Any linux library or docker container for:
receiving simultaneous voip SIP calls, and for each call,
recording the incoming audio stream to a distinct file,
having silence as outgoing audio stream.
Essentially, a simple sip call receiver and recorder, accepting simultaneous calls, and playing silence.
Drachtio did not work out-of-the-box. Is there an easier alternative? Maybe, a very simple softphone would be fine to start with? A Gstreamer pipeline?
If you're familiar with c# and can install dotnet on your Linux machine then my library can do that. Here's a single call example of recording an incoming call. It will need to be enhanced if you want to co-ordinate multiple calls.
Related
I want to make a server play a soundbyte every time it receives a request. Is there a way to do this if I'm using a Go based server? The idea would be the server is hosting a browser window, it receives a request and the browser goes 'ping!'.
It depends on which operating system you want the code to work. Afaik there is no generic cross-platform solution for playing sound from go:
On Linux you might need to rely on Pulse Audio with a package such as github.com/mesilliac/pulse-simple
On Windows and Mac you could use PortAudio with a package such as github.com/gordonklaus/portaudio
If you want a practical example there is a go-based multi-source music player project called "moggio" at github.com/mjibson/moggio that plays audio from multiple sources on Linux, Mac and Windows.
You can have a look at the github.com/mjibson/moggio/output package. There you will find the code that moggio used to play music on Linux, Windows and Mac.
While working on voip apps, I usually end up picking up one phone, talking to it, picking up the other phone and check if I hear myself. This even gets trickier if I'm doing apps with three way calling.
Using a softphone doesn't help.
Ideally, I want to be able to run multiple instances of some command line based sip ua wherein i can dial a number. Once the ua has dialed and the other party ha picked up, both agents exchange audio. But instead of having to hear some audio, the apps instead display some text which identifies the other end. Possibly some frequency pattern that can be converted to text. Then this text is displayed on the app.
Can something like this be done? I'm creating apps against freeswitch. Ideas how to debug voip apps are also welcome in the comments.
yes, absolutely. The easiest would be to have a separate FreeSWITCH server that is used for placing the test calls and sending/receiving your test signals.
tone_stream will generate the tones at frequencies that you need: https://freeswitch.org/confluence/display/FREESWITCH/Tone_stream
tone_detect can detect the frequencies and execute actions, or even better, generate events that you can catch over an ESL socket: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+tone_detect
The best way to generate such calls is to use a dialer script which communicates to FreeSWITCH via Event Socket. Here you can see some (working) examples that I made with Perl:
https://github.com/voxserv/rring/blob/master/lib/Rring/Caller/FreeSWITCH.pm -- this is a part of a test suite tat I build for testing a provider's SIP infrastructure. As you can see, it connects to FreeSWITCH, starts event listener, and then originates the call and also expects an inbound call. It then sends and analyzes DTMF.
https://github.com/voxserv/freeswitch-helper-scripts/tree/master/esl -- these are special-purpose dialers, you can also use them as examples.
https://github.com/voxserv/freeswitch-perf-dialer -- this one generates a series of calls, like SIPp does.
Another technique is to play a sample audio file and record the audio being received on the other end[call recording] and then comparing the two. This system works on setup where systems are located at various places and you are testing end to end quality.
There are lot of Audio Comparison tools [like PESQ] should help you not just detect the presence of Audio but also give stats about the degradation of various parameters in the audio stream.
This can be extended to do test analysis of FS patches as and when they are released and also for other hooks or quality standards you want to enforce.
I am looking to build an app that needs to process incoming audio on a phone call in real time.
WebRTC allows for this but i think this works only in their browser based P2P audio communications functionality but not for phone calls/ VOIP.
Twilio and Plivo allow you record the audio for batch/later processing.
Is there a library that will give me access to the audio streams in real time? If not, what would I need to build such a service from scratch?
Thanks
If you are open to using a media server (so that the call is not longe P2P but it's mediated by the media server using a B2B model), then perhaps the Kurento Media Server may solve your problem. Kurento Media Server makes possible to create processing capabilities which are applyied in real time onto the media streams. There are many examples in the documentation of computer vision and augmented reality algorithms applied in real time over the video streams. I've never seen an only-audio processing module, but it should be simple to implement just by creating an additional module, which is not too complex if you have some knowledge about C/C++ and media processing concepts.
Disclaimer: I'm part of the Kurento development team.
I would like to create a web application that sends and receives ALSA MIDI messages on Linux. Only one web client is intended.
What kind of architecture / programs do I need for that?
I am familiar with django but can't find the missing link to ALSA (or any system with a gateway to ALSA on my Ubuntu machine). Also, I have the small program ttymidi (http://www.varal.org/ttymidi/), that sends messages from a serial port to ALSA.
You might be able to use Python's miscellaneous operating system interfaces, but web applications aren't often designed in this way. You may also have to worry about latency and buffering in your program.
The simplest way of doing what you want without any third-party library is to use pyALSA, which is the official python wrapper around the C ALSA library.
I recommend you dealing with the Sequencer API instead of the RawMIDI stuff, which is lower-level. Check out some of the test apps and the C API documentation, it will definetely help you to write your code.
I want to make (for fun, challenge) a videoconference application, I have some ideas about this:
1) taking the audio/video streams (I don't know what an audio/video stream is)
2) pass this to a server that lets communicate the clients. I can figure out how to write a server(there are a lot of books and documentation about this) but I really don't know how to interact with the webcam and with the audio/video in general.
I want some links, book, suggestions about the basics of digital audio/video expecially on programming. Please help me!!!
I want to make it run on a Linux platform.
Linux makes video grabbing really nice. As long as you have a driver that outputs the video stream to the /dev/video/v* channels. All you have to do is open up a control connection to the device [an exercise for the OP] and then read in the channel like a file [given the parameters set by the control connection. Audio should be the same way, but don't quote me on it.
BTW: Video streaming from a server is a very complex issue. You have to develop or use an existing protocol. You have to be very aware of networking delays, and adjust the information sent (resize or recompress) to the client based on the link size between the client and the server.