I'm currently researching different streaming methods both for live and on-demand streaming.
I've read about both multicast and unicast, and now I got the following question, which I can not find an answer to.
"Is it possible to make on-demand streaming with multicast?"
The way I understand it is, that when using multicast, the media server creates a stream of the video, which only is played once, which users can connect to and watch.
It it because multicast only allows live streaming? If not can someone please explain to me how it works?
"Is it possible to make on-demand streaming with multicast?"
Technically, yes. Practically, no.
The way I understand it is, that when using multicast, the media server creates a stream of the video, which only is played once, which users can connect to and watch.
You understand it correctly. And that is that.
Well, you can do it, but the bigger question is why would you want it?
On-demand suggests that you start the broadcast at the time that a single viewer wants to see that particular piece of content. If a single user chooses the content and the time it is started, why would you want to multicast it?
Yes, it can be done, but there are caveats. If you take a flight on an old plane you may see an old entertainment system that offers say 20 channels with a movie on each. The channels are all rolling and once the programmes have finished they restart. This is better than having just one channel broadcast on a projector as it gives the user choice of what to watch but doesn't give them the freedom of when to watch.
Modern flight entertainment systems are all on-demand, every passenger can watch any film at any time. So how can multicast help there is the question? If you detect that multiple users are watching the same film, and the caveat being at the same time, you can replace the streams to each user with just one multicast channel. Which is technically savvy but you have to ask why would you do this? This only makes sense if the communication medium is feeliable or insufficient to serve every user simultaneously.
Designing a flight entertainment system that does not scale to every passenger actually using it is a bit short sighted. Therefore the system can handle the worst case of a stream for each user, meaning there is no benefit for multicasting anything.
Some cable/satellite networks implement multicast streaming and use time windows to group as many viewers together as possible. For example wait up to 5 minutes to watch a video whilst displaying the infamous phrase "buffering".
Related
We're writing a streaming service using wowza as our streaming server with a flash applet on the client. A recent concern we've had is a single client accessing the same stream from multiple tabs. In this scenario, we want to save bandwidth by sending the client only one copy of the stream.
How is this problem typically handled?
I think this is not possible for multiple technical reasons. This is my personal opinion, so experts can correct me if I am wrong)
The flash players on multiple tabs are not aware of each other and they can't be associated with each other on Wowza Streaming Engine side either. Wowza would see 2 different clients from the same IP, but that might be two different devices behind the same router.
The browser tabs are usually isolated from each other in the browser implementations for security and stability reasons, not allowing cross-browser data sharing
The 2 players are not in perfect sync, started at different points in time and lags, buffering might also introduce time shift
I haven't seen any attempt to reduce bandwidth in this scenario. There are solutions to prevent users from playing multiple streams in parallel though, which I would not like to link here, but do a search for "wowza forbidding concurrent stream" and you'll find it.
I think what you thought of is multicast streaming, where network packets are broadcast over the network and every player receives the same stream. This is used in IP TV systems, but that is not possible over internet, only on intranet.
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 am about to develop a service that involves an interactive audio live streaming. Interactive in the sense that a moderator can have his stream paused and upon request, stream audio coming from one of his listeners (during the streaming session).
Its more like a Large Pipe where what flows through but the water can come in from only one of many small pipes connected to it at a time with a moderator assigned to each stream controlling which pipe is opened. I know nothing about media streaming, I dont know if a cloud service provides an interactive programmable solution such as this.
I am a programmer and I will be able to program the logic involved in such interaction. The issue is I am a novice to media streaming, don't have any knowledge if its technologies and various software used on the server for such purpose, are there any books that can introduce on to the technologies employed in media streaming, and I am trying to avoid using Flash,?
Clients could be web or mobile. I dont think I will have any problem with integrating with client system. My issue is implementing the server side
You are effectively programming a switcher. Basically, you need to be able to switch from one audio stream to the other. With uncompressed PCM, this is very simple. As long as the sample rates and bit depth are equal, cut the audio on any frame (which is sample-accurate) and switch to the other. You can resample audio and apply dithering to convert between different sample rates and bit depths.
The complicated part is when lossy codecs get involved. On a simlar project, I have gone down the road of trying to stitch streams together, and I can tell you that it is nearly impossible, even with something as simple as MP3. (The bit reservoir makes things difficult.) Plus, it sounds as if you will be supporting a wide variety of devices, meaning you likely won't be able standardize on a codec anyway. The best thing to do is take multiple streams and decode them at the mix point of your system. Then, you can switch from stream to stream easily with PCM.
At the output of your system, you'll want to re-encode to some lossy codec.
Due to latency, you don't typically want the server doing this switching. The switching should be done at the desk of the person encoding the stream so that way they can cue it accurately. Just write something that does all of the switching and encoding, and use SHOUTcast/Icecast for hosting your streams.
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 developing an iPhone application that allows to basically click through a series of actions. These series are predefined and synced with a common configuration server.
That app might be running on multiple devices at the same time. All devices are assumed to have the same series of actions defined on them. All devices are considered equal, there is not a server and multiple clients or something like that.
(Only) one of these devices is used by a person at any given time, it is however possible that the person switches to a different device at any given time. All "passive" devices need to be synchronized with the active one, so that they display the same action.
The whole thing should happen as automatically as possible. No selection of devices, configuration, all devices in the same network take part in the same series of actions.
One additional requirement is that a device could join during a presentation (a series of actions) and needs to jump to the currently active action.
Right now, I see two options to implement the networking/communication part of that:
Bonjour. I have implemented a working prototype that can automatically connect with one (1) other device in the network and communicate with that. I am not sure at this point how much additional work the "multiple devices" requirement is. Would I have to open a set of connections for every device and manually send the sync events to all of them? Is there a better way or does bonjour provide anything to help me with that? What does Bonjour provide given that I want to communicate with every device in the network anyway?
Multicast with AsyncUdpSocket. Simply define a port and send multicast sync events out to that port. I guess the main issue compared to using bonjour with tcp would be that the connection is not safe and packets could be lost. This is however in a private, protected wlan network with low traffic if that would really be an issue. Are there other disadvantages that I'm not seeing? Because that sounds like a relatively easy option at this point...
Which one would you suggest? Or is there another, better alternative that I'm not thinking of?
You should check out GameKit (built in to iOS)--they have a lot of the machinery you need in a convenient package. You can easily discover peers on the network and easily send data back for forth between clients (broadcast or peer to peer)
In my experience Bonjour is perfect for what you want. There's an excellent tutorial with associated source code: Chatty that can be easily modified to suit your purposes.
I hobbled together a distributed message bus for the iphone (no centralized server) that would work great for this. It should be noted that the UI guy made a mess of the code, so thar' be dragons there: https://code.google.com/p/iphonebusmiddleware/
The basic idea is to use bonjour to form a network with leader election. The leader becomes the hub through which all the slaves subscribe to topics of interest. Then any message sent to a given topic is delivered to every node subscribed to said topic. A master disconnection simple means restarting the leader election process.