Any working SIP softphone for FreeBSD?

mickey

Aspiring Daemon

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I am looking for a working, lightweight SIP softphone client that runs on FreeBSD. Right now I am using net/linphone which works but has some audio buffer problems and I don't like the UI one bit. I also tried net-im/ekiga which I had been using on Windows many years ago, but it fails to detect any sound devices whatsoever, rendering it useless. Every other solution I found searching the internet so far, either has no FreeBSD port or it had one which has been removed (net/kphone, net-im/jitsi) for one reason or another. On Windows I had been using MicroSIP, but unfortunately it's only available for Windows, Android and iPhone.

Any suggestions what else to try?
 
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mickey

mickey

Aspiring Daemon

Reaction score: 301
Messages: 747

I don't use it myself, found it in ports: audio/baresip
So I installed audio/baresip to give it a try. A commandline interface was not exactly what I was expecting, but I had it configured and working in no-time, including account setup to hook it up to my IP PBX (asterisk). So far it looks very promising, audio quality is better than in net/linphone (most importantly it does not have those buffer artifacts, where when a call is hung up, part of the audio gets cut off and plays when the next call starts) and also it has less dependencies. Like A LOT less. HD telephony using G.722 wideband codec also works right out of the box.
 
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mickey

mickey

Aspiring Daemon

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Messages: 747

It is PJSUA. It is an option in PJSIP.
I have been using PJSIP as part of asterisk for quite some time now, but I wasn't aware that it has a commandline softphone client too. Guess I will check it out. Thanks.
 

getopt

Aspiring Daemon

Reaction score: 616
Messages: 866

It is PJSUA. It is an option in PJSIP.
For those who are trying to make a use of this hint:

PJSUA is a port option of net/pjsip. You need to compile the port with having set the default option from off to on.
Code:
PJSUA=off: Command line SIP agent
 
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