I need to stream a video which is located in a local network between two computers using C#. I am thinking to use TCP rather than UDP since I need reliability. I already build a simple program for data transfer with TCP/IP. Now I want to build a TCP communication for video streaming. Is there any sample code or suggestion for both the server and client side for that?
I already try to open video in local network and tried to encode the video same as I do on data transfer but it didn't work.
Thanks.
I'm facing problem with restreaming .h264 video received from my device via tcp to wowza streaming engine. The problem is that I do not know how to forward byte array (byte[]). I have read that it is possible via rstp/rtmp/mpegts but I have not found any library to do this operation. I know that video I receive is ok because after saving frames to file I'm able to send it to wowza using ffmpeg. I have been also trying to use ffmpeg to listen on udp ip and port and on http ip and port but nothing happened.
My question is:
Is it possible to send bytes to ffmpeg without saving file on hard drive?
I an trying to develop a windows application using C# that can play streamed audio data. Basically, I will have a client application that is responsible for playing different audio files. Currently, from the client application, I am extracting the hardware config param from the file header and then will stream the file data (PCM stream) over network.
So is it possible to use the hardware config params sent from client to configure the actual hardware (on the server end) and then give it the file data stream to it so that it can play the audio data.
While searching, I got to know about NAudio. Is NAudio capable of doing this stuff or the better option for me would be to switch to nativ C/C++ code using Directsound APIS.
update:
By configuring hardware, I mean setting the param related to audio playback. These param would include sample rate (eg: 44100 Hz), number of channels (eg: stereo), storage format (eg: 16 bit little endian) etc.
My client application is on Linux and I have planted an ALSA driver that intercepts PCM stream and hw_params configuration and then send them to server.
update ends
Thanks.
If you look at the latest NAudio code, you will see there are two examples in the NAudioDemo sample app that play streaming audio. One is a rudimentary chat application that sends compressed voice via UDP, the other plays streaming MP3 internet radio. I'd suggest you have a look at that and try the sample app to see if it meets your needs.
When you have a network path open in Windows explorer, and you drag it to a local folder does it open a socket? Also, when you use c# FileStream fin = new FileStream(#"//networkpath/file); does that open a socked? my question is this, would it be just as fast to stream a file over a socket manually as it would be to read it over a network using c#'s filestream?
The Windows file service works over TCP/IP by default (although not necessarily), so typically, there's a socket involved. Yes, there's some overhead from the SMB protocol that Windows uses. However, for files where transfer time matters, the overhead is small compared to the data.
In addition, coming up with your own file sharing protocol without a very good reason is a bad idea. It's a lot of development and debugging work, you have to install the server part somehow, you have to think of security implications (user authentication, etc), firewalls will break it... Just not worth it.
To gauge the amount of work involved, read the description of the FTP protocol.
We need to capture a live video stream from WebRTC (or any other capturing mechanism from the client webcam, even if it is not supported on all browsers, but as a PoC).
This live video needs to be handled by a server component (ASP.Net MVC / Web API), I imagine that the code on the server will look like:
[HttpPost]
public ActionResult HandleVideoStream(Stream videoStream)
{
//Handle the live stream
}
Looking for any keyword or helpful link.
We have already implemented a way to send individual frames using base64 jpg, but this is not useful at all, because there is a huge overhead of the base64 encoding and because we could use any video encoding to send the video more efficiently (send the difference between the frames using VPx -vp8- for example), the required solution needs to capture a video from the webcam of the client and send it live (not recorded) to the server (asp.net) as a stream -or chunks of data representing the new video data-.
Your question is too broad and asking for off-site resources is considered off-topic on stackoverflow. In order to avoid opinion-prone statements I will restrict the answer to general concepts.
Flash/RTMP
WebRTC is not yet available on all browser so the most widely used way of capturing webcam input from a browser currently in use is via a plugin. The most common solution uses the Adobe Flash Player, whether people like it or not. This is due to the H.264 encoding support in recent versions, along with AAC, MP3 etc. for audio.
The streaming is accomplished using the RTMP protocol which was initially designed for Flash communication. The protocol works on TCP and has multiple flavors like RTMPS (RTMP over TLS/SSL for encryption), RTMPT(RTMP encapsulated in HTTP for firewall traversal).
The stream usually uses the FLV container format.
You can easily find open-source projects that use Flash to capture webcam input and stream it to an RTMP server.
On the server-side you have two options:
implement a basic RTMP server to talk directly to the sending library and read the stream
use one of the open-source RTMP servers and implement just a client in ASP (you can also transcode the incoming stream on the fly depending on what you're trying to do with your app).
WebRTC
With WebRTC you can either:
record small media chunks on a timer and upload them on the server where the stream is reconstructed (needs concatenating and re-stamping the chunks to avoid discontinuities). See this answer for links.
use the peer-to-peer communication features of WebRTC with the server being one of the peers.
A possible solution for the second scenario, which I haven't personally tested yet, is offered by Adam Roach:
Browser retrieves a webpage with javascript in it.
Browser executes javascript, which:
Gets a handle to the camera using getUserMedia,
Creates an RTCPeerConnection
Calls createOffer and setLocalDescription on the
RTCPeerConnection
Sends an request to the server containing the offer (in SDP format)
The server processes the offer SDP and generates its own answer SDP,
which it returns to the browser in its response.
The JavaScript calls setRemoteDescription on the RTCPeerConnection
to start the media flowing.
The server starts receiving DTLS/SRTP packets from the browser,
which it then does whatever it wants to, up to and including storing
in an easily readable format on a local hard drive.
Source
This will use VP8 and Vorbis inside WebM over SRTP (UDP, can also use TCP).
Unless you can implement RTCPeerConnection directly in ASP with a wrapper you'll need a way to forward the stream to your server app.
The PeerConnection API is a powerful feature of WebRTC. It is currently used by the WebRTC version of Google Hangouts. You can read: How does Hangouts use WebRTC.
Agreed that this is an off-topic question, but I recently bumped into the same issue/requirement, and my solution was to use MultiStreamRecorder from WebRTCExperiments. This basically gives you a "blob" of the audio/video stream every X seconds, and you can upload this to your ASP.NET MVC or WebAPI controller as demonstrated here. You can either live-process the blobs on the server part by part, or concatenate them to a file and then process once the stream stops. Note that the APIs used in this library are not fully supported in all browsers, for example there is no iOS support as of yet.
My server side analysis required user to speak full sentences, so in addition I used PitchDetect.js to detect silences in the audio stream before sending the partial blob to server. With this type of setup, you can configure your client to send partial blobs to server after they finish talking, rather than every X seconds.
As for achieving 1-2 second delay, I would suggest looking into WebSockets for delivery, rather than HTTP POST - but you should play with these options and choose the best channel for your requirements.
Most IP cameras these days will use H264 encoding, or MJPEG. You aren't clear about what sort of cameras are being used.
I think the real question is, what components are out there for authoring/editing video and which video format does it require. Only once you know what format you need to be in, can you transcode/transform your video as necessary so you can handle it on the server side.
There are any number of media servers to transform/transcode, and something like FFMPEG or Unreal Media Server can transform, decode, etc on server side to get it to some format you can work with. Most of the IP cameras I have seen just use an H264 web based browser player.
EDIT: Your biggest enemy is going to be your delay. 1-2 seconds of delay is going to be difficult to achieve.