- Part 1
- Part 2
- Client setup checklist
- Linux softphone comparison
Linux softphone comparison and the snom 370
January 1st, 2009
Updated June 17th, 2009
Updated January 2011: We now recommend Ekiga as our main SIP client choice. It works well on various platforms (Windows and Linux) and has good video calling support on both. The instability we observed a couple of years ago in the snom phones is gone. They are now running recent releases of version 8 firmware, and it is extremely stable, with good sound quality. I have removed information about outdated software packages.
In this series, we have set up Asterisk, OpenVPN and a business-quality SIP desktop phone. We also set up KPhone on Linux. Here's a quick comparison of the other softphones we tried in Linux.
KPhone
KPhone was discussed in the original (late 2008) version of this article, but as of now, we consider it to be obsolete and do not use it. Development of KPhone ceased several years ago.
Linphone
Linphone was discussed in the original (late 2008) version of this article, but as of now, we consider it to be obsolete and do not use it.
The winner: Ekiga
Ekiga won in the original (late 2008) version of this article, and it wins even more today. It's been around for a while, formerly as GnomeMeeting. Ekiga is under active development. It has support for many SIP capabilities, including video and (like all modern SIP software) out-of-band DTMF. When I originally wrote this article, I was unable to get Ekiga to register to receive inbound calls. However, after upgrading to Ubuntu 9.04 (Jaunty) I decided to try again. Jaunty comes with Ekiga 3.2.0, so that is the version covered in this review. Later versions of Ekiga all register easily with Asterisk.
Installation was trivially simple; it
doesn't get any easier than apt-get install ekiga.
I
configured OpenVPN as normal (see the other
sections of this article). I did not turn on
routing on the host, because Ekiga will be running
on the host machine itself. If I had wanted to use
the host (my laptop) to route OpenVPN for other
machines, I could have turned on routing.
Getting Ekiga to register was also trivially simple.
I added a block in /etc/asterisk/sip.conf
to allow it to log in. Note: the section
name in /etc/asterisk/sip.conf should be the
same as the auth name the SIP client sends. I also
added the extension in the appropriate places
in my dialplan in extensions.conf.
I went to the Accounts menu in Ekiga, and added the account,
using 10.8.0.1 (the server's OpenVPN address) as the SIP
registration server.
The Ekiga softphone registered as expected and I tested both incoming and outgoing calls successfully. It couldn't be any easier. Sound quality is good.
And Ekiga has other great aspects as well. It runs on both Linux and Windows, so we can use one softphone for our entire user base. It's simple enough that ordinary users can install it and use it. You can't beat the price. Video is a great built-in feature.
Snom 370
Updated January 2011
Unlike the other options in this article, the snom 370 is a hardware phone. It runs Linux, although users and administrators do not interact with Linux directly on the phone. There's no way to get a shell, for example.
But it is Linux, and users can download a software image that includes OpenVPN support. We found it to be fairly easy to follow the instructions and configure both OpenVPN and SIP on the phone. The trick is to get OpenVPN working first. The easiest way to do that is to put together an OpenVPN configuration directory on a computer, test the tunnel to see that it works, and then move that configuration to the snom 370. At that point, configuring SIP is simple. Make it connect to the server's VPN end point, which is 10.8.0.1 in this series of articles.
The phone itself looks modern and professional, has all the expected features, has an easy-to-use web interface, and works well. I would recommend it for any office which needs a standalone phone that runs both OpenVPN and SIP. In fact, it can at as an OpenVPN router for other devices, by connecting them to its second interface.
The phone costs just over $200 from on-line sources. This makes it affordable as a basic office phone. The built-in OpenVPN feature means that the Asterisk administrator can configure it and let an employee take it home, plug it in to the home DSL, and suddenly have a working extension at a reasonable cost.
When we first reviewed this phone (late 2008, the 7 series software) we encountered occasional software crashes, requiring the phone to be unplugged and rebooted. The article said, "I suppose that when you have so much software in a device, there are more opportunities for reliability problems. I hope it will become more stable with future software updates." Two years after I wrote that, the price is now very reasonable for a professional office phone, and the software is very stable. We are now using firmware 8.4.x in all our snom phones and it is stable, with good sound quality. Fortunately it is very easy to do firmware updates on these phones.
We recommend the snom 870 as an all-around office phone. It's more expensive (around $300) but it looks great, and is easy to use with a touchscreen. The web-based UI, and the core phone software, is very similar to the 370, so it's easy to administer a mix of 370s, 820s, and 870s.
Conclusion
KPhone worked, but stopped active development several years ago, and should no longer be considered an option. Linphone also seems to lack active development.
Ekiga works very well, works on both Linux and Windows, and is easy to use. It also supports video calls. It has been under continuous active development for years.
The snom 370 is a good option for users who want a standalone phone that supports OpenVPN. Two years ago, we said, "The drawbacks of the snom 370 are high cost and also some instability." Today, we will say that we like the low price and software stability. If you're looking for an office phone that looks more modern and has a great touchscreen interface, we recommend the snom 870.
If you need help designing and implementing a voice application or PBX based on Asterisk and SIP, contact us at 310 356 7869 to discuss.