This is not a problem for browsers, but for Kurento Media Server, which refuses to accept this. I think it is in accordance to the RFC https://datatracker.ietf.org/doc/html/rfc3264
Is there any way to make it use another port?
The way to go is to munge the SDP and change the port to something different
Related
I have been trying to connect two machines: both Virtual Machines, one is Xubuntu and the other is Ubuntu. I'm also very new to OpenDDS, but the best way -or so it seems- to do it is to use the .ini files.
However, when I try to connect, I seem to fail in changing the Discovery Server, since the default is localhost:12345. Can somebody help me with that so I can configure the file properly?
I have tried using the dds_udp_conf.ini and the tcp one, but it doesn't seem to work.
Also, I tried using unicast, but failed.
the ini file:
[common]
DCPSDebugLevel=0
DCPSInfoRepo=corbaloc::localhost::12345/DCPSInfoRepo
DCPSGlobalTransportConfig=config1
[config/config1]
transports=udp1
[transport/udp1]
transport_type=udp
And I use the syntax:
./publisher -DCPSConfigFile conf.ini
Well, the publisher and subscriber are supposed to connect, but the publisher sends some error messages and in the other VM nothing happens.
I seem to fail because I cant change the configuration in the localhost for discovery.
When I try to run the server with a different parameter than localhost:12345 it always sends error messages too.
It's unclear to me where you're running the InfoRepo if both the publisher and subscriber are told the InfoRepo is running at localhost. Regardless I would recommend using the RTPS discovery and transport instead. It's easy to set up because the participants can find each other through the network's multicast without InfoRepo. This config is the simplest way to use RTPS with OpenDDS:
[common]
DCPSDefaultDiscovery=DEFAULT_RTPS
DCPSGlobalTransportConfig=$file
[transport/the_rtps_transport]
transport_type=rtps_udp
Just give this to both the programs and they should find each other. If not that would mean there's probably something's wrong with how the networking is set up on your VMs.
I have small computer (something like Arduino or Raspberry pi) with Linux, camera and gstreamer installed on it.
I need to stream h264 video from this device to browser using WebRTC technology. Also, I use NodeJS as signaling server.
In simple words, I need to doing a WebRTC client from my device. What is the best way to do this? Can I use WebRTC Native API for this goal? How can I install it on my small device? Or, maybe, I just need to play with my gstreamer and install some webrtc plugins for it?
Since you will have to use a signalling server anyways, I would say you should use the Janus-Gateway. You mention CentOS for your signalling server, I am not 100% if it will run on CentOS specifically, but I have ran it successfully in Debian Jessie build with just a few dependency installations.
Janus handles the entire call set up with the gateway(signalling and everything). So, some port forwarding will probably have to be done so that the SDP exchange can occur(which you would have to worry about with any signalling server).
Install the gateway, there are a few dependencies but all were simple
installations
Take a look at the janus_streaming plugin. It has a gstreamer example that will stream from a gstreamer pipeline. Also, the streamingtest demo page to see how the Javascript API works for that plugin
The plugin listens on those ports given in the configuration file and will accept traffic from any IP address. So, I expect you can run a gstreamer pipeline on a different machine on the same network and send it to the plugin.
NOTE: You will have to modify the SDP that the JavaScipt sends to the gateway so that it includes H264(probably get rid of all other codecs as well just to force negotiation). You can do this by accessing the sdp through the jsep object passed to the success case for the createOffer function in the janus JavaScript API(jsep.sdp).
Another possibility for you is to use the Kurento Media Server (KMS), which has been written on top of GStreamer. I see two possibilities
You install KMS in a Ubuntu 14.04 box and bridge with your device, so that the device generates the video stream and sends it to the KMS box. From that, you can transcode it to VP9 and distribute it as a WebRTC stream quite easily using kurento client APIs (which may be used from Node.js). The application making the transcoding will require an RtpEndpoint (receiving video form the device in RTP/H.264) connected to a WebRtcEndpoint (capable of sending the video stream through WebRTC). This option is quite simple to implement because it's the standard way of using KMS. However, you will need to generate the RTP/H.264 stream on the device and appropriate SDP for it (this can be done using standard GStreamer elements)
You try to install KMS into your box directly. This might be more complex because it requires compiling KMS to the specific device, which may require some time investment. In addition, performing the transcoding in the device might be too expensive and you could starve its CPU.
Disclaimer: I'm member of the Kurento development team
You mentioned that you used a NodeJS signaling server. Recently Ericsson released an open source WebRTC gstreamer element: http://www.openwebrtc.io/, and along with their release they also published a WebRTC demo using node.js: http://demo.openwebrtc.io:38080/; the code here: https://github.com/EricssonResearch/openwebrtc-examples/tree/master/server.
For WebRTC for Raspberry Pi 2 you may want to consider UV4L. It allows you to stream live Audio & Video from the Rpi to any browser on a PC (HTML5).
I would like to make of some sort of a streaming server. I would like it to receive RTSP streams over the net from live streams (e.g. webcam, ipcam, etc.) then broadcast that same stream on my local network using a different URL. I know gstreamer can do it quite well but I don't know how. I'm quite confused with the way the documentation is written. Can somebody help me?
If you would like to retransmit the video streams using RTSP as well, you can use GStreamer RTSP Server. There is a lot of examples on the Internet how to use it. The best source of the examples is the gst-rtsp-server's examples directory:
http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/tree/examples
As you want to retransmit existing RTSP streams, you'll need to use the rtspsrc element for reception of the remote streams.
I think you are looking for something like this: https://github.com/jayridge/rtsprelay. It configures one rtsp server to receive clients on two urls with a record link and a play link.
This example uses a dynamic form:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1454
rtsp://server/path?uri=encoded-URI
you HTML encode the destination in URI form and add a path where it should register this camera to. The first time you connect; it will take some time, after that; the sessions are re-used.
I am implementing a client/server application where video streaming occurs between two computers (in one direction). I would like to have the server publish an SDP file when it starts streaming. The client would then be able to download this SDP file and use it to get the stream. In order to implement this it seems I need to include a RTSP server in my server application.
I am planning to use either libVLC or GStreamer for the client. Both are able to get incoming video streams using the info from an SDP file.
Server-side I don't really know where to start. Can anyone recommend a good C++ library that would allow me to create a small RTSP server?
Use Live555 LGPL library or for fun, read the RFC and implement :-)
Libcurl's library offers a simple example that can be usefull for the server side..
Take a look at: https://curl.haxx.se/libcurl/c/rtsp.html
I am using TCP/IP socket example i.e. "echoclient", and I am facing problems with writing and reading.
I am connecting to server socket but it shows null data. I don't know whether it is data conversion problem or any other issue.
Download Wireshark and see if it helps debug. It can peek all network traffic.
In case if you are connecting via mobile, check with your GPRS/EDGE/3G service provider if they are not restricting arbitrary IPs ?
Also in case if you're connecting from Simulator on PC, check if you have Proxies over your network before internet ?
It is always a good idea to use well tested libraries and implementation. Socket programming is c++ requires good understanding of OS if you want to write potable code.
Try Ace Wrappers - http://www.cs.wustl.edu/~schmidt/ACE.html It provide well tested pattern based C++ implementation for distributed network programming.