This was my first time using baresip for voice call. It was very simple, I just created an account at linphone.org and add my user and password to ~/.baresip/accounts:
<sip:[email protected]>;;auth_pass=mypasswordacassis@dev:~$ baresip
baresip v1.0.0 Copyright (C) 2010 - 2020 Alfred E. Heggestad et al.
Local network address: IPv4=wlp0s20f3|192.168.1.11 IPv6=wlp0s20f3|2804:7c78:4b:5e00:67c1:b6af:d375:9204
aucodec: PCMU/8000/1
aucodec: PCMA/8000/1
ausrc: alsa
auplay: alsa
medianat: stun
medianat: turn
medianat: ice
Populated 1 account
Populated 3 contacts
Populated 2 audio codecs
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
baresip is ready.
[email protected]: {0/TLS/v6} 200 Registration successful (Flexisip/2.4.1-34-g8eb3e330 (sofia-sip-nta/2.0)) [1 binding]
All 1 useragent registered successfully! (1403 ms)
/dial sip:[email protected]
ua: using best effort AF: af=AF_INET6
call: connecting to 'sip:[email protected]'..
call: SIP Progress: 100 Trying (/)
ua: using AF from sdp offer: af=AF_INET6
sip:[email protected]: Incoming call from: sip:[email protected] - (press 'a' to accept)
call: SIP Progress: 180 Ringing (/)
terminated by signal 2
ua: stop all (forced=0)