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The projects demonstrates how tenants softphones and FreeSWITCH PBXs can be connected to each other using Kamailio SIP router to enable advanced calls routing, logging, measurements and NAT traversal.

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## Simple, Hackable, Containerized Kamailio, RTPEngine and FreeSWITCH Configuration

This project demonstrates how Kamailio, a SIP router, can connect tenant FreeSWITCH PBXs and softphones. It enables features like advanced call routing, logging, measurements, and NAT traversal.

## Main Use Case

Remote softphones register with Kamailio, which routes SIP requests to pre-configured FreeSWITCH PBX. When a call arrives to FreeSWITCH, it rings both the local and remote softphones simultaneously and/or sends call to a PSTN gateway (Twilio). Local (internal) softphones can also make calls to remote softphones through Kamailio.

RTPEngine is used as media proxy to ensure devices behind symmetric NAT can send media to each other and to bridge traditional SIP clients with WebRTC clients.

```
|----------|   SRTP/DTLS    |------------| 
| Browser  |<-------------->|            |
| SIP.js   |----\   (S)RTP  | RTPEngine  |<-------|
| Ext 1002 |     \/-------->|            |        |
|----------|     /\         |------------|        |
                /  \              | NGCP          |                            
|----------|   / SIP\WS\WSS |------------|        |
| Mobile   |<-/      \----->|            |        | (S)RTP
| Linphone |  SIP\TCP\TLS   | Kamailio   |        |  
| Ext 1001 |--------------->|            |---------->|------------|
|----------|                |------------|        |  |            |
                                  | SIP\TCP\TLS   |  | PostgreSQL |
|----------|  SIP\TCP\TLS   |------------|        |  |            |
| Desktop  |--------------->|            |---------->|------------| 
| Linphone |    (S)RTP      | FreeSWITCH |<-------|  |------------|
| Ext 1000 |<-------------->|            |  ESL,REST |Configurator| 
|----------|                |------------|---------->|------------|
```

## Implementation

* Scalable SIP router and media proxy configuration
* Persistent TCP/TLS/WSS signaling connections and secure media
* SIP requests are routed to the specific instance where the target softphone/PBX is registered
* Locations and extension mappings are stored in a shared PostgreSQL database
* Uses standard Kamailio modules only: usrloc, registrar, postgres, sqlops, nathelper, tls
* Tested with Linphone, Zoiper softphones on Linux and Android and SIP.js WebRTC client
* FreeSWITCH uses a Node.js configurator app, that serves dialplan via REST API
* FreeSWITCH is configured to use PostgreSQL for core, sofia, and voicemail modules

## Usage

Follow the instructions in the `README.md` files (in order) to build Docker images and start RTPEngine, Kamailio, and FreeSWITCH. Update configurations if your network is not 192.168.0.0/24.

Once running, Kamailio and FreeSWITCH should be listening on standard ports (5060, 5061, 5080) and RTPEngine on ports 2223-2225 (verify with `netstat -ntlp` or `nulp`). Configure PBX mappings in the database, register softphones (extensions 1000-1002, password 12345) on Kamailio (external) or FreeSWITCH (internal), and start making calls.

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