Is Linux Voice over IP Ready?

A Voice over IP primer with special attention to using it on Linux.

What is Voice over IP really? What do you need for Voice over IP? What do you mean, I can't call my girlfriend? What's all that buzz about open and proprietary protocols? Can I start my own telecom service? This article addresses these questions and compares the most popular Linux applications for calling and conferencing.

What Is Voice over IP about?

Internet or digital telephony, or Voice over IP, often abbreviated as VoIP, allows parties to exchange voice data streams over the network. The big difference is that with VoIP the data flows over a general-purpose network, the Internet, whereas conventional telephony uses a dedicated network of voice transmission lines.

Under special circumstances, a VoIP network can be connected with the conventional telephone network. However, at the time of this writing, that is certainly not the standard. In other words, it is likely you will not be able to call people who are using a conventional telephone. Although currently various applications are available, both free and proprietary, telephony over the Internet has some major drawbacks. Most noticeably, the system is unreliable, it can be slow, or there can be a lot of noise on the connection. Therefore, it cannot be used to replace conventional telephony. Think about emergency calls, for instance. Although some providers take some precautions, there is no guarantee you will be able to reach the party you want to call. This is worsened, because in VoIP, there is no agreement yet on a standard for assigning numbers, like the E.164 standard we have for assigning and identifying traditional land lines and mobile phone numbers.

Even if there is some form of integration between VoIP and conventional telephony networks, this is still different for mobile networks. The major problem is that wireless network coverage is not as well developed as cellular network coverage. Additionally, there is the issue of costs when accessing the Internet from your mobile phone. For me, it would amount to 0.50 EUR (+/- $.60 US) per 100K of traffic. It is possible that integration of VoIP in the third-generation telephony network will ease these troubles.

You also should be aware that there is no encryption in VoIP. So, it is fairly easy for anyone to eavesdrop on conversations.

The bottom line is although VoIP is useful, it is not a replacement for land-line telephony (yet).

Let's look at what you'll need to get VoIP up and running.

On the Server Side

First of all, you need a provider offering the service. Some popular providers offer the service for free, and some require a subscription fee. Among the free ones are the following: SIPphone, Skype, SIP Broker and Google.

Most free services, however, do not allow you to connect with the conventional telephone network. This so-called full phone service is usually not free. Among the most popular full phone service providers are the following: Vonage, Lingo, AOL TotalTalk and SIPphone.

These lists are certainly not exhaustive, as new local and global providers join the pool on a near-daily basis. Also, many SMEs are currently setting up a VoIP network for internal use within companies. If you want to set up your own VoIP network, you might want to look into Asterisk server software or sipX, which are open-source PBX implementations.

Alternatively, if you want to use only the soft phone, meaning the audio system of your computer (audio boxes and microphone or headset) and accompanying software, check out Ekiga, formerly GnomeMeeting, as announced January 8, 2006, in the GnomeMeeting blog. Although Ekiga supports a range of hardware, it is usually set up to support (video) conferencing features implemented on the software level. Like sipX and Asterisk, it is open-source software.

Note: PBX stands for Private Branch eXchange, the system that centralizes all of a business' telephone sets.

Figure 1. A Typical VoIP Solution (image courtesy of BroadVoice)

On the Client Side

Depending on your network architecture, some applications might work better than others, due to the protocols they use. Most standards-based solutions use either the H.323 or Session Initiation Protocol (SIP). Apart from these two standards, there are a lot of proprietary protocols, such as Skype (from the company with the same name) and SCCP from Cisco. The main difference between them is that SIP stores the client IP address in its packages, resulting in difficulties when you are behind a firewall.

Microsoft NetMeeting and GnomeMeeting use H.323; Microsoft's Messenger and Apple's iChat and SIPphone uses SIP. Server software usually implements several different protocols.

Apart from your network architecture, available bandwidth also might be a limiting factor, as some applications are optimized for low bandwidth, and others expect to be on a broadband connection. This depends on the codecs the VoIP systems use for handling sound.

As far as client hardware is concerned, use a headset. Although your PC, especially if it is a laptop, might have a microphone and speakers built in, you will be far more comfortable using a headset, as it will suppress echo and noise from your environment. If you have the choice, opt for a USB headset. A USB headset is a separate audio device to your system. It functions independently from existing audio hardware, so it avoids any conflicts that might occur between VoIP and normal sound processing.

If the applications you use provide the features, you can redirect audio streams as desired. For instance, you can make the ring tone for alerting you that you have a call come through the normal speakers. When you pick up the call, the voice of the calling party is redirected to your headset.



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Hi all. I've always been a

Ben S's picture

Hi all.

I've always been a Windows user, and with the latest system crash (lsass.exe error following a registry defrag) had to reformat and reinstall everything. I've been reading a lot about Linux and am tempted to try it. I've listed the features that I usually use and got down to 2 that I can't seem to replace: MSN and voip.
For MSN, I found aMSN, but as far I've read it doesn't support voice calls. I know that there is Skype, but my girlfriend's internet is really bad and Skype calls won't work. They work in MSN though.
As I live abroad, and my parents are not computer-savvy, I use a VOIP provider that allows me to make free calls to landline phones (don't know if I can mention the name here), which saves me a lot of money. However, they don't seem to have the interface for Linux - they have a link for Mac and Linux users, ask for a signup and that's it. It's a dead end. I was wondering if you are aware of any voip provider that allows free calls and operates under Linux.
If not, I guess I'll just wait some more time.

Globe7 has Launched its

bradley's picture

Globe7 has Launched its Linux version in different versions where calling PC-PC is free and call landline/mobile at world's lowest rates.
Enjoy 1GB free space by uploading your favorite videos.Enjoy the other features like SMS,Watching Internet TV,SMS,Follow ME services.



Anonymous's picture

I have tried this,it is obsolutely good,watching TV and calling any destination is really great.

Linux Voip

cherry7's picture

I Have used Globe7 Linux softphone,where Voice Clarity is Excellent,calling PC-PC is Free and calling to any Landline/Mobile are at low rates.Can make Live Chats and send SMS,Create our own channel,communities.

Linux Voip

cherry7's picture

I Have used Globe7 Linux softphone,where Voice Clarity is Excellent,calling PC-PC is Free and calling to any Landline/Mobile are at low rates.Can make Live Chats and send SMS,Create our own channel,communities.

Disappointed . . .

bbc's picture

There are lots of confusing and misleading statements in this article. If you are going to spend time reading it, make sure you verify anything it says with your own research. The author didn't do that for you.

As for VOIP . . . I have been using Asterisk for two years. It works great. It talks to about any service. It talks to about any softphone. It has hardware interfaces for FXO and FXS. And, it is well supported . . . and tough as nails.

As for Linux as a desktop, voip/sound platform for softphones . . . pick your sound cards wisely. Don't get anything cheesy . . . and don't get anything that is really "edge of the envelope" . . . It likely won't work. BUT . . . with pretty standard sound-blaster-ish hardware, every softphone I tried worked fine on Ubuntu.

One thing the author says that I definitely don't agree with is the seperate USB sound device. That is a bunch of B.S. The author said that because it sounded good . . . and/or it works on Windows. It doesn't work well in Linux at this time.

And, forget the B.S. about encryption . . . SIP 2.0 supports it natively. All you have to do is turn it on.

Last but not least . . . is Linux ready for VOIP . . . "You bet it is!" It is so ready that I have seen dozens of products based upon Linux/Asterisk/SER and many others . . . and I would know . . . I work for a phone company.

What keeps Linux from being the VOIP desktop/softphone platform of choice isn't VOIP readi-ness or sound quality . . . It is the adoption of Linux as a desktop.

All in all, the author was lazy . . .

I think you completely

Caglar's picture

I think you completely forgot that Skype, Wengo, Gizmo and all these other programs do NOT allow to log into another server but their own ones. I have a Voip account from a provider, which has its own server, but Gizmo, Skype... will never allow me to use their software with another provider.
If you want to do this, you really have to use Ekiga oder KPhone. Ekiga is the best choice for me, since it has a really cool user interface which Kphone doesnt have. Xlite is also able to use other servers, but it isnt able to hide into the system notification tray, which makes it very unusefull if you want to be reachable all the time.
I think projects like Skype, Gizmo and Wengo should get a little more open and allow the user to chose THEIR provider.

Way off base

Brian A. Vincent's picture

Wow. I really find this article way off base and completely missing the whole concept and state of the art in VOIP.

Okay, there's some good info in here. But there needed to be a huge disclaimer up front that says, "This article is not about Voice Over IP as much as it is about various software clients you can use on your computer."

The state of VOIP is way, way beyond this article. VOIP not connected to PSTN is completely useless. Sure, it might be cute, fun, and possibly let you talk to Mom and Dad, but useless. The real state of VOIP completely integrates with the PSTN and you can call anyone, anywhere and likewise they can call you. The real point of all these SIP clients is to connect on the backend to a real PBX, like Asterisk or even something really advanced like I3's CIC platform. Register up and call anyone in the world.

Wanna talk about VOIP? Talk about Asterisk. Talk about Polycom sets. Talk about Cisco switching to SIP. Discuss real telephony features, like integration with speech recognition and voicemail. Talk about the things that people are barely thinking of.. text to speech applications using festival and advanced ACD options like multisite call centers with automatic load balancing of queues.

Ain't nuthin' interesting about a SIP client. The magic is in the backend and Linux is beginning to kick some serious ass there.


David Abreu's picture

what do you mean by PC-to-phone calls possible (!?) how's that? could you explain us a little more, please.

What about Linphone?

Anonymous's picture

An article about Linux VoIP, but not mentioning, which became quite mature since a couple of time.

This disqualifies the whole article.

the biggest feature of skype

Viktor's picture

The article left out one huge difference between the mentioned programs, and this difference is its subscriber base.

I tried before Gnomemeeting (at that time it was called like this), then I tried KPhone, later on Skype, and finally I installed Wengo and Gizmo too, but today I am only using Skype. Why? Because all my friends use it too!

We are talking about communications, a social phenomena that can be modeled using some sort of networks. (Like network economics.) And from the point of view of the badly informed end user this is the critical fact: how many people use the same program. It is not about protocols, server and client sides, no. The main feature is that I want to be connected, and Skype gives it to you.

Re: the biggest feature of skype

Anonymous's picture

the biggest and most important feature of Skype is encryption which is enabled by default.

How gullible. Everyone who

Anonymous's picture

How gullible. Everyone who cares about security would never trust a methode for encryption, if you dont know how it works. And as it is closed source, you have to trust them to use good encryption and not to deceive you. And give me reasons why to trust Skype Tech. Even worse, by using Skype you are let to feel safe, but Skype Tech. can always decrypt your data. That is one of the best reasons not to use Skype.

Better than nothing

Anonymous's picture

While I fully agree with your sentiment about secret encryption methods, having those is for me at least better than nothing (like Ekiga, sadly). With Skype, at least only the company can spy on me, and not every router on the way.

RE: Biggest Feature Of Skype

Len Morgan's picture

Your argument that Skype is best because it has the most users (right now) is the same argument countless IT managers use to continue using Windows instead of a better, more stable OS (like Linux). Pure numbers don't make a better product, just an arrogant supplier.


Anonymous's picture

The article completely fails to mention the potential security issues.
H.323 and SIP require that the system firewall be opened to incoming UDP connections from each and every potential call partner for all ports from 1024 upwards. The 'each and every' could probably be best translated as 'from anywhere'.
This could expose a lot of potentially vulnerable applications to incoming packets.

Opening ports...

Toby Haynes's picture

H.323 and SIP require that the system firewall be opened to incoming UDP connections from each and every potential call partner for all ports from 1024 upwards. The 'each and every' could probably be best translated as 'from anywhere'.

That's so far from true it's not funny. There are a few ports you *might* need to punch open for comms with Netmeeting but we are talking unprivileged ports 5000 to 5100 if you happen to run Symmetric NAT (about 1% of routers).

STUN support cuts down this need entirely. From the Ekiga FAQ:

Ekiga has extensive and improved NAT support thanks to STUN. In 99% of the cases, you do not have any configuration to do, and you can even be reachable from the outside without any port forwarding.

Go read up at before spouting more uninformed comment.

Toby Haynes

Skype and lots of connections

Anonymous's picture

I've noticed Skype making a lot of connections and wonder if it was using my machine as part of the skype architecture. Even with the firewall blocking incoming connections its hard to block outgoing ones without stopping Skype altogether, so I no longer run it. I'm on a fairly limited broadband account at the moment and don't want to run out of bandwidth allocation.

A firewall will not stop

ka9yhd's picture

A firewall will not stop Skpe from connecting to its servers.

I suggest you read this article about how Skype and other P2P applications are poking holes in firewalls to communicate.

Yes, Skype can make your computer into a server

Anonymous's picture

Yes, Skype will take your bandwidth if it wants to use your machine as a server. You don't have control over that.


Anonymous's picture

Despite only offering an *ancient* version for Linux, the one feature Skype offers, which no one else seems to have figured out is a basic requirement, is end-to-end encryption of all communications.

Why the heck isn't there some encryption functionality built into *every* VOIP app by default?!?

Re: Encryption?

Lindsey Rockwell's picture

Gizmo Projct is one with encryption. You can always install Zphone in both ends to gain "safe lines".

Gizmo is propritary

ka9yhd's picture

Gizmo is propritary software, that is.... you can only use it with their service and no one elses.

Linux isn't audio-ready.

Anonymous's picture

You still cannot count on hearing your phone ring, or not conflicting with a music player or Flash website that happens to be loaded..... because the OS allows any application that happens to be using /dev/dsp to block audio.

Software audio mixing should be standard behavior, but it isn't with Linux. So users must know about wrapper-commands and buzzwords and acronyms, and keep track of which type of audio app is running at any given time, in order to make these apps work together. And they are rudely told they didn't buy the "right" soundcard with multichannel hardware for $50 extra.

Audio output shouldn't require ANY effort from the user to 'manage' the audio-capable programs they may have running.

Here is another person's experience with Linux audio.

That's a problem with OSS,

Anonymous's picture

That's a problem with OSS, which was depreciated years ago.

These days decent applications use ALSA, which has software mixing capabilities.

If ALSA doesn't take over

Anonymous's picture

If ALSA doesn't take over the /dev/dsp functionality, then that's ALSA's fault. In the end, there's simply no excuse. It's a gross oversight to assume end-users will learn what OSS and ALSA are, and then uninstall anything that uses OSS.

> That's a problem with OSS,

Anonymous's picture

> That's a problem with OSS, which was depreciated years ago.

So if that is still a problem today then it's ALSA's fault!. ALSA completely sucks!. Have you ever tried programming with ALSA?. To write a 25 line piece of OSS code I have to write about 50 lines in ALSA and they do the same thing.

"So if that is still a

Anonymous's picture

"So if that is still a problem today then it's ALSA's fault!"

No its the fault of all the applications that still use OSS, Skype being ther perfect example of this.

A completely useless article

Anonymous's picture

A completely useless article from someone obviously unaware of VoIP technologies and applications. Get a clue or do a minimum of research on the subject. It's apparent she has absolutely no idea what she's talking about. How many calls centers are running Asterisk
and SER today? Plenty! I really expected more from LJ.

SER and Asterisk

Anonymous's picture

I cannot tell you which VoIP company I work for (not Vonage) but I can assure you that Linux, SER and Asterisk are a VERY large part of our infrastructure. Now if we can get rid of the Solaris and the Windows, we will be really able to take off.

Bad article

Anonymous's picture

I agree. Not only is it usless its confusing. She mentions gnomemeeting as being H323 only in one part then goes on to mention that its both H323 and SIP, she moves between the old and new name where she should use the new name. I'm also not aware that iChat uses SIP, nor does Microsoft Messenger (the free version) but the Live Communications server/client does but isn't free.

Ekiga no longer uses openh323 but rather OPAL and she mentions all the testing was done on Fedora. If that was the case you don't need to actually worry about dependancies as the OS deals with that. And by using Fedora why in the kphone article mention installing it using Synaptics on Ubuntu?

In general a disapointing article. While VoIP isn't the easist to setup most distros comes with the hard work done.

SJPhone is also good, much

Anonymous's picture

SJPhone is also good, much simpler interface (but just as featureful) as X-Lite:


Skibum - Derek Jacobs - CounterPath Support's picture

Hello Machtelt

I just thought you should know that it is possible to do IP to IP calling with X-Lite. Here are some brief instrictions:

Make a new SIP Proxy with the following details

Enabled: Yes
Display name: Harry (example)
Username: harry (example)
SIP Proxy: Your local IP
Direct Dial IP: Yes

Leave everything else alone.

Calling is as follows:


Gizmo Project from SIPphone

Anonymous's picture

I'm suprised you didn't review Gizmo (free download from SIPphone).

Because SIPphone was started by Michael Robertson, founder of Linspire (popular desktop Linux distro), he has always made sure Gizmo worked on Linux. It's certainly, in my view, the strongest free VOIP client out there, particularly for Linux, and offers both basic free service, and the option for advanced, paid-for services.


And what about OpenWengo?

Anonymous's picture

With the GPL code, SIP audio, h263 video, Jabber text and alike Skype/Gizmo features (call out, call in), this application could be a real challange for the other ones. As a right now you can test the 'classic' version while 'NG' (next generation) is under heave developement.
Have look at WengoPhoneNg Overview.

Thats right....Gizmo is

Chandra's picture

Thats right....Gizmo is working much better than sucks too much of RAM and also it doesnt co-operate well with other sound-applications....Gizmo sounds best for linux.

Gizmo and Twinkle

Hans's picture

I second the recommendation of gizmo. Gizmo is not only a decent sip softphone, but it also provides the same services Skype does, without being evil. Another great linux sip softphone is Twinkle.


JID's picture

Sipphone does allow access Ma Bell system for a small fee.
It is my main long distance at the moment.

Openwengo is the best GPL

Anonymous's picture

Openwengo is the best GPL VOIP client with lot features including firefox plugin :) Linux/Mac/Windows

OpenWengo is not the best GPL

Anonymous's picture

The best GPL client on Linux is from far Ekiga. Sorry to disappoint you, but I tried OpenWengo NG on Linux, and it was a big deception for me.

Go Ekiga!

VOIP : Compatibility and Choice is what End-Users Crave for?

Anonymous's picture

I will first of all congratulate the writer for pointing out some facts about VOIP and it application and choice of client software for the End-User.

Yes, VOIP has matured to a larger extend than this article shread light on, but for a new or average linux user who has head about something new called VOIP that all of his friends are using, we certainly do not want to turn them away from buzzing words and unnecesary jargons.

The truth is Skype on Linux is unstable and I have had similar experience with Skype as outline by the author. In addition, I believe what most Linux users as well as programmers are concerned about is the proprietary nature of the skype protocol. We End-Users of today crave for choice. I want to be able to use a Linux machine and be able to communicate with another user using a Mac, Windows, Linux or whatever they opt for. Ekiga, allows linux users to make voip calls to MSN users. This is a victory in itself considering the fuss and excuses windows user make about the switch to linux.

If there was a standardize protocol, then both open source and proprietary source software manufactures could make their product to work with clients of different platforms. I believe this is what the article indirectly was pointing out.

I have used Skype on windows and it's excellent. It provides encrypting of commucation etc. But what is the use when users don't have a real choice of using whatever protocol they desire?

Cisco is in the process of making some of their product compatible with SIP, which is great. Why can't Skyp do the same? Think about he future, will ya?

Re: "Skype on Linux is unstable..."

Sare Okirom's picture

I've been using Skype on Linux (Mandrake 10.1) for over a year now and never had a single problem. My daughter has used Skype on Mandrake 10.0 and 10.2 for more then one and a half year and never experienced a problem either.