I want to play an audio on WebSite but link of audio have to secret or encrypted on NodeJS
example: Spotify etc.
What I would suggest is to have your socket.io server which can send you an appropriate audio file for your request.
Once you are connected to socket sever send the name of the audio file to your server and the server will send the audio file over the socket.
After receiving the file you can play as you want.
This way you will not expose the actual URL of the audio file because that is handled by your server.
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 working on a simple RTSP server to emulate an IP Camera but instead stream a jpeg image from a file. I have been working through the rtsp protocol and cant find any specific data on what payload I should set in my DESCRIBE response. Any good documentation would be appreciated
Thanks
Matt
Here you go. It's the complete, official definition of RTSP: https://www.rfc-editor.org/rfc/rfc2326
Alternative, look at ffmpeg and/or VLC source code for a reference impl.
Btw, all this information can be obtained here: http://en.wikipedia.org/wiki/Real_Time_Streaming_Protocol
...which is the first link on google when you search RTSP.
The IANA the lists RTP payload type 26 for JPEG. You'll want to specify this in your media attribute of the SDP message in your response. See Appendix C.1.2 of the RTSP RFC for information on media streams. For additional information, see section 8.1 of the SDP RFC.
An example would be:
m=video 0 RTP/AVP 26
Be sure to reference RFC 2435 for the RTP payload format that should be used for your RTP packets.
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.
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!