WebRTC Browser Phone with Asterisk & Raspberry Pi 

I't my absolute pleasure! It's been fantastic fun putting these videos together, and there is a bunch more still to come. COVID has blown a hole in my plans this year, and I have not had a chance to produce as much as I have. I'll be making more soon... stay tuned!

My pleasure! Happy to share. Stay tuned, more videos on their way.

Glad it was helpful!

My pleasure!

Glad you liked it!

Glad you like it!

You are welcome!

Thanks for the feedback. Automatic or scheduled calling is a good topic idea. What is the NAT issue you are facing? If you want to ask a more specific question, it best to do so on the Asterisk Community Forum.

Hi Edward, Thanks for you feedback :) Yes, the GPIO is a great way to extend the functionality of Asterisk and to build awesome projects! I would suggest using python, and loading your script as a service, then program in a button press, that uses Asterisk AMI. The AMI can initiate call setups in the same way as if you are at the Asterisk CLI. I have done a video that (although doesn't cover this topic exactly) covers the principals: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-U9jEcvCyu0o.html

Yes, absolutely! FreePBX, is a great config editor for Asterisk, you really do have full control over everything. The webrtc settings that FreePBX generates are compatible with the Browser Phone. With the credentials - these values are entered into the Phone UI, and stored locally (In the browsers local storage). Generally this is ok, but you may need to know this if you are using this software commercially.

You're welcome

Correct. The project has evolved a little since this video, but the mechanics are about the same - like WhatsApp, you have to create contact first, but it has a number of features that speed this process up. You can now click the quick dial button, enter the number and hit dial, it will then create a contact for you, and setup the call. You see the reason for the contact is that after the call, you can go back and get a full CDR details and quality analysis.

This and more coming soon.

From what i recall the default supported operating system is a version of Debian, and Debian and Ubuntu are very similar. I would generally advise to use the default supported OS - they operate much faster and more stable like that.

bump

You can make your own scripts around this code (to perform the auto dial)

"Can you help to develop feature to dial mobile numbers also without using buddy feature ?" If you pull the latest code from Github, you will see there is now a dial option without first making a buddy. github.com/InnovateAsterisk/Browser-Phone It automatically creates a buddy for each incoming and outgoing call, but it's required fo the record keeping, and can easily be deleted. "Can we use this in intranet (without internet in local network)?" Yes, I believe this does work, however the STUN lookup will timeout (about half a second each time you setup a call). STUN lookups are mandatory on WebRTC - there isn't much you can do about this.

It may be tricky to get this working on iphone. There are a number challenges in the UI, permissions, and also relating to the default iphone behaviour, like network settings etc. For the moment I would not recommend using this on mobile devices.

Thanks for the feedback! :) There was quite a big difference between Asterisk 13 and 16, especially around chan_sip and chan_pjsip etc. In the end it's was better to use chan_sip with Asterisk 13 and chan_pjsip with Asterisk 16. So I think it will probably be fairly smooth sailing for Asterisk 17. I have not had a chance to test out 17 yet... please try and let me know.

It is GNU Lesser General Public License v3.0 Basically this means it can be used in a commercial project, yes. More here: github.com/InnovateAsterisk/Browser-Phone/blob/master/LICENSE "Permissions of this copyleft license are conditioned on making available complete source code of licensed works and modifications under the same license or the GNU GPLv3. Copyright and license notices must be preserved. Contributors provide an express grant of patent rights. However, a larger work using the licensed work through interfaces provided by the licensed work may be distributed under different terms and without source code for the larger work."

There was a missing dependancy in the installation, most probably related to OpenSSL.

The best way to approach this is to use the Web Hooks provided at most the the critical stages of a call. Take a look at the provided index.html page (github.com/InnovateAsterisk/Browser-Phone/blob/master/Phone/index.html), you will see some empty hooks that you could use to inspect the session as a comes in, and based on that, perform an answer action. I simple timeout action would work quite nicely.

When you clone the project, you also get a file called responsive.html Take a look at that file. www.innovateasterisk.com/phone/responsive.html

@@InnovateAsterisk Thanks!!!! ;-)

The reason the videos are based on Raspberry Pi is that it creates a predictable environment that from flashing the SD Card, to the end, will be exactly the same if you do it on your side. But... if you can muddle through the differences, then by all means go ahead. You will not able able to use Raspberry Pi OS on the VPS, but the closest OS will be Ubuntu. Also, the virtual CPU will be X_86 based, so there are a few things that you may not need to do, or would do differently. For example Asterisk directly supports and can automatically install the opus codec for you. Another issue to consider is that most VPS services like Amazon etc, will use a NATed inbound route. This may add complications to your SIP configuration. (Interestingly the WebSocket connection doesn't get affected by this as its TCP) Also, take a look at the Github project page for further information: github.com/InnovateAsterisk/Browser-Phone

Ubuntu and Raspbian (or Raspberry Pi OS as it's now called) are very similar, and stem from the same base Linux - Debian. You could even apt-get install asterisk, that would get Asterisk installed and then you can focus more on the WebRTC side rather than getting stuck on the OS.

You can just play around with it here: www.innovateasterisk.com/phone/ Or download it from Github - there is no restrictions on Github.

The default behaviour is to store the records on the device, in the internal storage of the Browser. You would have to save the records to your own server using your own methods.

Thanks for you feedback. I have a few videos planned for the future that may help with this.

@@InnovateAsterisk I hope ❤️

Try here: github.com/InnovateAsterisk/Browser-Phone/issues

The Github issues page is active, its best to get answers there: github.com/InnovateAsterisk/Browser-Phone/issues. In order for me or anyone to help you, you will need to show the error logs. This is the only way we can see what's causing this issue.

You can only call each other with video if the system picks up the Buddy as an extension it just uses a simple length check, i think the default is 6. So if the number you dial is less than 6, you should be able to use video - and it must be a buddy (not an address book contact)

Please make sure you follow the steps on generating the self-signed certificate. There is also more support and documentation here: github.com/InnovateAsterisk/Browser-Phone and github.com/InnovateAsterisk/Browser-Phone/issues

There is language detection based on your computer's (and therefor browsers) language settings. However there is also an override: var Language = getDbItem("Language", "de"); You need to have or set a localstorage key of "Language" set to "de" github.com/InnovateAsterisk/Browser-Phone/blob/2d964d355b2c195ae881a8e7bbefe222e085256f/Phone/phone.js#L139

@@InnovateAsteriskGot it! Thank you for the brilliant support. Probably I might come back in order to compile german sound files. Have a great day!

Ok it´s me again with a short question, Conrad. I downloaded german sound files in .sln16 format. Could you tell me how to convert them to .wav via command-line in terminal?advance. Hope there will be more of your great videos.

You can probably just download them. I saw this: www.asterisksounds.org/de/installieren I see it also describes the sox conversion. (Btw, the singed signed linear 16 bit is fine as a format for most situations, you may not need to convert it.) And then also language=de in the general section of the dial plan so that the various apps know to use the /de/ directory.

@@InnovateAsteriskThanks again...I didn´t know they do not have to be converted as in "make menuselect" w chose only .wav-files. Thumbs Up! :-)

Oh, that's strange... this didn't come up when I demonstrated the install in the video. The issue with Linux is normally the development packages and this is supposed to be taken care of during the command: > contrib/scripts/install_prereq install If for some reason this wasn't installed you can try run: > sudo apt install uuid-dev

@@InnovateAsterisk ​ I installed it on a Rasbian OS image on the vmare server, maybe that's why it didn't work? Unfortunately, physically I do not have raspberry. Thanks for the answer, maybe I will try on a regular debian 9, how would it look with these certificates? I just tried it before and something didn't work, Astersisk http sevrver only worked on port 80, and on http itself sound and vision probably won't work? Because now I don't know if I made a mistake somewhere yet.

@@InnovateAsterisk ​ @Innovate Asterisk I installed it on a Rasbian OS image on the vmare server, maybe that's why it didn't work? Unfortunately, physically I do not have raspberry. Thanks for the answer, maybe I will try on a regular debian 9, how would it look with these certificates? I just tried it before and something didn't work, Astersisk http sevrver only worked on port 80, and on http itself sound and vision probably won't work? Because now I don't know if I made a mistake somewhere yet.

You may not be able to set a custom port for 443 unless you run your service as root. Also take a look to the new video WebRTC in the Cloud: www.innovateasterisk.com/s2e2-webrtc-in-the-cloud/

Yes, if your web browser can play back or use the Bluetooth device, the browser phone will list it as a usable device. I tested this with a few Bluetooth devices, and even a headset - works fine.

Take a look at the developer console, something would have failed to load. Please post technical questions to the Github issues page: github.com/InnovateAsterisk/Browser-Phone/issues

Yes, no problem. Not all the commands will be exactly the same, but check out the other videos for WebRTC in the cloud

Excellent! thank you. Please go here: github.com/InnovateAsterisk/Browser-Phone/issues/15 and attach or copy-paste your file. If you already have a working copy running, and can't wait for the next release, just add the file to the lang folder, and add the code to the availableLang array: github.com/InnovateAsterisk/Browser-Phone/blob/master/Phone/phone.js#L86 This should pick up es from your PC settings, and use the language file. If your PC is not set to es in its own settings, or you want to force it, check here: github.com/InnovateAsterisk/Browser-Phone/blob/master/Phone/phone.js#L139 For the styling changes you simply need to know the ID of the text field. Its called "dialText" so, just make a small CSS file, and set the style as you want, but you probably need to use !important in the class settings.

Hi: I have attached the file es.json where you indicate. It is like zip. And is Spanish from Spain. I have also activated the language in my system!!!. Perfect!!! ;-).

...and the field of the telephone, perfect ... thank you !!!

Please feel free to add an issue to the GitHub site: github.com/InnovateAsterisk/Browser-Phone/issues You will probably need describe the problem in a little more detail.

Please post technical questions to the Github issues page: github.com/InnovateAsterisk/Browser-Phone/issues

It's quite important to remember this is not WhatsApp, even tho the UI makes it look a lot like it. There is an exposed public function on the page script layer called DialByLine with the following parameters: /** * Primary method for making a call. * @param {string} type = (required) Either "audio" or "video". Will setup UI according to this type. * @param {Buddy} buddy = (optional) The buddy to dial if provided. * @param {sting} numToDial = (required) The number to dial. * @param {string} CallerID = (optional) If no buddy provided, one is generated automatically using this callerID and the numToDial * @param {Array} extraHeaders = (optional) Array of headers to include in the INVITE eg: ["foo: bar"] (Note the space after the : ) */ function DialByLine(type, buddy, numToDial, CallerID, extraHeaders) Also try the Github issues page: github.com/InnovateAsterisk/Browser-Phone/issues

@@InnovateAsterisk yes I totally agree with u but... my intention was to implement audio and video calls in my app using webRTC with astrick server am wondering how to make app calls in a web app.