I would recommend Mumble
Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming.
Some notable features include
Low latency, high quality, encrypted communication
Manageable group and user based permissions
Optional In-Game Overlay
Mumble is free, open source, ...
I think Linphone is what you may be after.
It's free and open source, under the GPL license. It's rather lightweight, allows for multiple connections, works on Linux, Windows, Android and iOS. It uses an SIP protocol, so you'll need an account of sorts. You can go to linphone.org to create one, or use any other SIP account.
As we have looked into the librariers quite extensively I would like to share our findings with you.
Now basically i've come to accept that the Interface Quality of basically all the VoIP libraries is.. well.. bad! At least from a Clean-Code developers view-point.
Exceptions like "License is invalid or expired or any other error occured" are standard.
My project, Managed Media Aggregation, does everything you need with rtp and rtsp, it includes a client and server which are both efficient and standards compliant.
The server supports over 1000 clients without any issue and memory usage is rarely above 150 mb even with those clients consuming media.
The solution also has some codec implementations but ...
I also evaluate Ozaki for some weeks. It seems very promising, but as a new comer, I find that is not well documented (at least for me that I am not experienced in VoIP domain, I wish more verbosity, for example in what concern errors), somehow inconsistent between consecutive releases, and their site seems to me very cluttered with much information, but not ...
Any SIP-enabled software (aka softphone or SIP client) will allow you to receive calls on a local or toll-free number. That's not a problem - there are lots of various free or commercial SIP clients out there (for example, X-Lite and , correspondingly). You can either download some SIP clients from providers' websites for free, or download from various ...
While Linphone might be a good software, I have experienced some problems with it (don't remember exactly the nature - it was a couple of years ago, but, hopefully, it's more stable now). Anyway, my recommendation would be another open source, multi-platform VoIP software suite, called Jitsi.
It is powerful and modern VoIP software, satisfies all your ...
create announcement file (wave S16LE, 8ksps); it must be as long as the longest call voicemail should record as call would be disconnected when file would end, thus it should consist of few seconds of prompt, period of silence and perhaps warning at the end that call is about to be terminated
configure softphone to register to your provider, enable ...
I cannot tell how well it would meet all your requirements – but Linphone might be a good match. It supports the SIP protocol, and thus can be used with many providers and even self-hosted solutions. I basically use it when abroad for calls home using my VoIP provider. It worked well even in hotel rooms with barely 2 bars WiFi – which might come close to ...
Almost any softphone has auto-answer option, but if you would like to compile something yourself I'd point to
libre + librem + baresip. It can be built for any platform you mentioned (out of the box it's mainly console application but GUI can be added), it's simpler than pjsip and it has BSD license.
If it is supposed to be maintenance free make sure that ...
well there are two ways that comes in my mind:
the first is sicking with SIP (which is fine)... so i would try Jitsi:
it is free (as in beer and speech)
capable of conference calls
isn't available via the default repositories but they provide pre-compiled packages and maintain their own repository
i have to add, that i don't like the GUI of Jitsi but the ...
I found out that Skype offers phone numbers. You can choose between a three-month plan and a one-year plan. Then if you keep Skype running on your computer, someone (to whom you have given your phone number) can dial your number and a pop-up will notify you when you have an incoming phone call.
I have received two phone calls so far, and it went well.
See either in softphone logs (but I don't know if cisco is making them) or using Wireshark where does the call is coming from. It might be direct IP call or call from PABX. If it comes from PABX then refer to PABX logs.
You can use Asterisk PBX, that will control outgoing calls.
It has API so you can tell asterisk to dial numbers and the agents (workers) will not need to see them.
What I have done in the past is create a queue, and a script that for each contact will place a call in the queue (with command originate from asterisk) and will display whatever info you want on ...
After testing numerous apps I ended up using an app called JusTalk. It doesn't tick the open source box, but it does work as intended. The ring tone comes out of the phone speaker and the conversation happens on whatever is connected to the headset connector.
Take a look at some affordable web-based software like CrazyCall. It's a simple online calling application that offers local phone numbers from different countries worldwide, including those, from North America and Canada. The first number is free, but you'll need to pay $10/month per seat ($20 in total per month in your case).
Since it's web-based, your ...
You can try CrazyCall for this purpose as well. There are quite a lot of phone numbers from different countries to choose from and incoming calls are completely for free. It works both as a web-based app and a browser extension for Chrome, so there is no need to log into the application each time you're waiting for a call. All the fun for $10/mo, which is ...
There are currently (I can't speak for the future) three related services that when used in combination can help you, although most people probably don't know that they can be used without a mobile phone:
The Gmail Phone (this isn't hardware; it's a web app you can use from your Gmail account)
Google Voice (this is what actually gives you a phone number ...
Signal from open whisper systems would be a good bet as it is open source and is recommended by the EFF and many security experts. It meets most of your needs as it is end to end encrypted, supports voice calls and text messaging.
One thing I am not sure is practical is how would you use a different phone number than the one you have? Signal needs a phone ...
if you want to stick with the well known standards sip, srtp and zrtp have a look at Ostel or to be more concrete OSTN from the Guardian Project.
for a "closed environment/user-base" Mumble (with its server Murmur) can be interesting.
well... or you can take a look at "new" projects in early stages like Tox (NaCl-encryption) or Ring (because of dht without ...
You can use Matlab or its open source alternative Octave.
An example of how this can be accomplished in Matlab:
Here is the code that I used:
load handel.mat % An example dataset included in Matlab
plot(y) %Plot the graph without noise
y = y + randn(size(y))*.1; % Add additive white ...
You can set up one of the UK SIP-providers accounts which provide free DIDs. For example, Sticky Numbers provide toll-free UK-numbers (+4470) for free. However, they are expensive for the caller, not for you :)
After sign up, in the account dashboard you can set up forwarding to their (or your custom) SIP trunk, and then just add this number to your Android ...
Not tested myself, but SIP Voip Checker might be what you're looking for:
Free of charge (gratis): Check.
Works on Android KitKat: 3.0+, so Check.
No unnecessary permissions: Only INTERNET and ACCESS_NETWORK_STATE – Check.
No advertisements: I've not tested it, but I strongly think so (arguing by the permissions, at least no intrusive ad networks).
Consider DIDWW. They are nice enough for your requirements: they offer DIDs in US jurisdiction in almost all states.
The average cost of a DID number is 1-1.5$ monthly, which perfectly fits your upper limit:
They allow creating a SIP-trunk to the number, so you can add DIDWW account to any SIP soft- or hard-phone.
You have multiple, only tangentially related questions that would be better served being posted separately. Answering one of them...
Any tutorials/docs on how to get Jitsi work in walkie-talkie mode are highly appreciated.
I just attempted to use this featured and determined that is works in the same way as the Skype PTT feature, which is to say it is ...
I am using Ozeki SDK,
According my experience the latest versions(1.3.7 and 1.3.8) is not working as expected in some cases.
Also almost there is no support at all.
I am testing https://sites.google.com/site/sipekvoip/ rigth now, It is promising.