Is there any way to cache rtmp streams?
Normally squid successfully caches http based videos. What about rtmp?
Do you know any other tool?
about rtmp: http://en.wikipedia.org/wiki/Real_Time_Messaging_Protocol
Thank you,
Related
I'm developing an application that receives an audio stream over a WebSocket and needs to forward the audio to a SIP server.
Currently, I've managed to connect to the audio source using a Websocket and receive the media stream (encoded u-law) using Node-Red, but I'm struggling to figure out how to send the media stream to the SIP server. Any advice would be much appreciated.
I looked into this for a similar question a while back, can't find where it was now.
As you probably know the media part of SIP is RTP, so its a fairly separate stack to the call signalling.
I didn't find any nodes that supported it and the few node.js libraries for RTP were all very incomplete and out of date.
In theory it might be possible to craft your own RTP streams using the UDP nodes and then create the relevant SDP in the SIP response but I'm not sure how robust or scalable this would be.
The other option is that there are a couple of Programmable Comms platforms out there that support both SIP and Web sockets so you could possible utilise one of those and connect from Node-RED via websocket letting them do the SIP work.
I've done SIP|<>Websocket stuff with both the Vonage API (Previously Nexmo) and Jambonz (open source)
I am trying to make a peer-to-peer game streaming platform. At this point I managed to capture the OpenGL frames and I have a functional Java websockets server, I can have 2 clients that establish a peer to peer connection (I have solved the STUN/TURN servers part) and transfer text at this point.
I do not quite understand how I could stream a video made out of the Opengl frames with a low latency (<100ms). The problem mainly lies in the FFMPEG part, I want to use this to encode the frames, get the result (stdin/stdout redirect for ffmpeg ?), somehow link to the the JS API of the host (maybe a local websocket to which the JS of the hoster will connect to).
I tried several FFMPEG arguements/commands with stdin and stdout pipes and they did not work.
What WebRTC Client are you using? What is the H264 Live stream flowing into?
WebRTC in the browser has a few restrictions (just because the implementation is naive). Try doing constrained-baseline, and do a very small keyframe interval (every second is usually good for a prototype!)
If you don't have a WebRTC client you can do something like webrtc-remote-screen
I am trying to stream some content from my browser's webcam implementation to a random RTMP server. I got it working to the part where it sends blobs of WEBM (VP8 i believe) encoded bits of movie to my server every 2 seconds, but the tricky part is getting it to an RTMP server from that part on.
A bit of fiddling with FFMPEG showed that it can successfully stream to the server I want to stream to, but so far I have only managed to get it working with regular files. Attempting to stream the blobs are unsuccessful, it simply does not upload anything. It also only seems to accept mp4 encoded with the h264 codec.
The question: what is the best way to get the raw video data from my webbrowser's webcam implementation, encode it with the h264 codec and send it to an RTMP server?
Without using a server to convert your blobs to an RTMP stream, The only way is you use flash. RTMP is an adobe protocol that no browsers support natively. Another option is WebRTC which uses RTP protocol.
i was installed red5 server on ubuntu 12.04 lts for live and vod video streaming. i just want to convert my RTMP protocol stream to RTSP and HTTP protocol stream. i was studied and search about FFMPEG. but i did not understand correctly. so please guide me any one. thanks advance
the sample URL
rtmp://xxxxx.com/live
to
rtsp://xxxxx.com/live and http://xxxxx.com/live
There are red5 plugins for hls(http live streaming) and rtsp. I don't know how stable they are so you can try. Here are the links.
https://github.com/Red5/red5-plugins/tree/master/rtspplugin
https://github.com/Red5/red5-hls-plugin
You could write a transcoding application that uses Xuggler and converts your streams on-the-fly to RTSP (RTMP -> RTSP). That isn't meant to sound simple, but it can be done as well as the other way around (RTSP -> RTMP).
I want to establish a video stream between a C# application and a Browser.
Im using Websockets for the communication.
The Video source is a webcam.
I am able to request single PNG frames but it is slow as hell.
The websocket Server(ratchet) is message based but is it possible to use VP9 compression or something similar by using some kind of buffer?
WebSockets implement a messaging protocol over sockets. This is not desirable for video. I think a better suited technology for this is WebRTC.