I'm having a problem here. I want to play RSTP streaming on ipad and iphone. but I find out that it will be much more easier if I use Http Live Streaming. I want to convert my RTSP streaming to Http live streaming using lighttpd. but I really have no idea how to do that. Do lighttpd accetp rtsp streaming url as input? Can anyone help? thanks!
you have two choices:
1) Run a server on your network that re-streams rtsp as hls.
a) wowza - popular, expensive
b) live 555 - free, lots of work
d) ffserver - free and as basic as it gets tons of work to make work.
Advantage :
No bandwidth restrictions over cellular or wi-fi
play with native apple players
Disadvantage
High server bandwidth - if your paying for server time you may want to watch this.
high letancy - forget any kind of live video.
2) Run FFmpeg based player on device
advantages :
a) A lot easier than it used to be, we do this all the time
b) deal with lgpl license, clear guidelines at ffmpeg.org and not a huge hassle
c) all on device, no server load issues.
Disadvantages
Limited bandwidth over celluar (about 10 min intervals), unlimited over wifi
lighttpd doesn't accept RTSP as an input. You will need some sort of translator program to read the rtsp stream and output the files to the website storage. I think you could do it with the avconv/ffmpeg program.
Related
I have a live video chat application and I use a TURN server which supports STUN/TURN and both UPD/TCP transmission.
Sometimes users can be connected to the network which blocks that much ports and protocols that WebRTC connection just cannot happen (usually those are corporate networks). I would like to check if a WebRTC connection is possible before users try to connect to each other (actually, perform a technical check).
How can I do it? Ideas I have in my head:
Try to download a hosted chunk of data (audio file, for example) via WebRTC - is it possible and would this be enough to make sure both inbound and outbound connections are open?
Use a TURN server as a host to make a connection to and see if it fails (have no idea if I can do it or not)
Use Flash to try to download/upload a chunk of data over specific ports and protocols. May be even using Cirrus. However, I am not sure this test will be accurate from WebRTC prospective.
Any other ideas?
Additional requirement: the checking technique must support Chrome, Opera and Firefox. Preferably also IE/Safari via Temasys plugin.
Edition 1 - gathering ICE candidates is a good idea, however, it is not 100% reliable. Once I checked logs in my application and it actually gathered relay ICE candidates, but video/audio transmission failed. Tested on Apprtc as well and got same results.
The best way to check is to connect with just a data channel first. Your users won't notice. If that works then audio and video are almost guaranteed to work. As a bonus, you can use the data channel for signaling for super-fast connecting when your users are ready.
the typical WebRTC approach to this is to create a peerconnection with STUN and TURN servers, call createOffer and setLocalDescription and watch the candidates gathered. See e.g. http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/
If you get srflx candidates, your stun server works (i.e. UDP is not blocked). More interesting is whether you get relay candidates. If you do, using TURN as a fallback will work. Quality might suffer if TURN/TCP is used. If you don't get relay candidates... calls are very unlikely to work.
I am trying out VOD streaming with latest VLC as described in this wiki:
https://wiki.videolan.org/Documentation:Streaming_HowTo/VLM/ and trying to stream an HD movie file with extension mp4.
i am able to open the stream in VLC running on the same machine, but it takes lot of time to even start playing back. (minutes). The test is run in Fedora 20.
My machine is capable enough to play HD video.
I have tried with and with trans-coding and results are same.
Help is appreciated.
Found the issue and it is strange.
I had the streaming server and client in the same machine. The streaming was always extremely slow. However, disconnecting from the network made it work fast.. It did not matter whether the network connection was wifi or ethernet.
However, if you were to run the client in a different machine, it work properly.
I'm looking to find a way to stream a user's desktop LIVE (through some piece of software, such as Open Broadcaster Software) to a web application.
I'm assuming I should use a CDN to get the live streamed video to my web application, but how (and what software should I use) to get the user's desktop to a streaming service? Should I use a service such as Red5 or an AWS service? Or if only a few viewers are using it, should I host the service myself?
Although I have built my share of web applications, I have never dealt with live media streaming before, and I would appreciate any assistance anyone could lend.
By far the best resource for video on Rails is OpenTok
Our own demo here: http://bvc-video.herokuapp.com/broadcasts/1
--
Streaming
Video streaming is a tough one
The problem is really dependent on what you're trying to stream. If it's "live" video - I.E captured & sent directly to the viewers, you'll have to use some sort of server to process the video.
Although I don't have huge experience with this, the main issue we've found is the compression / distribution of the feed. It's actually very simple to acheive video streaming on iOS - all the software / hardware is the same (just use the same API / drivers)
This often negates the requirement for a central server, although it's highly recommended (almost required) for many cases. Problems arise when you try and beam to multiple clients on multiple systems; as you'll run into compatibility issues
--
Solutions
The solutions we've found are thus:
The most stable part of the app is to take the stream & send to a server
The wizardry will then be to beam that stream to multiple clients
The way to do this is typically to use a flash widget & pull the stream from the server
WebRTC is becoming the standard (OpenTok is built on this)
I'm not sure about video compression / distribution. Akami is an industry heavyweight, but I've never used it. Brightcove too
I am designing an iOS app for a customer who wants to allow real-time (with minimum lag, max 50ms) conversations between users (a sort of Teamspeak). The lag must be low because the audio can also be live music, played with instruments, so all the users need to synchronize. I need a server, which will request audio recordings to every client and send to others (and make them hear the same sound at the same time).
HTTP is easy to manage/implement and easy to scale, but very low-performing because an average HTTP request takes > 50ms... (with a mid-level hardware), so I was thinking of TCP/UDP connections kept open between clients and server.
But I have some questions:
If I develop the server in Python (using TwistedMatrix, for example), how are its performance ?
I can't develop the server in C++ because it is hard to manage (scalable) and to develop.
Anyone used Nodejs (which is easy to scale) to manage TCP/UDP connections?
If I use HTTP, will it be fast enough with Keep-Alive? Becuase usually the time required for an HTTP Request to be performed is > 50ms (because opening-closing connection is hard), and I want the total procedure to be less than that time.
The server will be running on a Linux machine.
And finally: which type of compression can you suggest me? I thought Ogg Vorbis would be nice, but if there's anything better (and can be used in iOS), I am open to changes.
Thank you,
Umar.
First off, you are not going to get sub 50 ms latency. Others have tried this. See for example http://ejamming.com/ a service that attempts to do what you are doing, but has a musically noticeable delay over the line and is therefore, in the ears of many, completely unusable. They use special routing techniques to get the latency as low as possible and last I heard their service doesn't work with some router configurations.
Secondly, what language you use on server probably doesn't make much difference, as the delay from client to server will be worse than any delay caused by your service, but if I understand your service correctly, you are going to need a lot of servers (or server threads) just relaying audio data between clients or doing some sort of minimal mixing. This is a small amount of work per connection, but a lot of connections, so you need something that can handle that. I would lean towards something like Java, Scala, or maybe Go. I could be wrong, but I don't think this is a good use-case for node, which, as I understand it, does not do multithreading well at this time. Also, don't poo-poo C++, scalable services have been built C++. You could also build the relay part of the service in C++ and the rest in whatever.
Third, when choosing a compression format, you'll have to choose one that can survive packet loss if you plan to use UDP, and I think UDP is the only way to go for this. I don't think vorbis is up to this task, but I could be wrong. Off the top of my head, I'm not sure of anything that works on the iPhone and is UDP friendly, but I'm sure there are lots of things. Speex is an example and is open-source. Not sure if the latency and quality meet your needs.
Finally, to be blunt, I think there are som other things you should research a bit more. eg. DNS is usually cached locally and not checked every http call (though it may depend on the system/library. At least most systems cache dns locally). Also, there is no such protocol as TCP/UDP. There is TCP/IP (sometimes just called TCP) and UDP/IP (sometimes just called UDP). You seem to refer to the two as if they are one. The difference is very important for what you are doing. For example, HTTP runs on top of TCP, not UDP, and UDP is considered "unreliable", but has less overhead, so it's good for streaming.
Edit: speex
What concerns the server, the request itself is not a bottleneck. I guess you have sufficient time to set up the connection, as it happens only in the beginning of the session. Therefore the protocol is not of much relevance.
But consider that HTTP is a stateless protocol and not suitable for audio streaming. There are a couple of real time streaming protocols you can choose from. All of them will work over TCP or UDP (e.g. use raw sockets), and there are plenty of implementations.
In your case, the bottleneck with latency is not the server but the network itself. The connection between an iOS device and a wireless access point (AP) eats up about 40ms if the AP is not misconfigured and connection is good. (ping your iPhone.) In total, you'd have a minimum of 80ms for the path iOS -> AP -> Server -> AP -> iOS. But it is difficult to keep that latency stable. (Typical latency of AirPlay on my local network is about 300ms.)
I think live music over iOS devices is not practicable today. Try skype between two iOS devices and look how close you can get to 50ms. I'd bet no one can do it significantly better, what concerns latency.
Update: New research result!
I have to revise my claims regarding the latency of wifi connections of the iDevice. Apparently when you first ping your device, latency will be bad. But if I ping again no later than 200ms after that, I see an average latency 2ms-3ms between AP and iDevice.
My interpretation is that if there is no communication between AP and iDevice for more than 200ms, the network adapter of the iDevice will go to a less responsive sleep mode, probably to save battery power.
So it seems, live music is within reach again... :-)
Update 2
The ping-interval required for keep alive of low latency apparently differs from device to device. The reported 200ms is for an 3rd gen. iPad. For my iPhone 4 it's more like 50ms.
While streaming audio you probably don't need to bother with this, as data is exchanged on a more frequent basis. In my own context, I have sparse communication between an iDevice and a server, but low latency is crucial. A keep alive therefore is the way to go.
Best, Peter
I am building an iOS App which displays video streams from a somewhat complex backend. Now while developing I want to be able to have some sort of test video stream, which I can use. Ideally this would also work without internet connection.
The video stream could show for example the current time or just a simple animation. What would be a good way of doing this on a Mac without having to install a whole suite of tools.
On you Mac you can setup a webserver or streaming server to provide you with a constant video stream for testing purposes. You won't need Internet access. You will, of course, need to ensure that the OSX firewall is either disabled or allows requests to the ports (80, most likely).
Two simple approaches I can see:
Wowza MPEG-TS stream of the Webcam on your mac
Install Wowza Media Server; developer license is free
Configure a basic applicaiton with MPEG-TS streaming
Use an encoding applicaiton, like Flash Media Live Encoder (free), Wirecast (demo version free), or some other software and start streaming from your webcam to the WMS
alternatively, with a bit more effort, you could setup Wowza to stream a file in a loop
be sure to get the codec settings correct
M3U8+MPEG-TS static files over plain HTTP
Simple Setup a basic webserver (lighttpd, Apache httpd, Apache Tomcat, whatever) to server static files
Whip up an M3U8 file to first point to a .ts media file, and then secondly back to itself
Have a look at MPEG-TS/M3U8 live stuff to work out the details. You'll need a properly segmented video file to start with.