i'm wondering how does the media servers work, do they require large bandwidth if you are doing, let's say, live streaming something like ustream, and there are 10k people watching, do you need a large bandwidth or it is something like p2p ?
I'm more on the client development side with Flash than server admin, but more than likely, yes, you would need a lot of bandwidth to have 10k people watching. The good thing is that with streaming video, you're only downloading the data your watch (unlike progressive). More of an issue would be the number of concurrent connections you could handle per FMS install. 10k would probably require a lot more than 1 server running FMS apps to handle. I'm currently working on a project where we are streaming from 2 installs (beyond the installations of FMS, not sure how they load balanced it) with the hopes of supporting up to something like 2k concurrent connections. I found this article to be pretty helpful (users + bandwidth stats):
http://www.adobe.com/devnet/flashmediaserver/articles/performance_tuning_webcasts.html
The part where "code" meshes with server administration can get pretty daunting (if you ask me)...and every client wants "youtube but with X feature." At 1K a license plus BW, this can get super pricey.
Depending on your needs, you may want to use a 3rd-party FMS company to handle your streaming (especially if it's just for a single event; you can get 'per-event' pricing). Also, I recently used the justin.tv api to create a streaming video feed in Flex. It was pretty painless and all the BW is on them :)
The good part is that once FMS is running, it's super easy to develop with in Actionscript :)
Related
I have a client with a 1-2 thousand viewer audience, with everyday streams, same concurrent number of viewers.
Ive got a server set up for their website etc, but am in the process of figuring out the best way to stream with OBS onto that server, and than re-distribute that stream to clients (as an embed on the website).
Now from the calculations i did, running that kind of concurrent viewers is very problematic, as it forces you into a 10gbit link - which is very expensive, and i would ideally like to fit within 1-2gbps, if possible.
A friend of mine recommended to look into "Multicast" which supossedly uses MUCH less bandwith than regular live streaming options. Is multicast doable? Ive had a NGINX live stream set up on my server by a friend before, but never looked into the config and if multicast is supported within that. Are there any other options? What would you recommend?
Also, the service of that live stream isnt a high profit / organisation type of deal, so any pre-made services just dont make sense, as it would easily cost 40+ dollars per stream, which is just too much for my client.
Thank you for any help!
Tom
Rather than Multicast, P2P is more practical solution on Internet, to save money not bandwidth.
Especially for H5 browser, it's possible to use WebRTC DataChannel to transport P2P data.
But Multicast does not work on internet routers.
Multicast works by sending a single stream across the network to edge points where clients can 'join' the multicast to get an individual stream for them.
It requires that the network supports multicast protocols and the edges align with your users.
It is typically used when an operator has their own IP network for service like IPTV, rather than for services over the internet.
For your scenario, you would usually use an organ server and a CDN - this will usually reduce the load on your own server as the video will be cached on the network and multiple user can access the same 'chunks' of the video.
You can see and AWS example for on demand video here - other vendor and cloud providers have solutions too so this is just an example:
https://docs.aws.amazon.com/AmazonS3/latest/userguide/tutorial-s3-cloudfront-route53-video-streaming.html
You can find more complex On Demand and Live tutorial also but they are likley more that you need: https://aws.amazon.com/cloudfront/streaming/
Exploring P2P may be an option also as Winton suggests - some CDN may also leverage P2P technology internally.
After knowing about some great features of WebRTC, I thought of using WebRTC one to one audio/video calls in my web application. The web application is for many organizations/entities of a category who can register and keep recording several records daily for their internal working and about their clients. The clients of these individual organizations/entities also have access to the web application to access their details.
The purpose of using WebRTC is for communication between clients and organizations. Also for daily inquires by new people to these organizations about products and services.
While going through articles on google etc. I found broadcasting or one to many calls requires very high bandwidth to users if we don't make use of Media Server.
So what is the case for one to one calls?
Will it affect the performance of web application or bring any critical situation if several users are making audio/video calls(one to one) to each other simultaneously as a routine?
The number of users will be very large and users will be recording daily several entries as their routine work. But still it is manageable and application will be running smoothly but I am not sure about the new concept WebRTC. Will it require a very high hosting plan? Is using WebRTC for current scenario suitable or advisable?
WebRTC by its nature is Peer-to-Peer. Meaning that the streaming data is handled CLIENT side. All decoding, encoding, ICE candidate gathering/negotiation, and media encrypting/transmitting will happen on the client side and not on server side. So, you will be providing the pages, client side JS, and some data exchange(session negotiation signalling) but all in all, it is not a huge amount of work. It should be easily handled without having to worry about your host machine being over worked.
All that said, here are the only a performance concerns that would POSSIBLY affect your hosting server.
Signalling session startup, negotiations, and tare down. This is very minimal(only some json data at the beginning of a session). This should not be too much of a burden but you should be aware that if 1000 sessions start at the same time, you will have a queue of messages to direct to the needed parties. How you determine the parties, forward the messages, and what work you do server side could all affect performance. If written smartly(how to store sessions, how to forward messages, etc.) should not be a terrible burden.This could easily done with SignalR since you are on ASP.NET or you could use a separate one running Node.js(or the same box, does not matter) if you so desired.
RTP TURN relay if needed. This will probably be through a different server(or the same one as your hosting server if you want). For SOME connections, a TURN server is needed and any production ready WebRTC solution should take this into account. Here is a good open source turn server. Bandwidth usage here could be very high as RTP packets are sent to this server and the forwarded to the peer in the connection.
If you are recording the streams, you may have increased hosting traffic depending on how you implement it. Firefox supports client side recording of the streams but Chrome does not(they say it is in the works currently). You could use existing JS libraries to record the feeds client side and then push them anywhere you want. You could also push all the data through a MediaServer that will mux, demux, and forward the data to be recorded anywhere you like. Janus-Gateway videoroom is a good lightweight example of a mediaserver.
Client side is a different story.
There are higher level concerns in the Javascript. If you use one of the recording JS libraries, this is especially evident as they do canvas captures numerous times a second which are a heavy hit and would degrade the user experience.
CPU utilization by the browser will increase as the quality of the video being streamed increases. This is rather obvious as HD video frames take more CPU power to encode/decode than SD frames.
Client side bandwidth usage can also be an issue. Chrome and Firefox try to modify the bitrate of each video/audio feed dynamically but the video Bitrate can go all the way up to 2 Mbps. You can cap this in Chrome( by adding an attribute in the SDP) but not in Firefox(last I checked) as of yet.
In My office website,webpage has 3css files ,2 javascript files ,11images and 1page request total 17 requests from server, If 10000 people visit my office site ...
This may slow the website due to more requests??
And any issues to the server due to huge traffic ??
I remember My tiny office server has
Intel i3 Processor
Nvidia 2Gb Graphic card
Microsoft 2008 server
8 GB DDR3 Ram and
500GB Hard disk..
Website developed on Asp.Net
Net speed was 10mbps download and 2mbps upload.using static ip address.
There are many reasons a website may be slow.
A huge spike in Additional Traffic.
Extremely Large or non-optimized graphics.
Large amount of external calls.
Server issue.
All websites should have optimized images, flash files, and video's. Large types media slow down the overall loading of each page. Optimize each image.PNG images have an improved weighted optimization that can offer better looking images with smaller file size.You could also run a Traceroute to your site.
Hope this helps.
This question is impossible to answer because there are so many variables. It sounds like you're hypothesising that you will have 10000 simultaneous users, do you really expect there to be that many?
The only way to find out if your server and site hold up under that kind of load is to profile it.
There is a tool called Apache Bench http://httpd.apache.org/docs/2.0/programs/ab.html which you can run from the command line and simulate a number of requests to your server to benchmark it. The tool comes with an install of apache, then you can simulate 10000 requests to your server and see how the request time holds up. At the same time you can run performance monitor in windows to diagnose if there are any bottlenecks.
Example usage taken from wikipedia
ab -n 100 -c 10 http://www.yahoo.com/
This will execute 100 HTTP GET requests, processing up to 10 requests
concurrently, to the specified URL, in this example,
"http://www.yahoo.com".
I don't think that downloads your page dependencies (js, css, images), but there probably are other tools you can use to simulate that.
I'd recommend that you ensure that you enable compression on your site and set up caching as this will significanly reduce the load and number of requests for very little effort.
Rather than hardware, you should think about your server's upload capacity. If your upload bandwidth is low, of course it would be a problem.
The most possible reason is because one session is lock all the rest requests.
If you not use session, turn it off and check again.
relative:
Replacing ASP.Net's session entirely
jQuery Ajax calls to web service seem to be synchronous
I'm developing a multi-player game and I know nothing about how to connect from one client to another via a server. Where do I start? Are there any whizzy open source projects which provide the communication framework into which I can drop my message data or do I have to write a load of complicated multi-threaded sockety code? Does the picture change at all if teh clients are running on phones?
I am language agnostic, although ideally I would have a Flash or Qt front end and a Java server, but that may be being a bit greedy.
I have spent a few hours googling, but the whole topic is new to me and I'm a bit lost. I'd appreciate help of any kind - including how to tag this question.
If latency isn't a huge issue, you could just implement a few web services to do message passing. This would not be a slow as you might think, and is easy to implement across languages. The downside is the client has to poll the server to get updates. so you could be looking at a few hundred ms to get from one client to another.
You can also use the built in flex messaging interface. There are provisions there to allow client to client interactions.
Typically game engines send UDP packets because of latency. The fact is that TCP is just not fast enough and reliability is less of a concern than speed is.
Web services would compound the latency issues inherent in TCP due to additional overhead. Further, they would eat up memory depending on number of expected players. Finally, they have a large amount of payload overhead that you just don't need (xml anyone?).
There are several ways to go about this. One way is centralized messaging (client/server). This means that you would have a java server listening for udp packets from the clients. It would then rebroadcast them to any of the relevant users.
A second way is decentralized (peer to peer). A client registers with the server to state what game / world it's in. From that it gets a list of other clients in that world. The server maintains that list and notifies the other clients of people who join / drop out.
From that point forward clients broadcast udp packets directly to the other users.
If you look for communication framework with high performance try look at ACE C++ framework (it has Java bindings).
Official web-site is: http://www.cs.wustl.edu/~schmidt/ACE-overview.html
You could also look into Flash Media Interactive Server, or if you want a Java implementation, Wowsa or Red5. Those use AMF and provide native functionality for ShareObjects including synching of the ShareObjects among connected clients.
Those aren't peer to peer though (yet, it's coming soon I hear). They use centralized messaging managed by the server.
Good luck
I'm no expert in audio, so if any of you folks are, I'd appreciate your insights on this.
My client has a handful of MP3 podcasts stored at a relatively high bit rate, and I'd like to be able to serve those files to her users at "different" bit rates depending on that user's credentials. (For example, if you're an authenticated user, you might get the full, unaltered stream, but if you're not, you'd get a lower-bit-rate version -- or at least a purposely tweaked lower-quality version than the original.)
Seems like there are two options: downsampling at the source and downsampling at the client. In this case, knowing of course that the source stream would arrive at the client at a high bit rate (and that there are considerations to be made about that, which I realize), I'd prefer to alter the stream at the client somehow, rather than on the server, for several reasons.
Is doing so possible with the Flash Player and ActionScript alone, at runtime (even with a third-party library), or does a scenario like this one require a server-based solution? If the latter, can Flash Media Server handle this requirement specifically? Again, I'd like to avoid using FMS if I can, since she doesn't really have the budget for it, but if that's the only option and it's really an option, I'm open to considering it.
Thanks in advance...
Note: Please don't question the sanity of the request -- I realize it might sound a bit strange, but the requirements are what they are. In that light, for purposes of answering the question, you can ignore the source and delivery path of the bits; all I'm really looking for is an explanation of whether (and ideally how) a Flash client can downsample an MP3 audio stream at runtime, irrespective of whether the audio's arriving over a network connection or being read directly from disk. Thanks much!
I'd prefer to alter the stream at the client somehow, rather than on the server, for several reasons.
Please elucidate the reasons, because resampling on the client end would normally be considered crazy: wasting bandwidth sending the higher-quality version to a user who cannot hear it, and risking a canny user ripping the higher-quality stream at it comes in through the network.
In any case the Flash Player doesn't give you the tools to process audio, only play it.
You shouldn't need FMS to process audio at the server end. You could have a server-side script that loaded the newly-uploaded podcasts and saved them back out as lower-bitrate files which could be served to lowly users via a normal web server. For Python see eg. PyMedia, py-lame; or even a shell script using lame or ffmpeg or something from the command line should be pretty easy to pull off.
If storage is at a premium, have you looked into AAC audio? I believe Flash 9 and 10 on desktop browsers will play it. AAC in my experience takes only half of the size of the comparable MP3 (i.e. a 80kbps AAC will sound the same as a 160kbps MP3).
As for playback quality, if I recall correctly there's audio playback settings in the Publish Settings section in the Flash editor. Wether or not the playback bitrate can be changed at runtime is something I'm not sure of.