I have peculiar request where I need to transmit audio between Cisco IP phones when they are on a call. The reason I am doing this is so that I can get call quality reports for determining audio quality of a call. However, I am absolutely stuck right now. Is it possible to do something like this through Jtapi? Any advice would be greatly appreciated.
As of now, I have tried Cisco Media Terminal (CTI port) and used commands like RTPRx and RTPTx to receive and transmit rtp streams with no success.
You could just launch Wireshark to capture the call and then use the Wireshark VoIP/RTP tool to analyze the quality. (I don't know it's exact name, it can be found in the menu)
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I need some help because I don´t know how to approach this challenge.
I want to build a device, that's receiving a Bluetooth audio signal and is forwarding it to a Bluetooth speaker. It´s also running some algorithms with the audio data and also simultaneously sending results via UDP to a different device.
I already thought about using two or three ESP32s, using one with an extra Bluetooth module, or searching for a whole different MCU with Bluetooth 5.0 or higher and Wifi 5GHz. But I don´t know approach the best is, or maybe a completely different one.
Some context, why we want to do it:
We want to create a real-time light show, based on the current playing song. It is already working for PC, but also want to make it accessible for phone users. Sadly there is no way to capture the internal audio on iPhone or Android phones. Our Idea to make the music sync with the phone possible is that you are connected with the phone via Bluetooth to our "sync box" which is then connected to the speaker via Bluetooth or AUX. The "sync box" is running our algorithms for creating the light shows and then sending the data to the microcontrollers from the light strips.
So maybe you have an idea how we can sync the lights to the music completely differently or how I can approach the challenge with Bluetooth.
Any help is highly appreciated.
Thanks a lot.
I am trying to build an open-source in-ear monitoring system. I have created the UI and was wondering how I would get the channels that are on an audio mixing console so that I can edit the channels and stream them to each musician. Is there a certain protocol that all the mixers use? You can find the project at https://gitlab.com/openstagemix. We would love to have contributors.
I can't really test whether this is the correct answer as I am trapped in my house during the coronavirus time. But, all mixers use something called OSC which is a protocol between mixers, synthesizers, etc. to computers. You can find more information here http://opensoundcontrol.org/introduction-osc.
Update:
It's neither! I am going to use the AES67 standard to receive information from my mixer and with that process the audio. This is because my mixer is ethernet capable.
So I'm kinda stuck here.
I have a radio station, but we are mobile. So I have a studio on wheels. The problem is, we have an antenna, but we always have to place that really close to our studio. Now I want to make an device that can stream the audio from the audio mixer to the internet and can be received by another device in another network and send that signal to the antenna (audio output).
to make this clear, I made a schema with raspberry pi's;
I want this to be plug and play So I only have to plug in the device in the modem (or network we have) on both sides and the devices should find each other.
I don't know HOW I can do this, so I need to know a couple of things:
What hardware should I use?
What software should I use?
What is the best configuration to accomplish this?
Can I use 2 raspberry pi's?
How can I let the devices find each other over the internet?
There need to be some features;
The system needs to be able to buffer the audio for 5-10 seconds
It needs to be direct, so it's live and not a file that needs to be played
The system must be failless (beside the fact the internet can die).
Plug and play is a must, I don't want to have a really messy configuration to do. (if possible, without any kind of portforwarding).
I would really appreciate help and a decent explaination.
regards,
Robin
Well, it depends on your capabilities as a programmer.
If you're really fixated on the RPi for it's convenient form factor, there's a ton of community support, so I'd start with something like this project to kick start you in the right direction. If you already know python pretty well, modify away and have fun.
If you have no programming experience, you'll probably want to put a desktop in place of the RPi and launch some instances of VLC. It's not necessarily plug n play, but you can get close enough by getting a command line VLC to launch at startup.
Either way, the more difficult problem here is the "over the internet" part. This would really need to be a server-client model, but who is your server depends on who is more stationary (I'm guessing Location 2?) because the client will need to know the IP address of the server somehow. There are dozens of ways to make this happen, but at the end of the day, you'll want to use sockets accomplish the
It needs to be direct, so it's live and not a file
... which unfortunately gets complicated. See this answer for confirmation. Would love to help with some tips on implementation, but we need more information about your willingness to "dig into the code", the necessity of the RPi, and whether the stationary location has a static web address.
What I would like to do involves a small bit of hardware. 1) a phone headset, 2) a PCI-modem, and 3) a phone wire. What I would like to do is read audio from the modem, and then digitize it for processing. I'm sure the best way to do this is with Linux, but if it can be done in Windows as well that would be awesome. A second extension of this, is that I would like to be able to translate digital audio to analog audio and send that to the modem so it can be heard from the headset.
Any advice would be greatly appreciated. ( Also, if anybody has a general "pointer" to what I should investigate to replicate the audio stream to a TCP server so it can be accessed over LAN, that would be even cooler. I know how to handle TCP well enough, but I haven't a clue about audio encoding / decoding ).
If anybody's curious, I'm wanting to create a home-wide audio-stream with ears and mouths. Since the phone cables can do that with normal headsets, I thought "why not".
Not just any modem will do. You need a "voice modem", which includes audio capability as well as general modem functionality. These devices usually expose themselves as a regular sound card on the system, once the drivers are installed. From there, you can use any mechanism you want to read/write from those audio streams.
Be warned though that your plan of a whole-house speakerphone isn't simple at all. There are significant feedback issues when using regular POTS lines. There are entire companies that work to solve this problem. The best of them use microphone arrays that are steerable in software. You would be better off using one of these off-the-shelf systems.
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.