I have:
one pc with videolan installed (windows 7) 192.168.1.2,
one pc with linux ubuntu 11.4 installed 192.168.1.12,
I've try to send a video via UDP (in videolan) from windows 7 to in linux-pc
throught port 8081
in linux I've run netcat listening on port 8081 and redirect to file, (netcat -lu 8081 >file.avi)
(my goal is to redirect to a serial in embeded linux snapgear,
by the way now I need to rediret to a file)
but don't works and I can figure out how config all, netcat don't write every byte sent on 8081?
thanks
update:
my problem is that I dunno address of source, take a look at the following scenario:
linux pc send stream to a serial, and other pc send to that pc a stream.
linux pc is alway in listening and don't know the address of source / sources,
this is my problem I dunno how to send with VLC, I know how send vlc to vlc (and it works)
but I dunno how send to 192.168.1.12:8081
thanks again
Your setup will probably not work: AVI is not designed to be streamed as-is over an unreliable transport layer protocol.
When you use VLC, it probably uses an encapsulation format like an MPEG transport stream or RTP, which you would have to properly decode before saving the contents to a file.
Even if you really manage to just stream the raw AVI file via UDP, you will run into problems because of packet loss and reordering - AVI is simply not designed to withstand that kind of errors.
You should either switch to a reliable transport protocol like HTTP (and even then use something else than AVI - it's really not designed for streaming), or use some other protocol to encapsulate your media, like an MPEG transport stream or RTP.
Please check your IP-Adresses. Do they really have the same?
Does the netcat work for other protocols? You may test it with the dns-port and nslookup on windows.
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 have to create a service which captures the audio from the PC microphone and to broadcast it as UDP packets. I am on a Debian platform and I have to use Python (3.7).
I would like to use PyAV because I have to link this broadcasting system to a local custom WebRTC service using aiortc, which relies on PyAV.
I have to do this because I cannot access the same audio source (ALSA) from several processes (RTC peers), so I was thinking to create a UDP broacasting system in a localhost environment. Is this the best practice? Have you any other idea?
I have noticed here that with the call: av.open("udp://xxx:nnn", format="alsa") I should be able to receive audio UDP packets, but I am not sure how to generate a UDP server which captures from the mic and sends the UDP packets, so, how to create the server side of this implementation? In particular, I managed to capture the audio with: av.open("hw:0", format="alsa"), how can I send the captured buffer over UDP sockets?
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 am trying to write an ONVIF video stream at the camera side on Linux Fedora (not the client side).
I already installed the WSDL2H and ran:
wsdl2h -c o onvif.h http://www.onvif.org/onvif/ver10/device/wsdl
This command built the onvif.h
I wrote a UDP, HTTP, TCP server on my Linux.
In my UDP server program I catch the Hello message from the ONVIF Device Test Tool.
How should I proceed from this stage/point?
I would like to know how to write the "answer" function in C?
At first you need soap tool like as gsoap.
Using gsoap you have to implement WS-Discovery
(Hello message is the one of the WS-Discovery)
In the gsoap FAQ, You can find useful information.
Also in the gsop documents, there is a information for WS-Discovery
I would like to create a demo with the following topology:
... --->[switch] <---> [Host]
DEMO: A switch sends a packet to a monitoring Host (The packets original destination is not this Host, yet switch will send to it via a mirror port). The monitor-Host is to capture that packet, do something with it (e.g., just dump the L2-L4 header fields into some log file), and then send the original packet back towards the switch.
Host Environment: Ubuntu 12.04 Linux.
Dilemma: What's the simplest way I can capture the packet in Host and send back the original packet to the switch?
Possibilities to Explore:
Create a packet sniffer program in C (looks complex ... libpcap, AF_Packet sockets etc.).
Try to use python scapy (don't know how complex this will be).
Try to install some open source proxy server of some kind to which I can write a plugin that will examine the captured packets.
Question: Any better recommendations (if I can avoid going the programming route, it would be preferred. Is there any simple scripting method to do this?). Looking for a quick-and-dirty method here. Thanks.
well, you can use packets sniffers available such as wireshark, ettercap which will capture all network packets (using promisc mode) and dump them in readable format.
Or, you can preety much easily code sniffer for yourself in python, linux which is not so much complex to understand..