other peers.
Is it possible to do?
I tried this using constraints{audio: false, video: false} in regular webRTC and it works on a machine that has no microphone and webcam.
How to accomplish this using simpleWebRTC?
Thanks!
I am not familiar with simpleWebRTC, but I use PeerJS. To answer even if you don't have a webcam or mic, all I do is I answer with 'null' instead of the stream.
So in Peerjs instead of
call.answer(window.localStream);
you can say
call.answer(null);
Try to find the code in SimpleWebRTC where the call is made or answered and try this.
I've posted a similar question about the more general case, but this question deals specifically with RTMP.
We have an Adobe AIR app written in AS3 that can view live video streams from other parties. That being said, when trying to call BitmapData.draw() on one of those remote video streams (technically we're calling ImageSnapshot.captureImage()), we're getting a 2123 error - a security sandbox exception. I've seen a lot of people refer to a real simple configuration in the AMS that will allow this to work for RTMP streams, but they keep posting broken links, links to posts that only vaguely mention this configuration, etc. The one thing I did find is something that I'm having trouble applying:
http://help.adobe.com/en_US/FlashMediaServer/3.5_SS_ASD/WS5b3ccc516d4fbf351e63e3d11a11afc95e-7ec3.html#WS5b3ccc516d4fbf351e63e3d11a11afc95e-7fcb
To make the above work, I've tried setting audioSampleAccess and videoSampleAccess both to "/", but to no avail. On the AMS, in Application.xml, I've set these configurations:
<FolderAccess>true</FolderAccess>
<AudioSampleAccess enabled="true">*</AudioSampleAccess>
<VideoSampleAccess enabled="true">*</VideoSampleAccess>
I've also seen references to setting up a wide-open cross-domain policy file, but I've seen people emphasize that it's not necessary, and I don't see any clear mention of where on the server that policy file needs to be. I have, however, started setting NetStream.checkPolicyFile to true, and I have placed a cross-domain policy file in the root directory of the website itself (as opposed to the AMS).
Please don't just give me another link, unless it's to a quick, clear explanation. How do you configure the AMS to allow people to capture screenshots from RTMP video streams? For the above article, I've tried setting videoStreamAccess and audioStreamAccess both to "/", and even it didn't work. We also need to be able to do this for P2P RTMFP streams, but that's really a different question. Thanks.
I have following FMS code to fix that error in application.onConnect method:
client.videoSampleAccess = "/";
client.audioSampleAccess = "/";
Is it possible for a Chrome extension to listen for streaming audio from any of the browser's tabs? I would like to capture the streaming audio data and then analyse it.
Thanks
You could try 3 ways, neither one does provide 100% guarantee to meet your needs.
Before going into more detailed descriptions, I must note that Chrome extensions do not provide convenient tools for working on per connection level - sufficiently low level, required for stream capturing. This is by design. This is why the 1-st way is:
To look at other browsers, for example Firefox, which provides low-level APIs for connections. They are already known to be used by similar extensions. You may have a look at MediaStealer. If you do not have a specific requirement to build your system on Chrome, you should possibly move to Firefox.
You can develop a Chrome extension, which intercepts HTTP-requests by means of webRequest API, analyses their headers and extracts media urls (such as containing audio/mpeg MIME-type, for example, in HTTP-headers). Just for a quick example of code you make look at the following SO question - How to change response header in Chrome. Having the url you may force appropriate media download as a file. It will land in default downloads folder and may have unfriendly name. (I made such an extension, but I do not have requirements for further processing). If you need to further process such files, it can be a challenge to monitor them in the folder, and run additional analysis in a separate program.
You may have a look at NPAPI plugins in general, and their streaming APIs in particular. I can imagine that you create a plugin registered for, again, audio/mpeg MIME-type, and receives the data via NPP_NewStream, NPP_WriteReady and NPP_Write methods. The plugin can be wrapped into a Chrome extension. Though I made NPAPI plugins, I never used this API, and I'm not sure it will work as expected. Nethertheless, I'm mentioning this possibility here for completenees. This method requires some coding other than web-coding, meaning C/C++. NB. NPAPI plugins are deprecated and not supported in Chrome since September 2015.
Taking into account that you have some external (to the extension) "fingerprinting service" in mind, which sounds like an intelligent data processing, you may be interested in building all the system out of a browser. For example, you could, possibly, involve a HTTP-proxy, saving media from passing traffic.
If you're writing a Chrome extension, you can use the Chrome tabCapture API to record audio.
chrome.tabCapture.capture({audio: true}, function(stream) {
var recorder = new MediaRecorder(stream);
[...]
});
The rest is left as an exercise to the reader; MDN has more documentation on how to use MediaRecorder.
When this question was asked in 2013, neither chrome.tabCapture nor MediaRecorder existed.
Mac OSX solution using soundflower: http://rogueamoeba.com/freebies/soundflower/
After installing soundflower it should appear as a separate audio device in the sound preferences (apple > system preferences > sound). Divert the computer's audio to the 2ch option (stereo, 16ch is surround), then inside a DAW, such as 'audacity', set the audio input as soundflower. Now the sound should be channeled to your DAW ready for recording.
Note: having diverted the audio from the internal speakers to soundflower you will only be able to hear the audio if the 'soundflowerbed' app is actually open. You know it's open if there's a 8 legged blob in the top right task bar. Clicking this icon gives you the sound flower options.
My privoxy has the following log:
2013-08-28 18:25:27.953 00002f44 Request: api.audioaddict.com/v1/di/listener_sessions.jsonp?_method=POST&callback=_AudioAddict_WP_ListenerSession_create&listener_session%5Bid%5D=null&listener_session%5Bis_premium%5D=false&listener_session%5Bmember_id%5D=null&listener_session%5Bdevice_id%5D=6&listener_session%5Bchannel_id%5D=178&listener_session%5Bstream_set_key%5D=webplayer&_=1377699927926
2013-08-28 18:25:27.969 0000268c Request: api.audioaddict.com/v1/ping.jsonp?callback=_AudioAddict_WP_Ping__ping&_=1377699927928
2013-08-28 18:25:27.985 00002d48 Request: api.audioaddict.com/v1/di/track_history/channel/178.jsonp?callback=_AudioAddict_TrackHistory_Channel&_=1377699927942
2013-08-28 18:25:54.080 00003360 Request: pub7.di.fm/di_progressivepsy_aac?type=.flv
So I got the stream url and record it:
D:\Profiles\user\temp>wget pub7.di.fm/di_progressivepsy_aac?type=.flv
--18:26:32-- http://pub7.di.fm/di_progressivepsy_aac?type=.flv
=> `di_progressivepsy_aac#type=.flv'
Resolving pub7.di.fm... done.
Connecting to pub7.di.fm[67.221.255.50]:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: unspecified [video/x-flv]
[ <=> ] 1,234,151 8.96K/s
I got the file that can be reproduced in any multimedia pleer.