I want to use an IP camera with webrtc. However webrtc seems to support only webcams. So I try to convert the IP camera's stream to a virtual webcam.
I found software like IP Camera Adapter, but they don't work well (2-3 frames per second and delay of 2 seconds) and they work only on Windows, I prefer use Linux (if possible).
I try ffmpeg/avconv:
firstly, I created a virtual device with v4l2loopback (the command was: sudo modprobe v4l2loopback). The virtual device is detected and can be feed with a video (.avi) with a command like: ffmpeg -re -i testsrc.avi -f v4l2 /dev/video1
the stream from the IP camera is available with: rtsp://IP/play2.sdp for a Dlink DCS-5222L camera. This stream can be captured by ffmpeg.
My problem is to make the link between these two steps (receive the rstp stream and write it to the virtual webcam). I tried ffmpeg -re -i rtsp://192.168.1.16/play2.sdp -f video4linux2 -input_format mjpeg -i /dev/video0 but there is an error with v4l2 (v4l2 not found).
Does anyones has an idea how to use an IP camera with webRTC?
Short answer is, no. RTSP is not mentioned in the IETF standard for WebRTC and no browser currently has plans to support it. Link to Chrome discussion.
Longer answer is that if you are truly sold out on this idea, you will have to build a webrtc gateway/breaker utilizing the native WebRTC API.
Start a WebRTC session between you browser and your breaker
Grab the IP Camera feed with your gateway/breaker
Encrypt and push the rtp stream to your WebRTC session from your RTSP stream gathered by the breaker through the WebRTC API.
This is how others have done it and how it will have to be done.
UPDATE 7/30/2014:
I have experimented with the janus-gateway and I believe the streaming plugin does EXACTLY this as it can grab an rtp stream and push it to an webrtc peer. For RTSP, you could probably create RTSP client(possibly using a library like gstreamer), then push the RTP and RTCP from the connection to the WebRTC peer.
Janus-gateway recently added a simple RTSP support (based on libcurl) to its streaming plugins since this commit
Then it is possible to configure the gateway to negotiate RTSP with the camera and relay the RTP thought WebRTC adding in the streaming plugins configuration <prefix>/etc/janus/janus.plugin.streaming.cfg
[camera]
type = rtsp
id = 99
description = Dlink DCS-5222L camera
audio = no
video = yes
url=rtsp://192.168.1.16/play2.sdp
Next you will be able to access to the WebRTC stream using the streaming demo page http://..../demos/streamingtest.html
I have created a simple example transforming a RTSP or HTTP video feed into a WebRTC stream. This example is based on Kurento Media Server (KMS) and requires having it installed for the example to work.
Install KMS and enjoy ...
https://github.com/lulop-k/kurento-rtsp2webrtc
UPDATE 22-09-2015.
Check this post for a technical explanation on why transcoding is just part of the solution to this problem.
If you have video4linux installed, the following command will create a virtual webcam from an rtsp stream:
gst-launch rtspsrc location=rtsp://192.168.2.18/play.spd ! decodebin ! v4l2sink device=/dev/video1
You were on the right track, the "decodebin" was the missing link.
For those who would like to get their hands dirty with some native-WebRTC, read on...
You could try streaming an IP camera’s RTSP stream through a simple ffmpeg-webrtc wrapper: https://github.com/TekuConcept/WebRTCExamples .
It uses the VideoCaptureModule and AudioDeviceModule abstract classes to inject raw media. Under the hood, these abstract classes are extended for all platform-specific hardware like video4linux or alsa-audio.
The wrapper uses the ffmpeg CLI tools, but I don’t feel it should be too difficult to use the ffmpeg C-libraries themself. (The wrapper relies on transcoding, or decoding the source media, and then letting WebRTC re-encode with respect to the ICE connections’ requirements. Still working out pre-encoded media pass-through.)
Actually our camera can support webrtc. It uses ip camera with h5, from P2P tramsmitting, and two way talk for ip camera with web browser! The delay is only 300ms!
Related
How can i send all webcams to collect from one server.
For example:
there is pc_1, pc2, ..., pc_n they are sending camera view to some ubuntu server where i can connect with
ssh name#ip_adress
and all pc have a windows on them
i looked Sending live video frame over network in python opencv this but this worked only on localhost
and secondly i looked this Forward RTSP stream to remote socket (RTSP Proxy?) but couldnt figure out how to do it on my situation
Each IPC is a RTSP server, it allows you to pull/play RTSP stream from it:
IPC ---RTSP--> Client(Player/FFmpeg/OBS/VLC etc.)
And because it's a internal IPC and its IP is intranet, so the client should in the same intranet, that's why it works only on localhost like.
Rather than pulling from the internet client which does not work, you could forward the stream to internet server, just like this:
IPC ---RTSP--> Client --RTMP--> Internet Server(SRS/Nginx etc.)
For example, use FFmpeg as a Client to do this, please replace the xxx by your internet server:
ffmpeg -i "rtsp://user:password#ip" -c:v libx264 -f flv rtmp://xxx/live/stream
Note: You could fastly deploy a internet server by srs-droplet-template in 3 minutes, without any cli or knowledge about media server.
Then you could play the stream by any client and any protocol, like PC/H5 by HTTP-FLV/HLS/WebRTC, mobile iOS/Android by HTTP-FLV/HLS, please read this post
I've successfully set up Asterisk on my server using the res_pjsip Hello World configuration from their wiki, and I want to be able to forward the RTP data to a Node.JS app, which can interpret RTP. I've heard about directmedia and directrtpsetup (see this stackoverflow) but I'm not sure if that's what I want. So my question is this:
Should I use directmedia / directrtpsetup to send voice data to my Node.JS app, or should I use some sort of Asterisk functionality to forward RTP packets? If the latter, how can Asterisk forward just the voice data?
I can clarify if needed, but hopefully this is more specific than my last questions. Thanks!
UPDATE: Having poked around Asterisk docs and messing with Wireshark, I think I have two options.
Figure out if there's a channel driver for Asterisk that just sends RTP, without any signaling, or
Capture the RTP stream with Wireshark or something and send the packets to the Node.JS app, and inject the return packets into the RTP stream.
Asterisk is PBX. It not suitable to "redirect rtp data"
No, there are no reason in having channel driver "without signalling". For what anyone can use it? How to determine call started if "no signaling"? It will be useless.
You can write such app in c/c++ or use other soft designed to be traffic capture: libpcap, tcpdump etc.
You also can use audio staff: libalsa, jack.
Best option however will be create or find full featured sip client and use it.
I have cross compiled live555 library for my android board and using that I can able to stream video board to any other device using rtsp protocol.
I have used "live555MediaServer" program for stream board to other devices. I got "rtsp" url and using that I am able to stream.
But now I want to push this video to wowza server which is running in our local cloud. while reading wowza document I found that you can use any supported encoder to stream video to wowza by registering wowza steaming engine in your encoder. I found following lines
In your encoder, enter the following information, and then click Publish or Start:
Server URL: rtsp://[wowza-ip-address]:1935/live
Stream Name: myStream
User: publisherName
password: [password]
so I want to know how can i register wowza ip with live555. In live555 is there any way we can register this wowza ip and stream video in wowza streaming engine ?
I found one application in "live555/testProgs/registerRTSPStream" while running this application with out any command line option it is showing below usage
usage: registerRTSPStream [-t] [-u <username> <password>] <remote-client-or-proxy-server-name-or-address> <remote-client-or-proxy-server-port-number> <rtsp-URL-to-register> [proxy-URL-suffix]
So Is it possible to register wowza server using this application and If yes then how to do it ?
I'm implementing a solution for listening to on-going calls inside a LAN network.
Is there a way to provide WebRTC the ip address and port as to where an RTP stream is coming? All I want to do is to get that RTP stream directly streamed to the possible listeners of the call through WebRTC.
I'm not sure if it's feasible but I think it is given how WebRTC has evolved since the past months.
I've been looking around but I've got no luck on this.
The WebRTC RTP stream is encrypted with keys that are exchanged through DTLS. You cannot get the raw RTP stream from a WebRTC peer or even feed it a raw stream without some mediary system to handle the webrtc peerconnection, certificate exchange, and rtp encryption.
The only way to do what you want is to have a breaker or a gateway. An example of such a gateway is the janus-gateway though it is definitely not your only option.
I am in search of a server software which can stream different audios to a different clients.
For example every client will be able to create his own playlist and the server will stream it
Any help will be appreciated
You can check flash which has support for RTMP to stream audio real time using client server & RTMFP which works over peer to peer technology. You can use RTMFP in case peer is directly reachable else use RTMP. There is a open source red5 media server which also has support for RTMP protocol.