# Any working SIP softphone for FreeBSD?



## mickey (Sep 4, 2020)

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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## T-Daemon (Sep 4, 2020)

I don't use it myself, found it in ports: audio/baresip









						GitHub - baresip/baresip: Baresip is a modular SIP User-Agent with audio and video support
					

Baresip is a modular SIP User-Agent with audio and video support - GitHub - baresip/baresip: Baresip is a modular SIP User-Agent with audio and video support




					github.com


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## Lamia (Sep 4, 2020)

mickey said:


> On Windows I had been using MicroSIP,


MicroSIP I based on PJSIP.
You may try get one of these projects running in your box in order to enjoy the same set of features.


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## getopt (Sep 4, 2020)

Lamia said:


> ou may try get one of these projects running in your box in order to enjoy the same set of features.


Lamia 
as the OP asked


mickey said:


> for a working, lightweight SIP softphone client that runs on FreeBSD


which one works on FreeBSD?


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## mickey (Sep 4, 2020)

T-Daemon said:


> 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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## Lamia (Sep 5, 2020)

getopt said:


> Lamia
> as the OP asked
> 
> which one works on FreeBSD?


It is PJSUA. It is an option in PJSIP.


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## mickey (Sep 5, 2020)

Lamia said:


> 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.


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## getopt (Sep 5, 2020)

Lamia said:


> 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.

```
PJSUA=off: Command line SIP agent
```


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## Lamia (Sep 5, 2020)

It's version two of PJSUA. And being a high level API too, you make find it difficult to use. 
Here is a documentation on it - https://www.pjsip.org/pjsua.htm . It must have been updated for the version two.


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