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.
Related
I have been using UDP sockets to send and receive voice through RTP packetization. It is pretty straightforward. I just send my mic voice signals ( that are encoded ) over IP using User Datagram socket , and on the other end i receive the UDP-RTP packets and decode them to be able to play them on my speakers.
I have been searching on internet for a while to find a way to start up a session using UDP sockets. What i want to to is to a Handshake-like process between two ends of my conversation and after the requests were acknowledged the media layer ( which i described in first paragraph ) would fire and start sending voice.
I have not been able to find any tutorials on session request using UDP sockets but i suppose it shouldnt be impossible.( one user sends a request to build a session and if the other user confirms media layer starts)
Has anyone done something like this before? any info is welcome.
Firstly, UDP is a connectionless, unreliable protocol, you won't find anything like handshaking for negotiating connection i.e no session management. But, to transport RTP packets it's not a good idea to use tcp, it lacks realtime feature, so you have to stick with UDP. Now, to overcome the signaling problem you can use protocols like. SIP. It's standard signaling protocol used in VOIP. SIP initiates a connection before sending RTP packets. To properly use SIP and RTP you might have to take help of another protocol called SDP, which tells which port to use for transmitting RTP and other various info. You can get more info about these techniques here. Hope this will helps!
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..
Can anyone please tell me if it is possible to publish the voice from an IP based telephone to a RTP based media server - like a Wowza Media Server or Flash Media Server?
Thanks
You probably need to give more info if you want a more specific answer, but a quick response is that yes it is possible.
If the IP phones is something you are building or can modify then you can simply send any outgoing RTP packets both to the other end of the call and the media server, and forward any received RTP packets to the media server.
If you are not able to modify the phone, then you may still be able to achieve what you are looking for by 'mirroring' the RTP packets that are sent to and received from the IP phone, to the RTP media server (see http://en.wikipedia.org/wiki/Port_mirroring for an overview of port mirroring, and www.audiocodes.com/filehandler.ashx?fileid=43289 for some specific discussion on 'tapping' phone calls).
Note, that you need to be aware of the law if this is to record or 'tap' live calls. Even if they are your own calls on your own phone, different countries have different laws about what can and cannot be recorded and what notification you have to give the parties involved in the call (this is why calls to call centers etc often start with a message that the call may be recorded for training or other purposes).
I need to write a packet sniffer in Linux that detects HTTPS packet that are sent and save the url from the request. I found code for this in security-freak and ran it. This code runs and only sniffs the received packet but I need to get the sent packet in the sniffer.
How do I get the sent packet in this code?
I can't use any library like libcap (forbidden).
The code is :sniffer.c
You should be using ETH_P_ALL instead of ETH_P_IP as the protocol. ETH_P_IP only listens for incoming IP packets.
Why can't you use any library? Homework?
It's hard to answer without having examples from your code, for example how you set sll_pkttype.
The urlsnarf tool in the dnsiff suite could be worth a look.
With appropriate libpcap or DNET usage You should be able to get all network
traffic on the desired layer (protocol - 5) (also this outgoing).
But You should know that already.
You need to go through the above libraries manuals and find the appropriate filtering.