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.
Related
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.
We published the game on russian server and 1% of people couldn't connect to server on 46xx port through raw TCP while they can load it's HTML page (through HTTP). Most of such people live in Germany, Israel....
Why is it so? What's the politics decisions lay behind it? We discovered that their such ports (which are free on IANA) are closed. Does it mean that such people cannot run Steam (and, then, play all games which you can buy through it), play WoW and many other modern games which use TCP through 4xxx ports?
Thank you.
ISPs have been known to filter certain ports for various reasons. Users should complain loudly to them (or switch) in order to send a signal that such is not to be tolerated. You can encourage them to do so but of course that doesn't solve your problem (or really answer your question).
Common reasons are:
- trying to block bittorrent traffic
- limit bandwidth usage (largely related to previous reason)
- security (mistaken)
- control (companies often don't want employees goofing off)
The easiest thing for you to do is run your game over port 443 (perhaps as an alternate). That's HTTPS and so will not generally be blocked. However, because HTTPS is encrypted, there's no way to inspect the stream to know if its web traffic or something else and thus you can run any data stream (encrypted or not) that you wish over it.
That's precisely correct. In fact every public web site would by default block all ports except the ones they expect to be running some traffic they would want to.
This is the reason many applications often try to encapsulate their programs to use port 80 which can't be blocked as long as some one wants http traffic to run.
They simply don't want any application that they haven't approved to run through their servers. If you have a sensitive server in public you surely won't want any one to use your machine for any apps that you don't allow. A common reason is applications that eat up bandwidth such as bittorent, edonkey, gnutella as well as streaming, voip and other high bandwidth consuming apps
Quick question: do most chat applications (ie. AIM, Skype, Oovoo) use peer to peer UDP exchange for talking to other users or an echoing TCP connection with a server? Or some combination in-between?
Traditionally, most applications used a TURN-like solution (i.e., communication via a server) to overcome NAT traversal issues. Since chat does not consume much bandwidth, servers could support thousands of communications.
But now that P2P has evolved and the NAT traversal issues are now well understood, some use direct UDP communication provided that the users' NAT allows this (i.e., STUN-like communication). They still need a central server to punch the hole though. Direct communication is also helpful when lots of data needs to be transmitted.
I believe it is fair to say that most modern frameworks use a combination of both.
when you need small fragments of data, such as text messaging, there's no need of using P2P. data can be transmitted from client1 to server, and from server back to the client2.
When you need to transfer data quickly between clients, in cases such as VoIP (voice over IP), or file transfer, you will use P2P.
A pretty standard IM protocol is XMPP. I know it's used by Google Talk, as well as a few other big names in chat.
I'm making a network game (1v1) where in-game its p2p - no need for a game server.
However, for players to be able to "find each other", without the need to coordinate in another medium and enter IP addresses (similar to the modem days of network games), I need to have a coordination/matching server.
I can't use regular web hosting because:
The clients will communicate in UDP.
Therefore I'll need to do UDP Hole Punching to be able to go through the NAT
That would require the server to talk in UDP and know the client's IP and port
afaik with regular web hosting (php/etc) I can only get the client's IP address and can only communicate in TCP (HTTP).
Options I am currently considering:
Use a hosting solution where my program can accept UDP connection. (any recommendations?)
UDPonNAT seems to do this but uses GTalk and requires each client to have a GTalk account for this (which probably makes it an unsuitable solution)
Any ideas? Thanks :)
First, let me say that this is well out of my realm of expertise, but I found myself very interested, so I've been doing some searching and reading.
It seems that the most commonly prescribed solution for UDP NAT traversal is to use a STUN server. I did some quick searches to see if there are any companies that will just straight-up provide you with a STUN hosting solution, but if there even were any, they were buried in piles of ads for simple web hosting.
Fortunately, it seems there are several STUN servers that are already up and running and free for public use. There is a list of public STUN servers at voip-info.org.
In addition, there is plenty more information to be had if you explore SO questions tagged "nat".
I don't see any other choice than to have a dedicated server running your code. The other solutions you propose are, shall we say, less than optimal.
If you start small, virtual hosting will be fine. Costs are pretty minimal.
Rather than a full-blown dedicated server, you could just get a cheap shared hosting service and have the application interface with a PHP page, which in turn interfaces with a MySQL database backend.
For example, Lunarpages has a $3/month starter package that includes 5gb of space and 50gb of bandwidth. For something this simple, that's all you should need.
Then you just have your application poll the web page for the list of games, and submit a POST request in order to add their own game to the list.
Of course, this method requires learning PHP and MySQL if you don't already know them. And if you do it right, you can have the PHP page enter a sort of infinite loop to keep the connection open and just feed updates to the client, rather than polling the page every few seconds and wasting a lot of bandwidth. That's way outside the scope of this answer though.
Oh, and if you're looking for something absolutely free, search for a free PHP host. Those exist too! Even with an ad-supported host, your app could just grab the page and ignore the ads when you parse the list of games. I know that T35 used to be one of my favorites because their free plan doesn't track space or bandwidth (it limits the per-file size, to eliminate their service being used as a media share, but it shouldn't be a problem for PHP files). But of course, I think in the long run you'll be better off going with a paid host.
Edit: T35 also says "Free hosting allows 1 domain to be hosted, while paid offers unlimited domain hosting." So you can even just pay for a domain name and link it to them! I think in the short term, that's your best (cheapest) bet. Of course, this is all assuming you either know or are willing to learn PHP in order to make this happen. :)
There's nothing that every net connection will support. STUN is probably good, UPnP can work for this.
However, it's rumored that most firewalls can be enticed to pass almost anything through UDP port 53 (DNS). You might have to argue with the OS about your access to that port though.
Also, check out SIP, it's another protocol designed for this sort of thing. With the popularity of VOIP, there may be decent built-in support for this in more firewalls.
If you're really committed to UDP, you might also consider tunneling it over HTTP.
how about you break the problem into two parts - make a game matcher client (that is distinct from the game), which can communicate via http to your cheap/shared webhost. All gamers who wants to use the game matching function use this. THe game matcher client then launches the actual game with the correct parameters (IP, etc etc) after obtaining the info from your server.
The game will then use the standard way to UDP punch thru NAT, etc etc, as per your network code. The game dont actually need to know anything about the matcher client or matcher server - in the true sense of p2p (like torrents, once you can obtain your peer's IPs, you can even disconnect from the tracker).
That way, your problems become smaller.
An intermediate solution between hosting your own dedicated server and a strictly P2P networking environment is the gnutella model. In that model, there are superpeers that act like local servers, having known IP addresses and being connected to (and thus having knowledge of) more clients than a typical peer. This still requires you to run at least one superpeer yourself, but it gives you the option to let other people run their own superpeers.
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