Taken from
mumble and murmur are open source alternatives to Teamspeak and Ventrilo. mumble is the client and murmur is the server. The interface is simple and intuitive, user management is handled in the client as of the latest svn release, and the sound quality is fantastic. The major kicker for this one is its efficient use of bandwidth. mumble uses significantly less RAM and bandwidth than either Teamspeak or Ventrilo. It is built on QT4 so you'll need to get those extra libraries. This program could quite conceivably dethrone the big boys in their gaming VoIP dominance. mumble is available for both Linux and Windows so no matter which OS you game on, you can communicate with the rest of your friends.
Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming.
Mumble's superior audio quality comes in large parts from Speex, a high quality voice codec and DSP library for echo cancellation and denoising.
last modified 23 October 2007
Newsgroups: gmane.linux.debian.user Subject: Re: FLOSS VOIP Client Software Date: Fri, 25 Jan 2008
> I am trying to look for a decent FLOSS VOIP software that allows me to > communicate to Windows/Mac users. Skype is out of the question since it > does not even support Debian PPC (which I have). > > I have heard of Ekiga. How is that?
I too want to find a definite solution for this. I can just let you know my experiences with some software I tried to talk with my girlfriend, who uses Vista:
Ekiga. It certainly looks a nice option, but last time we tried (just over 3 months ago) the windows version wasn't very solid and we had to drop it. More over, it has loads of Gnome dependencies, which may, or may not, be a problem for you.
OpenWengo. Nice but trickier (just slightier) to get up and running, if I remember correctly, and the sound quality WE got wasn't as good as, say, Skype.
Kopete. Running this on my Debian machine it was straithforward to talk (and video conference too) with her using MSN. For this it uses some reverse engineered library so it still is FLOSS. The same about the libraries applies here, but for KDE. Our choice so far.
When we tried this she was abroad using a hall Uni network, and due to a lack of planning I couldn't put my hands on her computer before she was already there, so we were constrained to try the easiest solutions we could think of. With more time and full control of you and your friends' networks I guess you can obtain better results and even try other ones, like Linphone.
> Does that work with Windows/Mac users using Skype clients?
You can't talk with Skype users using anything but Skype. Kopete has an addon to do this, but it's just a workaround that launches a Skype instance and puts Kopete on top of it; I haven't even bothered:
A good while ago somebody claimed to have reverse engineered Skype protocol, but I heard nothing afterwards. If you find a good solution do let me know!
JoseC Rodriguez @gmail.com
> I have heard of Ekiga. How is that? Does that work with Windows/Mac users > using Skype clients?
There are several in Debian. Butthe two I would recommend are ekiga and twinkle.
Tzafrir Cohen @cohens.org.il
The easiest option assuming you have some control over your network and can open some ports is to use Ekiga at your end and NetMeeting at the other, on most windows PCs you can start NetMeeting by "start" "run" and then type "conf" you'll be using the old VOIP protocol H323 and not SIP but I used it for years and it's bloody good, only you will be able to receive the calls and you'll want to set up dyndns or similar so the other end has a consistent name to dial.
Failing that the only windows SIP clients that support H.261 [0] are Ekiga and Windows Messenger 5.1 (Messenger is not MSN) config tutorial here http://wiki.ekiga.org/index.php/Connecting_Windows_Messenger_5.1_to_ekiga.net Also see http://wiki.ekiga.org/index.php/Which_programs_work_with_Ekiga_%3F#Windows
Good luck, I'm really looking forward to Ekiga 3
[0] at present Ekiga 2.0.11 (current) doesn't support the newer H.263 video compression codec that most modern SIP clients use, however Ekiga 3 will http://wiki.ekiga.org/index.php/Roadmap_to_3.00
Bob @homeurl.co.uk
You don't need the same client on Windows and Linux. I would use SIP phones since I have found them the most reliable and the most widespread. That will change as IAX becomes the norm (I think).
There are two ways that I communicate with my Windows/Mac friends. Either is OK.
1) Both of you establish and account on Free World Dialup: http://www.freeworlddialup.com/
On Linux, use Ekiga. Ekiga will connect to multiple accounts, so you can have your Ekiga account and your FWD account active at the same time.
On Windows/Mac, use the Sjphone: http://www.sjlabs.com/ It is a very good sip-phone. The one for Linux is good too, but it is not free software and it does not connect simultaneously to multiple accounts.
2) Both of you establish an account on Gizmo:http://www.gizmoproject.com/
The Gizmo client for Windows is excellent. You still can use Ekiga on your Linux box, and I recommend that you do so. I speak to my daughter frequently with this Gizmo-Ekiga combination and the sound is excellent.
It is not necessary to use the Gizmo client to connect to the Gizmo network. Your friends can use the SJphone if they prefer. You can use Ekiga.
There are any number of SIP based systems out there that you can subscribe to. The two I mention are just ones that I know well.
It is not strictly necessary for both of you to be on the same "network", but it is convenient. Most networks have "peering" arrangements that allow you to connect easily to other networks. See also SipBroker: http://www.sipbroker.com/sipbroker/action/login
Alan Typhoon @aanet.com.au
> On Linux, use Ekiga. Ekiga will connect to multiple accounts, so you > can have your Ekiga account and your FWD account active at the same > time.
There are plenty of SIP service providers out there. If you want to safely experiment, use only those that don't require your credit card to register for starters. You won't be able to send calls out to normal phones, but you'll get the feeling.
> 2) Both of you establish an account on > Gizmo:http://www.gizmoproject.com/[]
They are also a nice SIP provider as-is: http://sipphone.com/ . e.g: to get free 1-800 calls to the US.
Tzafrir Cohen @cohens.org.il
> There are plenty of SIP service providers out there. If you want to > safely experiment, use only those that don't require your credit card . . .
Agree entirely - both FWD and Gizmo are free as in beer. Gizmo does have a "Call out" service that lets you connect to the regular telephone network. That part, of course, costs money.
<SNIP>
> They are also a nice SIP provider as-is: http://sipphone.com/[] . e.g: > to get free 1-800 calls to the US.
sipphone.com and Gizmo are the same bunch. They seem to encourage connection through Gizmo these days for some reason. In any case, the Gizmo client is good for Windows and Mac users. There is also a Linux one, but I have always preferred to connect through Ekiga.
Alan Typhoon @aanet.com.au
Newsgroups: comp.os.linux.misc > > Is it possible to set up a microphone/speakers on linux so that one can
Check xspeakfree (search freshmeat.net) for software.
Michael Heiming
Home Page: http://www.spearce.org/projects/xspeakfree/
Description: xspeakfree is a quick TCL/TK GUI front-end for the Unix Speak Freely package, giving Unix users a clean, easy-to-use interface to the command line tools.
GnomeMeeting allows Linux and FreeBSD users to video conference with industry standard H.323 applications such as the Microsoft NetMeeting program for Windows. The program has proved extremely popular with users, and is now included in many Linux distributions as well as with FreeBSD.
GnomeMeeting uses the GNOME desktop or the popular KDE environment. Audio is supported via standard sound cards and video can be used from capture cards or USB cameras. ILS and LDAP directories can be used, and H.323 protocol support is provided by the OpenH323 library written by Equivalence Pty Ltd (http://www.openh323.org).
No, because Netmeeting asks to the ILS directory to list people using Netmeeting and not other software like GnomeMeeting, but you will be able to see those people and contact them. ils.seconix.com accepts to list all people, so if you want your family using Netmeeting to be able to contact you, ask them to register on ils.seconix.com.
It takes more bandwidth because Netmeeting has better codecs than GnomeMeeting. OpenH323 and/or GnomeMeeting couldn't pay for those codecs.
The first thing to check is the "General History" (Tools->General History).
If you are transmitting sound, you should see that GnomeMeeting starts 2 channels, one for transmission, and one for reception. If no channels are opened for audio transmission and reception, it means that you have no common codec with the remote Endpoint. If that remote Endpoint is using Netmeeting, it could be a nice idea to make install the MS-GSM codec to your friend. (See the download section of www.gnomemeeting.org).
Is the Windows XP computer connected using dial-up? If so, please check if the personal firewall is enabled. It will block the communication and make GnomeMeeting to Netmeeting connections impossible. Disable it and it should work.
GnomeMeeting 0.94.1 is available in the downloads section. It is higly recommended to upgrade your OpenH323 (don't forget to remove the old versions before starting to compile). Here is a list of the new features:
NAT support has been improved. It is now enough to only forward ports to the internal host, you don't need anymore to patch the kernel, or install a gatekeeper or RSIP. The new technique is called IP Translation. You can even have several GnomeMeeting running behind a NAT/PAT gateway and listening on different ports, the callto URL is not dependant of the ports, for example, callto://ils.seconix.com/dsandras@seconix.com will always work whatever the port GnomeMeeting is listening on.
ILS support has been improved and extended into XDAP support, you can now define XML schemas defining how to register on servers, and thus also support LDAP servers.
ILS now displays a status icon, that way you will know if the person who is displayed is available, or not (in a call/in do not disturb mode). (That only works for people running 0.94)
Improved ILS browser, you can now apply search filters.
A new statistics window has been added, with a graphic displaying the speed of the different data streams.
A new call history window has been added.
Support for the BSD licensed Speex audio codec (better than GSM, less bandwidth) has been added.
The test image is now a moving GnomeMeeting logo. You can choose to transmit your camera image, the moving GnomeMeeting logo, or any picture in .jpg or .png file.
You have now a new video mode that has been added during calls: the possibility to see both remote and local video in resizeable separate windows. The size of those windows is remembered between sessions, the video mode that has been chosen is remembered too.
It is now possible to do a bilinear interpolation on displayed pictures to improve quality (quality is very good now, even when you are zooming).
There is a new status icon in the panel even for Gnome 2 users (patch to make it work under KDE will be ready for KDE 3.1)
New icons from Jimmac!
Improved Netmeeting compatibility.
Improved disconnection messages.
RedHat 8.0: Some people have reported that installing ALSA RPMs from freshrpms.net gave machine crashes while using GnomeMeeting. This bug has been investigated by some users, and their conclusion was that ALSA 0.9.0RC3 didn't give any problem. That could be a conflict between some RedHat kernel patches and ALSA 0.9.0RC5. Other users didn't complain (yet).
poldek> install gnomemeeting Processing dependencies... gnomemeeting-0.93.1-4 marks openh323-1.9.3-4 (cap openh323 >= 1.9.3-1) openh323-1.9.3-4 marks pwlib-1.3.3-5 (cap pwlib >= 1.3.3) gnomemeeting-0.93.1-4 marks SDL-1.2.4-5 (cap libSDL-1.2.so.0) There are 4 packages to install (3 marked by dependencies): I gnomemeeting-0.93.1-4 D SDL-1.2.4-5, openh323-1.9.3-4, pwlib-1.3.3-5 Executing rpm --upgrade -vh --root / --noorder... warning: /mnt/rpms-org//SDL-1.2.4-5.i386.rpm: V3 DSA signature: NOKEY, key ID db42a60e Preparing... ########################################### [100%] 1:SDL ########################################### [ 25%] 2:pwlib ########################################### [ 50%] 3:openh323 ########################################### [ 75%] 4:gnomemeeting ########################################### [100%]
Skype will enable you to make free calls anywhere in the world in minutes to fellow Skype users running Linux or Windows. Skype will also let you place landline calls from your computer to just about anywhere in the world for only a few penny's a minute.
Skype was created by the people who brought you KaZaA and uses P2P technology to connect you with other Skype users. If you are tired of paying for telephony (VOIP), Skype is for you!
Skype is quick and easy to install. Just download, register, plug in a PC headset to your souncard and call your friends. Skype calls have excellent sound quality and are highly secure with end-to-end encryption. Best of all, Skype does not require you to reconfigure your firewall or router — it just works and no spyware.
Newsgroups: gmane.linux.debian.user Date: Sat, 5 Feb 2005 14:04:45 +0000
> I would like to set up a system to communicate via voice from my Debian Linux > PC on my local lan - via a NAT D-link 604 broadband router to the net, to my > daughter running Windows XP - behind a NAT wireless router (Linksys WRT54G) > - which she shares with some friends at University.
Using a microphone and headphones this is can be done with SIP software for Linux (e.g. linphone, kphone, sjphone, LIPZ4) and Windows (e.g. x-lite, sjphone, firefly). The major problem is usually NAT but by opening ports for the SIP messages and the audio (RTP - Real Time Protocol) stream on the routers and/or employing an outbound proxy or a STUN server there is no reason why successful connections should not be set up.
Public STUN servers are easily available whereas you generally have to signup with a provider to use an outbound proxy. There are major differences between the two methods but essentially both are intended to overcome the problems of traversing firewalls and NAT.
> It would also be nice, but not as essential, to also add my wife's Win98SE > computer (also on the same LAN as my PC) into this communication system, with > one on one, or conference calling.
Equip your wife's machine with some suitable Window's client and have it operate on a different SIP port from the client on your machine. Forward this port on the router to her machine. Whether conferencing is possible is dependent on the clients.
> My local D-Link router has uPnP capability, and I can freely adjust which > ports are forwarded to which IP address inside my LAN. I "think" (but I am > not 100% sure) that I get my daughter to adjust the routing of her NAT router > if necessary.
If your daughter has difficulty with configuring her router or does not have access to it then NAT traversal can be accomplished by her using an outbound proxy.
> I have been trying to read how to do this, but everything seems to assume > that you already know all about what they are talking about. Could > someone who does understand it, give me some recommendations about what I > should read about.
These are by no means exhaustive but hopefully will be informative. The last two provide UK based SIP services.
> I think that I probably need some form of SIP server (to enable both my sofe > and myself to have independent addresses) which I can put on my debian server > on the LAN side of my router. Debian seems to have two possibilities > (siproxyd and asterisk) for this, but I can't find anywhere that makes this > 100% clear, or tells me which one I really should use. (The biggest problem > seems to be NAT traversal)
No SIP server is needed and neither do you want siproxd or asterisk at this stage. Basically you use someone else's server. Contrast using an ISP's smarthost with sending mail directly, which requires you to run a mail server.
Note that siproxd and asterisk perform different functions. siproxd is used to proxy requests from behind NAT to the internet whereas asterisk is a PBX.
> In terms of a client, I was thinking of using KPhone (since I already run > KDE) on my Linux PC, and am looking around for something for the Windows > machines. Xten seems a possibility - but then I got confused since it > seems to only have a number dial interface and I couldn't see how that > fitted with the servers.
kphone should be fine. I've used it without KDE. I don't have Windows and x-lite from Xten has no Linux version so I cannot comment on the cause of your confusion. However, the softphones I have used have all had a way of entering a SIP URL as username@domain in addition to a numerical keypad.
Something intended to get you going:
Go to voiptalk.org.
Read the voiptalk documentation on setting up a SIP phone.
Obtain an account.
Configure kphone with your username (User Part) and password.
Use voiptalk.org for the domain (Host Part).
Use voiptalk's outbound proxy without any forwarding of ports.
Test by dialing 901, 902, 903 and 904.
Test by phoning me if you wish.
ABrady has also given a very nice description of Skype and no doubt it will enable you do what you want. However, on a list such as this I feel it is fair to point out that Skype uses a proprietry protocol and although it is probably possible for it to interwork with SIP based software its developers do not appear to have any intention of allowing this.
Brian Potkin
> This isn't an answer because I don't know. But a suggestion/question > that might solve (nearly) everything. > > Have you tried or considered Skype? It does all of what you say you're > trying to do. It runs on linux, Winders and Mac. It's free, though there > are paid options that allow you to dial outside the Skype network while > still retaining the free part for everybody connected to Skype. It does > conferencing, IM, file sharing (I've heard, not tried), connects to any > machine with a sound card and microphone, it supports some internet > phone equipment (but not SIP, I believe), it is portable in that you can > log on from anywhere in the world and use it, you can block or allow > specific users, you can search users, you can change available status > the way you can with IM, you can have multiple users running from the > same location (on different machines), you can have multiple identities > for each machine, etc. > > I'm not trying to sell it, since it isn't for sale. It just seems like > the easiest way to go to get what say you want.
Regarding Skype, I downloaded the latest linux version (1.0.0.1) 2 days ago. It was easy to set up and I can make and receive calls, BUT… there's a problem for linux users at least for now. Skype has a bell (or you can use your own ring sound file) that alerts you when there's an incoming call. This is useful if you're not planted in front of the monitor all the time. Well, it's not implemented for the linux version. I went to the Skype forum and queried why the Ringing option on the options > hand/headset is greyed out. I was told to wait for some later version when they get around to implementing this feature. For me, a phone that doesn't ring is pretty useless. If anybody has managed to get Skype to ring on Debian, I'd like to hear about.
Jonathan Kaye
Last release: 2000-07-23. This release is the first one after a major rewrite of the code initiated by timecop. The end result of 3 weeks of our work is much more stable efone with a lot of new functionality added.
Dead. Last News on March 20 2000.
Gphone is definitely a work in progress and you probably shouldn't bet your business on it; if it breaks you can keep both pieces. Don't be too hard on the program, though.
High Quality, low bandwidth, real-time full framerate video conferencing software using a radical new algorithm from Cornell University.
Latest File Releases CU30 Library, October 22, 2001
It describes how to configure Linux for interoperation with Microsoft NetMeeting.
H.323 is an ITU-T recommendation for multimedia conferencing over packet-based networks such as LANs and the Internet. H.323 is broad and comprehensive in its scope yet flexible and practical in its applicability. This report describes the components, protocols, and procedures in H.323;
The OpenH323 project aims to create a full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
Tuesday, September 3, 2002 - After two years of active development, the GNU project has released GNU Bayonne, Version 1.0. GNU Bayonne, licensed freely under the GNU General Public License (GPL), allows small businesses, large enterprises, and commercial telephone carriers to create, deploy, and manage their own web-integrated telephony voice response solutions. By making use of existing computer telephony commercially available for GNU/Linux, Bayonne can support all capacities — from a single analog phone circuit to multiple PRI spans.
As part of GNU Enterprise (GNUe), Bayonne 1.0 joins the ranks of a host of enterprise-level software for the popular GNU/Linux operating system. "In this 1.0 release, we have established a Free Software platform for the delivery of quality telephony services everywhere".
documented on: 2002.11.11 Mon
While most Open Source projects are applications and utilities intended for single users, David Bryan and David Kelly did something different. They created an infrastructure project — a VoIP phone system that either can run on a single box attached to a couple of IP phones or can scale up to a network of hosts processing hundreds of calls between thousands of users. In this informative technical article at ELJonline, Bryan and Kelly detail the 'Vovida Open Communications Applications Library' ('VOCAL') project, a fully functional phone system that can run on either Red Hat Linux or Sun Solaris.
FreeTel provides real-time full-duplex voice communication via the Internet. You can talk to friends and relatives around the world, free of long-distance telephone charges.
FreeTel includes full-duplex (including Sound Blaster 16) audio support, a real-time Electronic Phone Directory, Advanced Caller ID, Superior Audio Quality, and a keyboard communicator. Best of all, you can download this high-quality, fully-operational, unlimited use product for free .
Unlike other "free demo" Internet Phones, FreeTel does not time-out after a few minutes of operation. This free version is fully-operational because it is advertiser supported.
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documented on: 1999.09.29
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Businesses were abuzz about voice over Internet Protocol technology in 2003, announcing new deployments almost daily, but the reality is that the actual work is only just beginning.
Analysts say it could be years before most companies are willing to throw out their old PBX (private branch exchange) phone systems and run end-to-end Internet telephony. And that could mean a slower road to the IP promised-land that Cisco Systems and other IP supporters have been preaching.
"We aren't seeing large enterprises ripping out and replacing old systems with new ones," said Tom Valovic, program director for IP telephony at market researcher IDC.
Most companies are taking a staged approach to their VoIP deployments. Companies using IP-based PBX gear from Cisco Systems or 3Com are often deploying VoIP in limited locations, such as remote or branch offices. Others using an IP-enabled PBX from Alcatel, Avaya, Nortel Networks or Siemens are deploying IP telephones alongside existing digital phones. And some companies are doing a little of both—deploying all Net telephony solutions in some locations, while slowly migrating other offices.
"About 90 percent of the deployments out there with IP-enabled PBXs are using a mix of IP and TDM (time division multiplexing)," said Brian Riggs, an analyst at Current Analysis.
TDM is also known as circuit-switched technology because it creates an end-to-end connection to complete a call. IP, on the other hand, uses packet switching, a method that involves breaking a call into pieces for more efficient delivery and then reassembling it at the other end.
Depending on the size of the company and the state of its existing IP infrastructure, rolling out a VoIP network can take years to complete. For example, Grant Thornton, a Chicago-based accounting and consulting firm servicing midsize companies, is already three years into a VoIP deployment that began in 2001.
The company has 51 offices throughout the country and employs roughly 3,500 workers installed a hybrid PBX solution from Avaya, which allowed it to maintain portions of its switched telephone network, so that some employees could continue using their existing digital analog phones while others used the IP phones.
"Once you begin a deployment like this, you realize how much there is to it," said Kevin Lopez, national manager of telecommunications for Grant Thornton. "When we first implemented the five-digit dialing system, we saw how many IP addresses we had to change. We realized how much work it was going to be to rollout the entire network, so we decided to do it over a longer period of time."
Since then the company has slowly deployed other services, such as unified messaging, and it is about to implement "soft" phone technology that will allow employees to make phone calls from their laptops. A few of the company's locations are all IP. But most locations still use a mix of IP and the traditional switched telephone network. In those locations, the VoIP network is only about 7 percent saturated.
The larger the company, often the longer it takes to deploy the service. Not only is the basic installation of the IP phones time-consuming and labor-intensive, but the IP technology used today also requires more management and maintenance than traditional digital phones.
"You never have to update a regular digital phone," Lopez said. "You just plug it in and it works. But the IP phones need software updates. You need to be sure you have the staff and a user base that understands when those updates need to happen."
Larger networks also require more testing to make sure the existing IP infrastructure can handle the additional load from the IP voice services. Some companies may even have to deploy additional infrastructure.
For example, Jenny Craig, the nationwide weight-loss company, has spent the past year and a half IP-enabling its network. It started with basic connectivity linking its 430 remote locations in North America and an additional 130 offices in Australia and New Zealand together via encrypted IP tunnels. The company is now just starting to roll out IP telephony to its branch offices.
"I'd say we are still dabbling in the technology," said Jeff Nelson, director of technology for Jenny Craig. "Going forward, everything will run over IP. But we are taking our time getting there."
Slower rollouts could mean that companies won't be spending as much on VoIP gear over the short term as some have predicted, Infonetics' Riggs said.
"With a staged environment, companies can move to the next phase of deployment whenever they're comfortable," he said. "That could be next year, or it could be in 2010."
But Cisco doesn't seem worried about revenue delays. The company boasts that it sold its 2 millionth IP phone in June. It also said that it's sold an additional 400,000 IP phones in the third quarter of this year, proving that momentum is building.
Rick Moran, vice president of product and technology marketing for IP communications for Cisco, admitted that some companies may take longer to deploy VoIP end to end than others. Either way, the movement toward VoIP means more business for Cisco—at some point.
"Some companies will be faster than others," he said. "It's interesting how companies are starting to look at VoIP like it's a business application rather than a telephony service. As a result, they tend to try it out in certain remote offices or in a particular department before they roll it out to the entire company."
A technical review conducted by the British government has found several security flaws in products that use VoIP and text messaging, including those from Microsoft and Cisco Systems.
The flaws affect software and hardware that support the real-time multimedia communications and processing standard, known as the International Telecommunications Union (ITU) H.323 standard.
The security problems can cause a product that supports H.323 to crash. For example, in Cisco telecommunications products running its IOS operating system, the vulnerability could be used to cause the devices to freeze or reboot. However, on Microsoft's Internet Security and Acceleration Server 2000, which is included with Small Business Server 2000 and 2003 editions, the vulnerability could allow an attacker to take control of the system.