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.
Related
I have 5 requirements:
I want to send sounds that are output of other programs over voice chat programs(e.g. TeamSpeak, Skype etc.)
I only want to send the sounds of certain programs. Not all my system sounds
I must still be able to talk to them (mice input should still be used).
I still want to hear the sounds of what I send.
It must be a software solution.
My scenario:
I am playing LoL/DoTA/CoD/BF (whichever makes you happy), I am on Teamspeak with some friends. Something happens and I want to play a fitting sound (e.g from http://www.myinstants.com/). So I want to send the sound from my browser over the chat.
What I tried:
I installed CheVolume (http://www.chevolume.com/Infos.aspx). This is for handling output devices, not sound input.
I set Stereo Mix as my default communication device. This works mostly, but then I also send my game sounds over chat.
I have installed VB-AUDIO (http://vb-audio.pagesperso-orange.fr/Voicemeeter/). It can be useful, but it is not what I want. I get similar results as using Stereo Mix.
I installed Jack (http://jackaudio.org/) shame to say it is to technical for me.
I tryed using Virtual Audio Cable (http://software.muzychenko.net/eng/vac.htm). Again, this only enables me to send all my system sounds.
but Voicemeeter allows to do that:
Exactly : See User Manual Case study #1
it is possible only if the application allow settings its playback device, then you will be able to route an application Voicemeeter virtual input or physical input through a VB-CABLE (Voicemeeter Banana version is better for that since it provide more I/O)
3,4,5: of course.
I'm trying to create an interactive voice-tree for an art project. Think of something like a choose-you-own-adventure, but on the phone and with voice commands. I already have a fair amount of experience working with Construct 2 (game-making software), and can easily build a branching, voice controlled interaction loadable through a modern browser with it. For reasons relevant to the overall story, I need players to connect to the interaction through a Google Voice number they will call.
I already have a GV number and have written an AutoHotKey script to auto-answer the Hangouts call, but I'm stuck trying to route the audio from the caller in Hangouts to the browser AND the audio response output of the browser back to the caller.
I know of an extremely primitive way to accomplish this, [which I've illustrated with this diagram:
Unfortunately, this is rather cumbersome and I suspect I can achieve my goal through virtualization or at the VERY least some sort of attenuation cables between two physical machines (I tried running a generic AUX cable between two laptops, but couldn't get speaker audio to go into microphone audio from one to the other).
I've been experimenting on Parallels running Windows 8.1 with Virtual Audio Cable(no luck), JACK(too robust), Chevolume(too limited), and IndieVolume(too limited).
I suspect VAC would be the best bet, but I can't seem to find a way to route Firefox audio output to a microphone input which directs to Chrome and vice versa. If I try accomplishing it all through just one virtual machine I have to use two different browsers for the voice-tree webpage and Hangouts call since Hangouts pushes its audio through Chrome (even the stand-alone application).
Is there any way to route microphone input and speaker output separately between two virtual machines? If not, could I still try and accomplish this with a specific type of cables between two laptops running windows 7/8 that have generic audio jacks?
I need a SIP server, hopefully simpler, more efficient and more secure than Asterisk or other full-featured PBX, to just terminate SIP calls and play audio, based on the incoming number. This would be in a linux environment.
I'm concerned about long-term support, and much-reduced attack surface. I'm a programmer but a no0b in the SIP world. I want to avoid the tons of security hassles of a full-featured PBX, let alone the crazy big bills after a hacker gets into it. Thanks in advance!
Personally I would build one with pjsip. Assuming you do not need built in registration server all calls would be handled by so called local account. You can use command-line pjsua with auto-answer and auto-play options for initial tests and then add some logic for audio wave selection and perhaps audio player allocation/deallocation (audio players could be active all the time and shared by multiple calls for efficiency if they e.g. contain something like background music).
For security disable not needed parts (TCP, TLS transports, not needed codecs?) and use port other than 5060. In my application I'm also using kind of application-level firewall that blocks requests from hosts that generate suspicious traffic (registrations, malformed SIP requests).
I was wondering whether it is possible to capture audio data from other sources like the system out, FM radio, bluetooth headset, etc. I'm particularly interested in capturing audio from the FM radio and already investigated all possibilities including trying to sniff the raw bluetooth communication between the phone and the radio device with no luck. It's too bad Android only allows recording audio from the MIC.
I've looked at the Android source code and couldn't find a backdoor to allow me to do that without rooting the device. Do you, at least, have any idea how to use other devices (maybe access somehow /dev/audio) say via NDK or even better - Java (maybe Reflection?) to trick the system to capture the audio stream from say, the FM radio. (in my case I'm trying to develop the app for the HTC Desire)
PS. And for those of you who are against using undocumented APIs, please don't post here - I'm writing an app that will be for my personal use or even if I ever publish it I will warn the user of possible incompatibilities.
I've spent quite some time deciphering the audio stack, and I think you may try to hijack libaudio. You'll have trouble speaking directly to the hardware (/dev/*) because many devices use proprietary audio drivers. There's no rule in this regard.
However, the audio hardware abstraction layer (HAL) provided by /system/lib/libaudio.so should expose the API described at http://source.android.com/porting/audio.html
The Android system, and especially audioflinger, uses this libaudio HAL to find available devices, deal with routing, and of course to read/write PCM data.
So, you could hijack the interaction between audioflinger and libaudio, by renaming the later, and providing your own libaudio which decorates the real one. Doing so, you should be able to log what happens and very possibly intercept FM radio output, provided that this is not directly handled by the hardware.
Of course, all this requires rooting. Please comment if you manage to do this, that interests me.
I have an application where I have about 10,000 pieces of monitoring equipment across the US that periodically dials into a bank of 32 phone lines. I have two receivers of 16 lines each that answer the call and temporarily stores a small alpha string. I then have a computer that polls the receivers and parses the string and copies it to a database.
I am looking to replace the phone lines and the receivers with a voip solution and rewrite the software to parse the data string.
Any ideas on where to get started?
Tom's suggestion about Asterisk is a good one for the overall system.
However you will still need to decode the data sent from your remote equipment from an audio signal to a data signal. That task is what the "dem" part of Modem stands for (Modulate/Demodulate). Either you do this with a canned hardware/software package (as you are currently doing with a commercial modem) or you have to emulate the modem in software yourself which will be extremely tricky to code at the very least if you attempt it yourself (heaps of standards that you have to comply with for a general modem solution, plus the solution needs to work in real time)
For the software approach could start with this page Linmodems.org (just a something I saw on google prompted by your question). Alternatively do lots of searches on google for software modems. Getting someone else's code is the best approach for this sort of code :)
Whatever you end up doing I suspect it will be rather custom.
A good place to start is probably Asterisk PBX.
I take it you don't want to replace the modems at the client sites (the easiest thing on the server side would be each clients had its own IP software stack, and used its modem to call an ISP and establish an internet connection, and then talk to your server using TCP or UDP or HTTP or whatever).
Assuming that you don't have IP capability on the client sites, Googling suggests that the relevent technology is called "Modem over IP" or "MoIP" (which Wikipedia seems to be confusing with "Mobile over IP").
VoIP consists of SIP for signalling (e.g. for call set-up and call tear-down) plus some codecs (e.g. H.323) for traffic (encoded voice) while the call is established.
I'm guessing that MoIP can keep the SIP signalling, but needs to use some different codecs.
V.150 Modem over IP White Paper looks like an introduction to the technologies. I don't know what vendors there are.
I presume you are looking to find a way to do this without mofidying the modem hardware at your remote sites. If this is the case you will have to find or write signal processing software to demodulate the encoded signal from the modem. Fortunately, signal encodings on a modem are designed to be easy to do this with.
Maybe somebody makes software modem libaries that do this sort of thing. The other parts of the problem will be emulating the handshaking on the modem so it plays nicely with the remote sites.
If you can modify the software (really just the number to dial, but it would have to include the data you want to transfer) at the 10000 sites (not likely!), you could in theory use DTMF in the "dial" string to key the data over into Asterisk. Ok, more than a bit hackey, but it would avoid having to have a software modem. Note: you'd want a checksum!! (and maybe send it multiple times) And a way to tell the caller if it was received correctly. Like I said, hackey but cute.