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:100@sip.host.com;auth_pass=mypassword;answermode=auto
/uanew sip:200@sip.host.com;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

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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

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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:100@sip.host.com;auth_pass=mypassword;;answermode=auto
/uanew sip:200@sip.host.com;auth_pass=mypassword;;answermode=auto
d 200
b

Check out the call statistics for both legs

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Call Statistics

Upon hangup, each leg will produce RTCP based statistics. Since both user-agents are under Baresip control, statistics are reasonably reliable:

sip:100-0x17efe40@172.17.0.2:59989: Call with sip:200@10.0.0.1 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:200-0x17efe40@172.17.0.2:59989: Call with sip:100@10.0.0.1 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)

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Credits

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