webcam recording with red5 - webcam

We have a red5 recorder/red5 server setup. This is working correctly and videos are getting created in the server. We are having a problem with the recording and playback etc not working in a streamlined way. ie after you record and play the recorded one, it may not always play. The player sort of gets stuck.
Are there any free or paid alternatives to red5 recorder that does the job really well?

The reason why the videos don't play can vary and it is not sure if another streaming server will resolve your issue. For example if you have network issues no matter what streaming server you use, it will fail.
However, alternatives might be Wowza or Adobe FMS.
Sebastian

Maybe you are suffering from the same problem I have, see:
WARN o.r.s.net.rtmpt.BaseRTMPTConnection - Response buffer was null after encoding
Are there any warnings in your log file after the stream did fail? This log file may reveal your problem, see /usr/local/red5/log/red5.log

Related

using webrtc for audio broadcast

I'm trying to stream a microphone/audio to multiple clients.
the broadcaster is a screenless raspberry, so I can't open a Webbrowser and click on "share mircophone"
The clients will be using their smartphone to listen.
the latency must be super low.
I did not find any WebRTC Demo that worked. All of them are either p2p or the scalable Broadcasting from muaz khan is only working for the initiator; not clients.
I came across Janus (which I didn't really understand what exactly this is doing) but I don't get how to install this and how to configure it.
Is there any way to easily share the microphone's output via WebRTC? Something like Apache hosting a simple website where the microphone audio is hosted on?
Thanks for all the ideas on how to solve it!
Is there any way to easily share the microphone's output via WebRTC?
No. There's nothing easy or simple about WebRTC.
the broadcaster is a screenless raspberry, so I can't open a Webbrowser and click on "share mircophone"
This is the simplest option... running a browser. Are you sure you need to actually allow it to access the audio device?
In the past, I've used a flag on Chromium to get around this problem. I don't remember exactly what that flag was, but looking at the list, it might have been...
--use-fake-ui-for-media-stream
You might also be able to use --enable-kiosk-mode.
At a minimum, if you were to open the browser interactively and enable access, that page would get automatic access in the future.
I did not find any WebRTC Demo that worked. All of them are either p2p
WebRTC is peer-to-peer, but remember that the "server" can be one of those "peers".
Finally, you can look into using GStreamer, but don't expect anything quick and easy. https://github.com/centricular/gstwebrtc-demos

Connection issues since update to 10.0

Since updating the system to version 10.0, my service has been experiencing connection issues while playing a track. I have not done any changes recently server side (however I don't want to rule out that I'm doing something wrong).
Here's what happens:
Hit Play
Track starts playing
After about 1 min to 1.30 min, I get the message in the controller: "Unable to play [track name] - the connection to [service] was lost.
After another few seconds the service skips to the next songs.
A few things to note:
I have received complaints about this issue from a few customers, only after the latest update, so I don't believe these are intermittent WiFi issues.
I have tested with serving the streaming files locally from my laptop and it works fine.
Based on all of this, I believe there is an issue in the connection between the Sonos player and the streaming server (or CDN). I'm out of ideas though, on what I should be looking for. Requests, headers etc. look fine to me.
Is there any other way to troubleshoot or see what is actually happening when the Sonos player "looses connection to the service"?
Thanks much

WebRTC -- can getUserMedia use local stream?

I want to let WebRTC encoded and play h264(NAL) stream(local file).
In the WebRTC tutorial, getUserMedia is use for get local camera connecting to the system, I don`t know if the getUserMedia function support
capture the local stream file like h264 stream.
If it doesn't work that way, may be I should modify WebRTC source code(I'm studying it).
Here is the question, If i change WebRTC code, how can i integration the new code into browser? Made it a plugin?
Firefox supports an extension to the <video> element that you can use to do this.
First, set the source of a video element:
v1.src = "file:///...";
Then you can call the (currently prefixed) mozCaptureStream or mozCaptureStreamUntilEnded function to get a MediaStream.
stream = v1.mozCaptureStream();
The proposed specification.
Note however that you need to ensure that the file is same origin with respect to the page. The same origin rules for file:/// are probably going to cause issues. Otherwise your MediaStream isn't going to be accessible to you. One way to ensure that is not to set the location directly, but to load the file using an <input type="file"> element.
As noted in other answers, Firefox currently only supports the baseline profile of H.264.
First, you are right getusermedia will not work for you. However, there are a couple of options.
Hack a stream together using RTCDataChannel. Breaking up the media stream and delivering each packet and then handling it on the client side.
Take a look at this demo for precorded media streams. I do not believe that H264 is addressed but it could help you on your way(probably for Firefox only)
Use some sort of webrtc breaker/endpoint that is native to stream the file. I know specifically that others(including myself) have streamed H264 to Firefox through the Janus-Gateway
Couple of asides:
Firefox only supports Baseline profiles in streaming h264 for a webrtc peerconnection
Chrome does not support h264 for webrtc at all
Are you trying to have getUserMedia return a h.264 encoded stream?
In which case, today it will only be possible with Firefox today, under some specific environment (cisco 264 plugin installed) and only for the base profile.
Chrome promised in november to add this capacity, but there is no timeline that I know of Expect at least Q2 2015.
Using our (temasys) commercial plugin you will soon be able to do that in IE and Safari.
Those are the only options on client side I can think of. On server side you can use whatever you want to transcode, including janus, kurento, powermedia, licode/lynkia, ....
Note: using other means like Datachannel or WebSocket are ok to transfer files, but would greatly reduce the user experience as you would not have all the added recovery (and security) mechanisms included in SRTP, DTLS, and would also not have specific mistreated media enhancements that are in webRTC like jitter, buffers, netQ, ect ...

OpenTok WebRTC - The video stream failed to connect error

Condition:
1. Two users use Chrome (latest version) on Mac
2. One-to-one video conference
Problem - sometimes (~ ones in 5-6 calls) video window shows the error - The video stream failed to connect. Please notify the site owner. It can be on the beggining, or after several minutes spent on call.
Where can be the issue?
If any addition info needed i will provide in the post.
Thanks
If I understand correctly, one in 5-6 calls results in "Video stream failed to connect error". This is most likely a firewall or router configuration issue.
If you are video chatting with different people and you are sometimes getting a "video stream failed to connect error", the person you are chatting with probably has a firewall or configuration issue.
If you are testing with yourself and sometimes getting that error, you might have firewall or configuration issues.
To verify that you firewall is configured correctly, try running this diagnostic tool.
If you would like to see when your users are experiencing this error and would like to educate them on the diagnostic tool or what the possible causes are, you can listen to exception events (#1013 in your case)
Disclaimer: I work at TokBox.
Good Luck!

Newbie question on Flash video players, products/SDKs, and API

I'm a C programmer and a total newbie to Flash/video/web world. Don't know where/how to start, and so would greatly appreciate your initial help.
Question
If I need to host flash videos off of my website (instead of embedding YouTube links on my webpages),
AND
If I need to provide player API like YouTube's that can be used, say, for supporting chromeless player versions customizable via this custom API of mine...
THEN
What do I need to do essentially...?
Write a custom Flash video player?
If yes, how? I mean, using which Adobe products / tools / SDKs / language(s)?
Is there anything free/opensource available for doing this? Especially, for Linux platform?
Write a new browser (firefox) plugin for users visiting my site?
Not sure how my custom Flash video player will get to the user visiting my site for the first time?
Any books, resources that cover this problem well?
Does the Flash content need to hosted off of a Windows server only?
Currently lost. Thanks in advance,
/SD
Flash has video playback support built-in, so all you need to do is use the Flash authoring environment or Flex to compile a .SWF file that uses the video API, with some buttons to stop and start the stream, volume, seeking, anything else you want your player to do.
Many people have already done this for you, in a way you can easily use from simple HTML. See eg. OSFLV, Flowplayer, JW...
Write a new browser (firefox) plugin for users visiting my site? Does the Flash content need to hosted off of a Windows server only?
Lord no! Flash video would never have taken off if it was just another custom-server+custom-plugin piece of unpleasantness. Though special streaming servers are possible, for the most part it's just an FLV file sitting on a web server.
(FLV is the video format supported by the Flash video playing functions. There are many, many tools you can use to convert other formats to it; I use Avidemux.)
If you are planning to use a "Progressive Download" approach, then your FLV files can be hosted on a Windows or a Linux box. Be aware that:
it is no as efficient as true
streaming.
you may not use it for live events
nor only for stored video files.
it cannot automatically detect the
end user's connection speed.
it is not possible to jump ahead to
another part while it's downloaded.
the video file will be saved on the
end user's computer.
If you are planning to use a "Streaming" approach then you can either buy and use Adobe's solution (Flash Media Server, available on both Windows and Linux box) or sign up for a hosted solution. On this page you will find recommended providers by Adobe. I personally have been using Influxis's hosting with success for a couple of years already.
You can also write your own streaming server but that would be a lot of hard work. If you are interested in that, I would recommend you have a look a Red5 which is an open source Flash Server written in Java.