<span style="display:block;height:15px!important"></span>
<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p>

<span style="display:block;height:20px!important"></span>

**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It connects to an Asterisk server to make and receive phone calls using SIP over WebSocket and WebRTC.**

For calls to and from regular phone numbers, a telnyx.com or localphone.com account is needed and a real phone number acquired from one of the two providers of SIP services. 
Nextcloud must use HTTPS. Asterisk has to be installed on a VPS or dedicated server. Once SIP Trip Phone gets connected to Asterisk, Asterisk can be connected to any SIP provider, 
but SIP Trip Phone has been tested only with Telnyx and Localphone. This application is based on the ctxSip phone.

<span style="display:block;height:20px!important"></span>

## Main Features
<span style="display:block;height:10px!important"></span>

* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can be even 90% cheaper.

* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls.

* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server.

* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls.

* 🚩 On-screen notifications on incoming calls.

* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud.

* 💻 On the underlying Asterisk server you can implement an IVR (Interactive Voice Response or voice menu) and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc.

* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts.

* 💸 Low per minute prices: you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone).

<span style="display:block;height:10px!important"></span>

Double Bastion is not affiliated with Telnyx or Localphone.

<span style="display:block;height:20px!important"></span>

<p align="center">Initial screen</p>
<span style="display:block;height:10px!important"></span>
<span style="display:block;margin:auto;width:412px;">![Image of SIP Trip Phone Interface](https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_initial_screen.png)</span>
<span style="display:block;height:40px!important"></span>
<p align="center">Dialpad</p>
<span style="display:block;height:10px!important"></span>
<span style="display:block;margin:auto;width:412px;">![Image of SIP Trip Phone Interface](https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_dialpad.png)</span>
<span style="display:block;height:40px!important"></span>
<p align="center">Making calls</p>
<span style="display:block;height:10px!important"></span>
<span style="display:block;margin:auto;width:412px;">![Image of SIP Trip Phone Interface](https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_making_calls.png)</span>
<span style="display:block;height:40px!important"></span>
<p align="center">Transferring calls</p>
<span style="display:block;height:10px!important"></span>
<span style="display:block;margin:auto;width:412px;">![Image of SIP Trip Phone Interface](https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_transfer_call.png)</span>
<span style="display:block;height:40px!important"></span>

## Browsers
<span style="display:block;height:10px!important"></span>

SIP Trip Phone works with all the major browsers.

<span style="display:block;height:20px!important"></span>

## Programming Languages
<span style="display:block;height:10px!important"></span>

SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug.

<span style="display:block;height:20px!important"></span>

## Minimum Requirements
<span style="display:block;height:10px!important"></span>

- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide.

- **A telnyx.com or localphone.com account and a phone number** associated with it.

- **Asterisk** (with **chan_pjsip** enabled) installed on a VPS or dedicated server.

You can also install **Coturn** as a STUN server, which helps when callers are behind routers.

<span style="display:block;height:20px!important"></span>

## Installation
<span style="display:block;height:10px!important"></span>

<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server 
explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn.

SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all 
the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>.

<span style="display:block;height:20px!important"></span>

## Contribute
<span style="display:block;height:10px!important"></span>

This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone
repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for
digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits.

If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, 
mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes we can include them in the 
project.

Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank">
issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com .

<span style="display:block;height:20px!important"></span>

## License
<span style="display:block;height:10px!important"></span>

SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with 
the terms of the GNU Affero General Public License Version 3.

This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files.

SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices.
