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https://github.com/qxip/baresip-docker

Baresip docker container (baresip git + libre + librem )
https://github.com/qxip/baresip-docker

baresip docker httpd

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Baresip docker container (baresip git + libre + librem )

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baresip-docker
==============

Baresip docker container for headless Call Testing

## Features:

* Baresip v0.6.6
- *HTTPD Module*
- *CONS Module*
- *GST Module*
- *ALSA Loopback (snd-aloop)*

## Install:
```
docker pull qxip/baresip-docker
docker run -name baresip --rm -ti --device=/dev/snd:/dev/snd qxip/baresip-docker
```

## Usage

### BARESIP CLI
##### Command Line Interface
Register two SIP accounts to create a route loop initiating and terminating at our agent
```
/uanew sip:[email protected];auth_pass=mypassword;answermode=auto
/uanew sip:[email protected];auth_pass=mypassword;answermode=auto
```
Once registered, Dial your loop from 100 to 200
```
d 200
b
```
Check out the call statistics for both legs

-------------

### HTTP API
##### Webserver User-Interface (UI) using HTTP Socket
This module implements an HTTPD server for connecting to Baresip using HTTP Protocol.
You can use programs like CURL to connect to the command-line interface.

Register two SIP accounts to create a route loop initiating and terminating at our agent
```
# curl http://127.0.0.1:8000/?/uanew%20sip%3A100%40sip.host.com%3Bauth_pass%3Dmypassword%3Banswermode=auto
# curl http://127.0.0.1:8000/?/uanew%20sip%3A200%40sip.host.com%3Bauth_pass%3Dmypassword%3Banswermode=auto
```
Once registered, Dial your loop from 100 to 200
```
# curl http://127.0.0.1:8000/?d%20200
# curl http://127.0.0.1:8000/?b
```
Check out the call statistics for both legs

-------------

### CONSOLE
##### Console User-Interface (UI) using UDP/TCP sockets

This module implements a simple console for connecting to Baresip via UDP or TCP-based sockets.
You can use programs like telnet or netcat to connect to the command-line interface.

Example, with the cons-module listening on default port 5555:

```
# netcat -u 127.0.0.1 5555
/uanew sip:[email protected];auth_pass=mypassword;;answermode=auto
/uanew sip:[email protected];auth_pass=mypassword;;answermode=auto
d 200
b
```

Check out the call statistics for both legs

-------------

### Call Statistics
Upon hangup, each leg will produce RTCP based statistics. Since both user-agents are under Baresip control, statistics are reasonably reliable:
```
sip:[email protected]:59989: Call with sip:[email protected] terminated (duration: 40 secs)

audio Transmit: Receive:
packets: 1798 1782
avg. bitrate: 63.8 63.6 (kbit/s)
errors: 0 0
pkt.report: 1648 1632
lost: 0 0
jitter: 0.0 0.1 (ms)

sip:[email protected]:59989: Call with sip:[email protected] terminated (duration: 40 secs)

audio Transmit: Receive:
packets: 2094 1793
avg. bitrate: 63.8 60.6 (kbit/s)
errors: 0 0
pkt.report: 1846 1641
lost: 0 0
jitter: 0.0 0.0 (ms)
```

-------------

## Credits
Baresip is Copyright (c) 2010 - 2020 Creytiv.com Distributed under BSD license