I know all about the streaming protocols and what they are good for. But what confuses me is the protocols the video stream is encoded. Because the encoders use usually RTMP or RTSP protocols and then it is up to the service provider or decoder how the video/stream is delivered (in which protocol: HLS, WebRTC, HDS, MPED-DASH etc).
So it might be a silly question but is there a way to change the encoding protocols from RTMP or RTSP? When I record a video/live-stream with a software. Right now I am using OBS. And my main goal is findign a solution how to stream one-to-many with as low latency as possible (>2s).
Also as far as I know the difference between RTSP and RTMP is one uses iOs and the other Windows OS.
So it might be a silly question but is there a way to change the encoding protocols from RTMP or RTSP? When I record a video/live-stream with a software. Right now I am using OBS. And my main goal is
Yes. There are many (many, many) streaming servers on the market. nginx, red5, wowza, etc.
Also as far as I know the difference between RTSP and RTMP is one uses iOs and the other Windows OS.
No. protocols and operating systems are not related at all. Any OS can utilize any protocol. Web browsers are limited to just a few.
When I record a video/live-stream with a software. Right now I am using OBS. And my main goal is findign a solution how to stream one-to-many with as low latency as possible (>2s).
This is a HUGE question, that really can not be answered on stack overflow. one-to-many; is many 10 or 1000 or 1000000, They are different answers. Does it need to work on web or not (different answers). What does your infrastructure look like, what is your operation budget. Are users located global, or geographically centralized? All this would change the answer. And some of those answer may be that your problem is not practical. for example >2s 100000 users globally on web would be very expensive.
Related
I am working on designing an API for consuming messages from an application that will generate a very large amount of data; 10+ of GB/s is likely, even for smaller clients. I am looking for a protocol that allows me to deliver this data in a way that is easy for clients to consume.
The obvious answer for me is: split up the messages so they are consumable over multiple connections. Each connection would consume a fraction of the overall load.
But if I do this, there are a few things I need to account for:
How does the user know they are falling behind and need to launch more connections?
Twitter says consumers should check timestamps, which could work for us
When they launch a new connection to consume more of the data, how do they specify that this is part of the same consumption session?
We could give the session a name, correlate that with a "direct" amqp queue, and let our queue do the hard work
Is there something very important I am missing.
Probably.
For this reason, I'd much rather a protocol that already exists.
The protocol would be considered extra awesome if it:
is websocket or streaming HTTP friendly
supports data compression
The problems you are describing are pretty much the same issues that video streaming has to deal with, which you probably already know. The key HTTP friendly streaming protocols are HLS (Apple), SmoothStreaming (Microsoft), HDS (Adobe) and MPEG-DASH (open protocol, but new).
When considering video streaming, it is also worth understanding whether your streams are more like 'live' streams or 'static' content - the former is generated on the fly and any given part of the live stream may only be available for a set tine, while the latter stored on the server in full and generally any part is available at any time (until the content is removed). How you stream and playback these is subtly different.
It may be that you can simply reuse one of the above video streaming protocols by wrapping your data as if it were video (or maybe it even is video), and implementing your own custom client on the receiving side.
Alternatively, these protocols could provide a good reference point if you wanted to create your own simpler protocol - there are several open source streaming servers you could look to for ideas or even adapt to your needs if that looks like a sensible route:
http://gstreamer.freedesktop.org
http://icecast.org
Video streaming is quite complex as you may already be aware, but if your use cases are simpler you may be able to ignore or remove much of the complexity - for example you may not need seek, multiple format and bit rate streams, accompanying streams (for subtitles etc). Being able to simplify like this might justify the effort to modify one of the above for your needs, if you are not able to use them out of the box.
One final point - video and audio streaming protocols usually have a built in way of dealing with delayed or lost packets. Depending on your application these may not be applicable to you so you should look carefully at this aspect if reusing a video or audio streaming protocol or server. For example, audio clients are typically tolerant of a small amount of packet loss, and will generally discard delayed packets rather than pause the audio (packets received outside the 'jitter buffer' window). If your application cannot tolerate any packet loss, then you will need to look carefully at the underlying solution and protocol to make sure it really meets your needs over all network conditions.
I want to setup a personal videoconferencing service for my family, friends and myself. The main problem I have with current options is that they are either closed-source and centralized (GG hangouts, skype) or open-source but not working in corporate environment or in hotels (due to strict firewalling rules and the "Skype is going through, if you want VOIP use that" kind of netadmin reaction).
I have two solutions then. Either setup a STUN/TURN relay server and use XMPP and SIP as I used to, but that would require my friends to setup that too. Or setup a whole VOIP server. 2 solutions come to mind: SIP and XMPP. Though to my knowledge, each of them ultimately uses the (S)RTP/RTCP protocol.
And that's the problem. Out of the specific signaling part used by the two of them, I really can't figure out the difference between them, their typical use case.
I think you're right in that as far as setting up a video conferencing system XMPP and SIP are equivalent. They both are signalling only protocols and the media sessions they set up typically use RTP (although they can both be used to set up any kind of session you want but RTP is the norm).
The biggest problem is also going to be the one you mention about getting video streams out of a corporate firewall. Skype overcomes this obstacle by sending it's media over an SSL connection and is thus able to get through firewalls. Theoretically you could do the same with RTP and in the past I once used openvpn connections with a SIP client to test some audio calls. My experience wasn't great as the audio was very choppy, assumedly as a result of all the extra packaging that is required to get the high volume of small audio packets from one end to the other. That was nearly a decade ago though so perhaps with the better CPU and bandwidth resources available now it would work better.
Personally I think I'd stick with Skype as it's going to be a big hassle to set up your own system. If you were to go ahead with your own the first option I would try would be Asterisk combined with openvpn so that if the clients were behind a firewall or had NAT issues they could connect over it.
This is probably not the best forum for such a specialized question, but at the moment I don't know of a better one (open to suggestions/recommendations).
I work on a video product which for the last 10+ years has been using proprietary communications protocol (DCOM-based) to send the video across the network. A while ago we recognized the need to standardize and currently are almost at a point of ripping out all that DCOM baggage and replacing it with a fully compliant RTP/RTSP client/server framework.
One thing we noticed during testing over the last few months is that when we switch the client to use RTP/RTSP, there's a noticeable increase in start-up latency. The problem is that it's not us but RTSP.
BEFORE (DCOM): we would send one DCOM command and before that command even returned back to the client, the server would already be sending video. -- total latency 1 RTT
NOW (RTSP): This is the sequence of commands, each one being a separate network request: DESCRIBE, SETUP, SETUP, PLAY (assuming the session has audio and video) -- total of 4 RTTs.
Works as designed - unfortunately it feels like a step backwards because prior user experience was actually better.
Can this be improved? If you stay with the standard, short answer is, NO. However, my team fully controls our entire RTP/RTSP stack and I've been thinking we could introduce a new RTSP command (without touching any of existing commands so we are still fully inter-operable) as a solution: DESCRIBE_SETUP_PLAY.
We could send this one command, pass in types of streams interested in (typically, there's only one video and 0..1 audio). Response would include the full SDP text, as well as all the port information and just like before, server would start streaming instantly without waiting for anything else from the client.
Would this work? any downside that I may not be seeing? I'm curious why this wasn't considered (or was dropped) from official spec, since latency even in local intranet is definitely noticeable.
FYI, it is possible according to the RTSP 1.0 specification:
9.1 Pipelining
A client that supports persistent connections or connectionless mode
MAY "pipeline" its requests (i.e., send multiple requests without
waiting for each response). A server MUST send its responses to those
requests in the same order that the requests were received.
The RTSP 2.0 draft also contains support for pipelining.
However none of the clients/servers I've used implement it AFAIK.
I am asking this question as a small part of my question series regarding game programming .
Refer this question as the main one .
Now suppose I want to develop a game on iphone - a simple online multiplayer board game.
Suppose its a casino table.
In the main question ChrisF has tell me of sort of Client - Server architecture for iphone online multiplayer game.
I want to know what sort network programming I have to do for this type of application .
What will be the responsibilities and activities carried out by client and server .
You can provide me link, tutorials or to the point answers , anything will be great help for me and will be really appreciable .
thanks
You'll want to write a socket application running on a server. When you have access to a wifi access point or edge/3g you can send data to it from your iphone application. This server can then handle the incoming data and send an appropriate reply to the people connected.
For server socket programming, take a look at this guide - http://beej.us/guide/bgnet/.
For iphone specific socket programming, take a look at the samples supplied with the Iphone SDK. This link also has some basic information.
A simple online multiplayer board game
Given that the iPhone isn't always connected to the internet, you might need a server to store state. Alternatively you could always stipulate that if one person loses their connection the game finishes.
Client to client would be the obvious choice for the latter. Both clients have a port they listen to, and send the other commands based on the board state. Like almost all online games the obvious choice would be to use UDP as it's fast and compact.
For the server architecture you will of course need some kind of server listening for commands and a game number. It would store your state in a datastore on the server, a mySQL database for example. UDP or even SOAP or JSON over HTTP would be the two obvious choices for this.
This second approach using JSON/SOAP route would be a lot easier for you to get started with, assuming the iPhone has a decent JSON or SOAP library which is not likely. I have no idea about UDP in Objective C, however in C it requires a certain level of knowledge which won't get you something to play with quickly.
As you already said, you will need a server, but you can have two kinds of design:
The server can serve only as a gateway between the players to connect one to each other: it's two uses are, first, to list the running games, and second, to list the IP addresses of the players so that each client will read the IP addresses and connect to them. This will require less bandwidth and processing power for the web server and more for the client which will host the game. In this configuration, one of the clients will act as a server and will send data to the others. If your game has only one of the players playing at a time, and not a huge lot a players, this is what you should use as what you pay is the server's power.
The server can also store all games' states: it might require far more processing power and/or bandwidth depending on your game.
Anyway, most of the time you will want only one machine (which can change during the game as in the first case) to do the processing and the others will only receive the computed data.
To program a networked game, you will need knowledge of sockets (deep knowledge in the first case because you will have to deal with NAT issues, routers blocking the way between clients). The guide to that is Beej's Guide to Network Programming, the number one ressource on this topic, although it doesn't focus on games.
If not too much processing is needed on the WWW server, you could deal with it with server scripting languages like PHP along with MySQL, but you're most likely to use your own server programmed in C++ (or in C). If you use C++, you might want to use an already existing library such as RakNet.
The client will obviously be programmed in Objective-C as it's on the iPhone. I believe there is a good networking framework looking at the number of online games, so you might as well not want to use an external server networking library.
It might be a bit too much for what you want to do, but you could also use the well known Torque Engine.
We need is to push sports data to a number of different client types such as ajax/javascript, flash, .NET and Mac/iPhone. Data updates need to only be near-real time with delays of several seconds being acceptable.
How to best accomplish this?
The best solution (if we're talking .NET) seem to be to use WCF and streaming http. The client makes the first http connection to the server at port 80, the connection is then kept open with a streaming response that never ends. (And if it does it reconnects).
Here's a sample that demonstrates this: Streaming XML.
The solution to pushing through firewalls: Keeping connections open in IIS
I would go with XML. XML is widely supported on all platforms and has lots of libraries and tools available for it. And since it's text, there are no issues when you pass it between platforms.
I know JSON is another alternative, but I'm not familiar enough with it to know whether or not to recommend it in this case.