I have been using Onvif for one month and I am able to receive stream URI and have the control over all the configuration stuff from my own client program designed in C#.
In my application I want to take the videos (1 or 2 min streams) from 10 IP Cameras and then create a 10 min video. So it is like embedding the videos from all cameras.
My question is - Can I use Onvif for this application ?
I am asking because I only found information about configuration stuff in all Onvif WSDL files. So I got a doubt whether I can use or not. Kindly requesting you to tell me the compatibility of Onvif with my specified application. I would be more glad if you also provide some information on how to make it possible.
You can use Onvif to configure the cameras for use with the application, however you would not use Onvif to actually acquire the video from the cameras.
You can use Onvif to configure the streams (Encoding format, multicast setup, network configuration, etc) and get the Uri for the stream (GetSreamUri), but you would then need to access the RTSP streams directly to get the actual video.
This can be done using something like ffdshow with Direct Show to grab the video from each camera and make a compilation.
Onvif has a Streaming Specification which describes how compliant cameras must implement streaming but it still results in the camera producing a video stream on the network. How clients end up acquiring the video is outside the scope of the specification.
Related
My primary intention is to setup a VoIP session between 2 users A & B; Here the raw audio / video media bytes are fetched from A's browser are played in B's browser and vice versa.
The reason is that, when the user C & D are added into this call, we need not have to create a P2P mesh network which limits the performance.
Tried recording media with getUserMedia() and playback, but it is not real time. It also gives a bad user experience. (However, haven't experimented yet with videos of small chunks as 200 ms)
Is there any approach where I can get the raw bytes of the media and play it on other browser? Currently I have a server in between which can connect to both peers if required.
Any online examples or libraries are welcome.
Have already asked 2 questions in this regard with 100-100 bounties, but not much of use:
How to use libsrtp or similar library to decrypt/encrypt the WebRTC data stream?
How to integrate part of WebRTC as a static / dynamic library with the existing C++ code?
Related: How to stream, live video playing on my browser to browser of another user?
If i understand you well is you're looking on how to have more than two users on the session right? without using mesh topology
thats possible and configurable as well by means that some maybe active speaker or everyone is active speaker not only receiver whatever configuration you choose but to me it seems that you're asking for video conferencing
there are couple of tools for this the best one i might recommend is mediasoup its a SFU as selective fowarding unit mediasoup
I don't know if I understand correctly, but it is not likely that you will get raw video data and play it on the browser, it will just kill your bandwith and performance because the raw data is huge.
You need to use the compressed data ( media codec ex.H264 ) and you need a protocol to send and receive it. If you are looking for sub-second latency than webrtc is your best choice in here already. If you have a server in between, distribute your media through that server instead of Mesh. Check this out for webrtc network topologies:
https://antmedia.io/webrtc-servers/
I am trying to build a device that will encode h.264 video on a raspberrypi and stream it out to a separate web server in the cloud. The main issue I am having is most implementations I search for either have the web server directly on the pi or have the embedded player playing video directly from the device.
I would like it to be pretty much plug and play no matter what network I am on ie no port forwarding of any sort all I need to do is connect the device to the network and the stream will be visible on a webpage.
One possible solution to the issue is just simply encode frames in base 64 as jpegs and send them to a an endpoint on the webserver, however, this is a huge waste of bandwidth and wont allow for the framerate h.264 would.
Any idea on some possible technologies that could be used to do this?
I feel like it can be done with some websockets or zmq and ffmpeg somehow but I am not sure.
It would be helpful if you could provide more description of the architecture of the device. Since it is an RPI, it is probably also being used for video acquisition via the camera expansion port. If this is the case, you can access the video device and do quite a bit with respect to streaming using the combination of available command line tools.
Something like the following will produce an RTMP stream from the video camera host.
raspivid [preferred options] -o - | ffmpeg -i - [preferred options] rtmp://[IP ADDR]/[location]
From there, FFmpeg will do a lot of heavy lifting for you.
This will now enable remote hosts to access the RTMP stream.
Other tools that would complement that architecture may be ffserver where the rtmp stream from the rpi host could be acquired and then be made available to a variety of clients such as a player in a webpage. Quick look shows ffserver may be obsolete, but that there are analogous components.
I am looking to build an app that needs to process incoming audio on a phone call in real time.
WebRTC allows for this but i think this works only in their browser based P2P audio communications functionality but not for phone calls/ VOIP.
Twilio and Plivo allow you record the audio for batch/later processing.
Is there a library that will give me access to the audio streams in real time? If not, what would I need to build such a service from scratch?
Thanks
If you are open to using a media server (so that the call is not longe P2P but it's mediated by the media server using a B2B model), then perhaps the Kurento Media Server may solve your problem. Kurento Media Server makes possible to create processing capabilities which are applyied in real time onto the media streams. There are many examples in the documentation of computer vision and augmented reality algorithms applied in real time over the video streams. I've never seen an only-audio processing module, but it should be simple to implement just by creating an additional module, which is not too complex if you have some knowledge about C/C++ and media processing concepts.
Disclaimer: I'm part of the Kurento development team.
i have just started to delve into streaming libraries and the underlying protocols. I understand rtsp/rtp streaming and what these 2 protocols are for. But if we need the ip address, codec and the rtsp/rtp protocols to stream the video and audio from any cameras then why do we have onvif standard which essentially also aims to standardize the communication between IP network devices. I have seen the definitions of onvif so thats not what I am looking for. I want to know why at all we need onvif when we already have rtsp/rtp and what additional benefits it can provide.
ONVIF is much more than just video streaming. It's an attempt to standardize all remote protocols for network communication between security devices. This includes things like PTZ control video analytics and is much more than just digital camera devices.
I am working on a project which will involve http live media streaming from a variety of devices like android phones/tablets, iphone, ipad, browser,etc. It will be a 2 way communication for all the devices with multiple devices connected to a conversation. I have implemented it partially i.e. one way by capturing audio from android phone(native app) and streaming to a web browser(HTML5 app) with a PHP server using ffmpeg and cvlc. I wanted to know of the best way to go ahead about it. Like, if there are any standards to be followed. Also what kind of a server should I be using? I don't want to use any streaming servers like Red5. I would like to implement the streaming logic similar to Http LiveStreaming by apple. I have come across MPEG-DASH that seems to be a standard for http streaming. I still have to look deeper into it. I was also thinking of using NodeJS for its popularity with streaming. Another worry was how do I go about capturing of media from devices? As in, should I use the native capability of the devices to convert media into an mp4 or any container that it supports and then stream it to the server or capture audio and images for a particular period of time and then send it to server and create a common output(I am not really sure of this idea). The separate capture is basically for simplifying the process of video streaming from the server end to any device. I was also thinking if I could completely bypass the server in any cases like a phone to phone or phone to tablet connection.
I just wanted to be sure of the things I will be using/implementing so that I wouldn't have to make drastic changes later on. Any help is deeply appreciated. Thank you.