Baresip docker container for headless Call Testing
- Baresip v0.6.6
- HTTPD Module
- CONS Module
- GST Module
- ALSA Loopback (snd-aloop)
docker pull qxip/baresip-docker
docker run -name baresip --rm -ti --device=/dev/snd:/dev/snd qxip/baresip-docker
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
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
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
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)
Baresip is Copyright (c) 2010 - 2020 Creytiv.com Distributed under BSD license