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Cross-Platform VoIP Software?
feilkin writes "With the release of Skype's Linux client, I'm wondering about alternatives. Namely, cross-platform solutions for voice communication. I've got friends who are using Windows, Linux and OSX, and I'm hoping that there is a way to communicate with all of them. I myself am using Linux, and I haven't been able to find any solutions that seem fitting to my situation completely. Does anyone have a solution that'll be useful on all three platforms, or solutions that may be coming in the near future?"
http://www.sipforum.org/
I have no problem with your religion until you decide it's reason to deprive others of the truth.
http://www.skype.com
------- ?
fwd.pulver.com it has clients on all platforms. the other one is called asterisk
http://www.freeworldialup.com/
I've got friends on it using windows and linux (I personally use both, and have clients installed on both). I'm pretty sure they've got osX clients aswell.
I know nikotel works great on windows and macOS and as it is SIP compliant it works with linphone and kphone on linux.
download and burn linux with one click on windows
OpenH323 is available on all 3 platforms and has very good voice quality. It can do video as well. Setup is not always trivial: it needs lots of open ports, udp and tcp. The license is MPL.
I haven't checked in on the project in a while, but Bayonne was coming along nicely in this area and is currently used in a few production facilities.
You might have to roll your own, but the framework is certainly there.
Want to improve your Karma? Instead of "Post Anonymously", try the "Post Humously" option.
http://xten.com/
At this point, all the tools needed to create an Open Source cross-platform VoIP system are easily available. The Speex codec is specifically designed for low-bit-rate voice, is BSD licensed, and is implemented in both C and Java. It would not be hard to take this codec, throw in some good sound libraries and some crypto libraries (OpenSSL perhaps) and roll up a VoIP client. In fact there is a Speex implementation for Java, so you could write one in Java, and yes, Java really is "write once run anywhere" these days. Someday when I have more time I might do this. As a Java applet it would be great because there would be nothing to install.
Does anyone have a solution that'll be useful on all three platforms?
Son, I have a solution! Pick up the phone and call your friends!
... just install windows everywhere
I have found that two tin cans connected with the best string money can buy is very cross-platform. This solution has no problems at all running on Linux or even BSD (despite the fact that OS is dying!)
You're looking for a standard protocol that can be used across all platforms, and that protocol is SIP. I've used several VOIP products that have SIP support and currently am using a Grandstream Budget Tone 100 VOIP phone ($65) to do my calling and can contact anyone on any platform that can support SIP.
I've just tested the Linux version of Skype here -Local box to box - Linux to windows - 2 accounts etc..works as advertised. But just from a technical point of view being an old coder myself, i'd like to know how they minimize the lag.. Dam this thing works better than my cell from a lag-latency point of view.
*--- Sometimes a majority only means that all the fools are on the same side. ---*
There are two main standards in use for VOIP:
SIP and h.323. There are lots of clients out there for both of them.
There should be a checkbox next to the "ask slashdot" submission box that says "did you use Google first?"
UT2K4 has built in VoIP support and is available for Windows, Linux (32 and 64 bit), and Mac! "No, sir, we're not wasting company time, we're _collaborating_! We're enhancing shareholder value! It's a whole new paradigm!"
What a coincidence- Skype releases on Linux on June 21, and someone posts a "question" about alternative VOIP- but the "question" is a thinly disguised publicity announcement of Skype on Linux. Is feilkin somehow associated with Skype?
SIP is a VoIP standard used by a lot of company doing VoIP comercial services like vonage or cisco.
For 29.99 USD a month, Vonage is the way to go.
No fscking around with codecs.
No gcc bullshit.
No patching, only to have an OS upgrade break your app.
Pay for it, be done with it, move on. What's your life worth? Time = money.
I want to delete my account but Slashdot doesn't allow it.
Sure, you can't talk to people who only have POTS.
Sure, its designed to be used with online games.
But really, its just IRC with voice. I talk to people in Australia with it all the time, for free. You can turn the bitrate down and even talk to people with dial-up. I usually just use gaim to tell someone "hey, join the teamspeak server, let's talk".
Also, this way I don't have to remember phone numbers.
The GeekNights podcast is going strong. Listen!
While reasearching this to play online game with buddies, I found that ventrilo and teamspeak were the most popular. Ventrilo has clients for many platforms (Win32, OSX, Linux, BSD, Solaris) but only a client for Win32. OSX and Linux clients are in development. Teamspeak seems to have only Win32 and Linux client and servers.
My friends and I couldn't get Teamspeak to work, while ventrilo worked straight from install (on Win32 client & server or linux client).
To bad neither are Free Software/Open Source since both are distributed as binaries.
http://ventrilo.com
Has Win32, Mac and Linux clients.
It is client/server, so you'll need a server, but you can get 8 users (I think) on the regular server. It is relatively bandwidth-friendly and awesome quality.
Probably a bit harder for computer illiterates to use but its very cool software.
VoIP is built around several "standards", mainly SIP and MGCP, both of which are network protocols.
... so you could use
:-
e r. htm
:)
Windows Clients
- NetMeeting
- Windows Messenger
Linux Clients
- Gnome Meeting
Mac Clients ?
To get them all talking to eachother in a group conference I recommend the battle tested CTM Conference Server (Windows or Linux)
http://www.fvc.com/eng/products/conference_serv
It's not free, but you didn't ask for free
No kidding. A friend and I played around with it in the same room.
The lag-latency was null and void.
I was using a 1 ear headset and between it and him it was almost stereo.
What you say? I did not read your post.
as many others pointed out, natural joices would be h323 (very wide-spread) and sip.
I don't know much about sip, but everyone tells me "stop using h323, use sip". Seems to be better, but never change a running system.
h323 is only for VoIP, not for calling real phones - unless you have an gateway to the "real" world.
There are many h323 programs available, like netmeeting (really hardcore connectivity problems through firewalls, better use...), openphone (openh323/windows), gnomemeeting(openh323/linux) and so forth. Normally, all h323-compatible apps should be able to communicate. You can use many different audio codecs, depending on your bandwith and data rate quality. There's even the (in)famoues GSM codec that's used in european cellphones, sounds quite good for 1.6k/s+overhead.
Why would you think there'd be a huge lag working local-local connections?
The problem with most voip is with network induced latencies. So a local-local test is less interesting as say a system - nearby network test.
It works across all platforms, including all future OSes, and it is cheap to buy and operate.
If you want mobile, get a cell phone.
For a complete VoIP Linux solution, check out Asterisk.
"Beware of he who would deny you access to information, for in his heart, he dreams himself your master."
H323 is a huge ball of protocols (H245/Q931/etc) for communications technologies. See the OpenH323 project for more. You can use it for connecting, say, VoIP and a regular phone using a Cisco PSTN gateway. The two main reasons for using something like the H323 protocol set are for session initiation (can you do video? What audio codecs do you support?) and data format (G721/722/726/etc).
NetMeeting and GnomeMeeting both support some subset of H323 that lets them talk. So you can do audio/video between platforms that way.
There's another protocol called "SIP," for Session Initiation Protocol. H323 is extremely complex (until OpenH323 you needed to pay $$$ to license an ASN1 compiler to compile the protocol into headers so you could code to the spec) and a pain to tunnel over HTTP. SIP was created by some folks who recognized the weaknesses in H323 and decided to create their own protocol.
Bottom line, you can use GnomeMeeting to talk to NetMeeting. I have no idea about to OS X world.
All of these will interoperate. They get tricky when used behind NAT. The best option I have found in that case is to use a gatekeeper.
Gnomemeeting for Linux
OphoneX for OS X
Netmeeting for Windows
. Ergo sum cogito - Yoda
Yes, that was the original question. Read all the posts to get more informed (and this is needed: if you decided quickly without information, you'll decide wrong).
Now, for an appetizer, search for GnomeMeeting on Google.
www.asterisk.org
This thing is a VoIP BEAST. It's an open source PBX which runs on Linux. This will solve your problems by connecting all of these incompatible VoIP clients, making them all seem like virtual telephones, each with their own extensions. (This is good, if you don't mind them using your bandwidth when they bounce off of your Asterisk server to communicate with each-other.)
"PBX" seems scary -- it's the same kind of system large businesses use to manage tons of phone lines, both inside their company and connecting to the outside world.
For the needs of people like you and I, don't think of it in terms of "a solution used by people with lots of phones" -- think of it in terms of the kinds of technology it uses and can connect with.
"Physical layer" stuff: with dedicated hardware it can talk to existing phones and existing phone lines. There's even a PCI card that can communicate with four T1 lines, for nearly 100 phone lines out to the telephone company. It can also do VoIP using standard interfaces like SIP, using its own unique (but open-source, not proprietary at all) interface called IAX, with existing programs like Netmeeting or MS Messenger, or with any number of Linux programs. (There's even an IAX client for my Zaurus PDA. That's not all that practical for receiving calls, but I have successfully placed phone calls with that client, over 802.11b.)
Logical stuff: each of these connections to the outside world is given a context, and you can do things with those contexts. A connection to your outside phone line will be used by unknown callers, so its context shouldn't have access to features that cost money. A connection to an inside phone is "trusted", so it should be given access to these features.
The system has something like a "dialplan", which is a rather flexible set of scripts you use to handle calls. There's a lot of room for creativity here -- you can make your system do anything you want with any call.
This is so flexible because you form your dialplan from a bunch of references to "applications", either built-in or external. Some are very simple: play this wav file, transfer to this extension, go to this voicemail box, etc; some are more complex, such as "shell out to this executable CGI-style and do whatever that executable tells you".
Asterisk also comes with a bunch of audio samples recorded by a "professional PBX voice", and many of them are saying some rather funny things, only useful for a home user. "All representatives of the household are currently assisting other telemarketers. Please hold, and you call will be answered in the order it was received."
Asterisk can email you your voicemail messages as wav files. This is a KILLER feature. But you weren't asking about voicemail, you were asking about VoIP.
Pros: VoIP BEAST. Take all your friends with VoIP clients, give them signins and extensions and voicemail, give them conference capabilities, etc. (Then they all use your bandwidth.)
Cons: Complexity. Even if all you want is a simple call routing tool to make incompatible VoIP systems talk to each other, you have to learn the entire system to make it work. This is a typical Linux problem: you have to read tons of documentation / visit forums / discuss with others to figure it out, but because it uses "real world" concepts and is designed intelligently, once you're finished you have spent 30% of your time learning the quirks of a single software package you could care less about, and 70% of your time learning about how the subject works, gaining knowledge about that field that will follow you to any other program.
(That's definitely true here: Since playing with Asterisk I've talked with professional telecom guys, and found what few terms and concepts I've learned from Asterisk definitely overlap with their "real world" stuff.)
Weird system service requirements. Some software features rely on a very high-resolution system timer, and (allegedly) can't get t
Its this really cool new protocol that lets you communicate and its supported by all major platforms. So far Microsoft has not embraced and perverted it so its cross platform future looks good. It great because it usually gets delivered right away but your friends don't expect a response within thirty seconds. You can send to an two way pager or a cell phone. You can even claim that you haven't been checking your email for a while if you don't feel like answering right now. And you can send HTML email that looks just like a web page including embedding graphics and sounds. You can even "attach" files to the message. Is that freaky or what? I think I just blew my own mind.
Liberals call everyone Nazis yet they are the closest thing to it.
Open your mixer app, e.g. aumix or kmix, and check the inputs. Chances are the mic inputs are muted.
Shameless self-promotion - check out the shtoom program. It's cross platform (although the Mac support is incomplete, it in theory works, thanks to portaudio[1]), it has user interfaces for command line, Tk, Gtk, Qt, and wxWindows. Audio support is via PortAudio and OSS. It handles most NATs correctly (using STUN).
C on2004/
It also includes 'doug', an application server for writing voice apps. There's a simple voicemail and simple conference server implemented in doug.
It's pretty rough - it's certainly not something you'd give to your mother to use, but hey, it's free software.
It's also entirely in Python.
At the moment, the best bet is to use the svn trunk.
URLs:
Software: http://shtoom.divmod.org/
PyCon paper (also possibly useful for an overview of VoIP): http://www.interlink.com.au/anthony/tech/talks/Py
[1] Native Mac support will be finished Soon, I have a mac being shipped to me.
Shtoom is a open-source, cross-platform VoIP softphone, implemented in Python. As well as the basic phone, the package also includes a number of other applications -
shtoom - the end-user phone
shtam - a simple answering machine/voicemail application
shmessage - an announcement server
Maybe?
GNAA in full effect, use your brain before modding.
VoIP requires H323 and other setuid scripts
Yes cos H323 being a protocol is really likely to be setuid.
The internet was simply never designed for realtime interaction
No because nobody ever thought humans might use it.
VoIP completely bypasses the government's anti-terrorist infrastructure
None of the two main VoIP protocol suites include any encryption so beep, 3 out of 3 wrong, but some dick gave you the informative mod so again here I am wasting my time putting crap right.
I think I just hit the point where I no longer want to visit, read or post to slashdot. When this kind of completely idiotic crap is getting modded up whats the point. I might aswell join GNAA and start trolling, at least then I'll get some fun out of it....
Shtoom is a 100% VoIP implementation in Python.
It rocks. Homepage of Shtoom, Shtoom is a open-source, cross-platform VoIP softphone, implemented in Python. As well as the basic phone, there are many other services. PyCon Paper on Shtoom.
BudgeTone 101 - $75
It is basically a phone with an ethernet port and SIP built in. Not bad.
Life is the leading cause of death in America.
The SIP RFC you linked to is obsoleted by RFC 3261
i am not one to toot any given distros horn but ...
mandrake10 installed all my drivers (the sound drivers were the ones that I had problems with in mdk9 AND appears to automount my keyfob USB flash memory "HD".
Kopete was(is always?) installed by default. and is a nice little program that manages and interfaces with all of the IM services I've heard of and a few i hadn't. If someone can configure or tell me(you) how to configure audio chat via Kopete i'm pretty sure there must be ONE of the IM programs that will run on MAC and MS
"He's a real midnight golfer"
re: Protocols and the rest of your first two bullet-points. I have no idea how this works / should work. So i don't have an opinion.. but your last point... I know im going to come of as a troll here, accepting this flamebait.. but i don't care... I am a reasonable and responsible adult and i will exorcise my right to express my views. I am thinking that you forget that the majority of people on this planet doesn't answer to your goverment or falls under its "anti-terror infrastructure" protection. You say 6-800 american people dead per year because the U.S. goverment doesn't have insight into VoIP? I then wonder how many swedish people it would protect if we let the U.S. monitor our phonelines. My guess is none. Why should "Homeland security" be alowed to listen in on my phonecalls you ask? There is no reason for it. Can you garantee me that this obvious power wouldn't be missused to listen in on the swedish calls? I am all for democracy and freedom for all men and women to live in peace, but I wouldn't want american goverment listening in on my phonecalls no matter what. I'm swedish. I don't answer to Gerorge W.Bush and his obviously hypocratic people. I don't want the swedish security agencys to listen either, but i realy don't think the U.S. should police the internet. Lets form an international board for that one! Then i might be ok with insight into my phonecalls. Peace /casa
don't fuck around with a stupid software phone.
just get a grandstream budgetel or a sipura or a wisip or any number of other SIP hardware phones.
You will be happy you did. I am.
plus wearing stupid headsets looks retarded.
--- ask me about nihilism, I will have nothing to tell you.
The lag is minimal even for long-distance connections. I live near the west coast of Canada, and my girlfriend lives in north Florida. (oh, how I love the Internet 9_9) Judging by the rhythm and flow of our conversation, I would guess that the worst it got on a regular basis was a lag of less than one xecond. (of course, there'd be some severe peak latencies now and then, lasting three seconds or so.)
I can understand how they reduced bandwidth-induced lag... their codec seems to be very well tuned for speech, degrades gracefully, and cuts out all sound below a certain threshhold... but once they reduce the number of bits, how on earth do they push those bits so damn fast across the continent?
Standing at the very edge of my imagination, I peered into the inky void and realised -- I couldn't think up a new sig.
I would add that even calls from russia to usa dont have any noticable lag (given both sides have good connection..)
The NAT problem is definitely NOT overrated for residential gateways. There is no SBC in this case.
b erg-midcom-turn-04.txt) addressed the problem, but in an inelegant manner, by routing all RTP through a public server.
m usic-ice-01.txt) describes a method of using STUN and/or TURN to discover, describe, and prioritize many potential addresses. Using ICE, two SIP clients can choose the best possible route for RTP, through several NATs that might separate them.
Sure, SIP can use TCP as a transport, so a client can punch a signaling connection out through a residential NAT/gateway, but a SIP client still needs to advertise an address and port to receive RTP on. He can't very well advertise his NATed address if he's connecting to a client outside of his NAT.
STUN - RFC 3489 - (http://www.ietf.org/rfc/rfc3489.txt) partially addressed this problem, by allowing an application to discover its public address and port mappings.
TURN (http://www1.ietf.org/internet-drafts/draft-rosen
ICE (http://www.ietf.org/internet-drafts/draft-ietf-m
And it's backwards compatible.
It should also be noted that ICE is independent of SIP, and could also be applied by H.323 clients, or RTSP streaming client/server for that matter.
Vovida.org is pretty comprehensive. Thier Vovida Open Communications Application Library (VOCAL) is pretty comprehensive, and works with many different vendor's phones, soft phones, and even Cisco's high-capacity PSTN gateways (H.323, MGCP, and SIP).
Go to Help -> About... there you can see the
licenses and also what codecs Skype is using.
Such as Global IP Sound and
Joltid
Just get the $15/mo. package. 500 anywhere minutes, and all calls to other Vonage subscribers are free.
I'm using Vonage, and I love it. The sound quality is not pindrop-clear, but it's good enough for general, everyday phone use. I also like that Vonage emails me all of my voicemail messages, so I know immediately who called my home while I'm at work.
Voice Over I. P.
I use Road Runner through Time-Warner Cable, and have no issues with port-blocking. Also, the Vonage VoIP-box goes outside of your router/firewall, so you won't compromise your own network, either.
I am a fan of teamspeak2 ... it it not open source but it is free, cross platform and works quite well for conference/gaming type communication.
I'll take a stab at how.. This is just a guess but instead of doing a transmit and recive, they might open a stream like one from a streaming video server but smaller to only use enough bandwidth neccesary for low-medium quality voice. durring the non talking time they could just introduce sub audible noise to keep the conection alive and recive the same from the other end. with a stream like this, it will also give them the ability to have a checksum on the other end to mesure link quality and adjust the streams bitrate as neccesary to acomidate for the conditions of the conections.
This stream could be somethign like the internet radio but because it is voice it might only need half the bandwidth and if they compress it a little they can push a lot of trafic with little bandwidth. If they cut out some of the highs in the transmission they can also cut out some of the tinny sounds asociated with low bitrate conections. also they might take a voice sample at the beguining of the call and use a checksum based on that to set a software EQ that would pull the distortion out but that seems like it would use alot of proccessing power
Okay, now I will take the flamebait hit. What are you, Swedish or a Moronian? When the parent post is THAT far off that it makes no sense, you should consider that it might be a joke. Maybe? Thank about it.
"I am a patient boy. I wait I wait I wait. My time is water down the drain..." Fugazi
For 0 dollars per month, Skype is the way to go, if your friends have computers.
We need an open source Skype. Sooner or later Skype will start charging. For what? Just some programming, that could be done better if it were Open Source.
Walking around with your head crunched over onto your shoulder to squeeze the phone there, getting an ache in your ear from flattening it, and having to hold a cellular/cordless phone close to your head for hours-long conference calls all look very retarded to me. Having a quality headset (like those from Plantronics) and being able to use two hands, walk upright and avoid ear/headaches looks very smart.
Of course, if you meant that wearing a stupid-looking headset looks retarded, you're absolutely right - just look at the enormous contraptions that Britney Spears, Garth Brooks, and all those other stage performers have to put up with. But regular users just wanting to have a conversation have a wide range of options, if they're willing to spend some money. Even Radio Shack has some decent choices.
Personally, I like the trend toward USB headsets with built-in sound chips. It makes more sense to put the audio D/A hardware in the actual output device than in the computer, these days - keep it digital for as long as possible, preventing noise and cable mess.
-Elentar
The wheel it turns, around and around, with an ancient rumbling sound.
KDX is a powerful "BBS"-style (Bulletin Board System) encrypted internet communications system that provides voice chat (Internet Telephone), text chat, messaging, news, file and folder transfer, remote access, trackers and more. It uses strong encryption to protect your communications for security and privacy. It is very useful for groups that need to collaborate on a project via the Internet. It is also very useful for remote administration of a computer. KDX uses a client/server architecture (NOT peer-to-peer).
The software is available for Mac OS 9/X, MS Windows NT/2000/XP. The Linux server is currently in beta, and the client is coming soon.
"I seem to have mastered a certain amount of control over physical reality."
By the way, of course you will accept the charges , ie call collect
It gets expensive !
no, performers wearing stupid headsets look lame, and people who listen to popular music like that should be shot.
--- ask me about nihilism, I will have nothing to tell you.
How long do you think it takes the average packet to travel from your machine to your girlfriend's? I'll wager that it's less than a tenth of a second, and that the round-trip time on a ping is probably less than 200ms... this is typical of any two points on the net in north america.
unless some links are either badly configured or saturated, or both, bandwidth isn't really an issue.
Half of streaming radio? Far, far less than that.
The VoIP codecs I've commonly seen use about 6kbps. That's nothing by today's standards.. acceptable quality for voice is far, far below radio.
You can make voip calls quite easily over a 28.8kbps modem dialup connection.
SO get a Sipura, which lets you plug in any phone you like.
there is no reason to involve the computer in this task.. there are a number of cheap broadband telephone adapters out there, and SIP is easy to use.
Saying "Keep it digital as long as possible" only makes sense if you aren't transcoding things with lossy codecs along the way in order to accomplish it.
On June 16, 2004, there was an internal demonstration at Skype of the alpha version of Skype for Mac.The alpha version worked well and the development team is working towards a beta launch of Skype for Mac in about 2-3 months. The other major OS that Skype doesn't support is Windows 98, and there aren't official plans for that as of yet.
- Allen Pike
Altering time, one time at a time.
``It handles most NATs correctly (using STUN).''
I always thought the most obvious solution to connecting to a NATed node is to have the NAT box act as a relay to the nodes it performs NAT for. One way to implement this (without running out of address/port pairs) is to use IP-IP tunneling. This is described, for example, in RFC 2003).
Nodes can then negotiate (or, with an extension to DNS, look up) the parameters to use to traverse all the NAT boxen in between them. This schema doesn't require traffic to go through a server it would not otherwise pass through. The cost is one additional IP header per anti-NAT traversal.
Please correct me if I got my facts wrong.
What about for those users who don't have the hardware to spare for another isolated dmz machine? Or did I missunderstand somthing in all those "GNU/Linux is more secure than Windows" posts in the "Lessons learned from Blaster" discussion....;-). I don't think Skype is the solution for those of us who were already too paranoid about the firewalling for h323...
Troll? This was a rather funny comment about that Windows Administrator who installed 9 linux distros (and did Gentoo two or three times as well) in 2 days and couldn't get his sound going.
9 Distros plus Gentoo twice in two days. I wan't whatever hardware he's got...
sigaar
SIP is a lot more than just a VoIP standard, but I will grant you that the primary use of SIP today is very probably VoIP connections.
SIP = Session Initiation Protocol. You use it to set up, control, and tear down sessions.
Before SIP, the primary session management protocol for VoIP was H.323, which came out of the videoconferencing space. For basic audio calls, H.323 is a pretty heavyweight protocol. Also, it doesn't take advantage of many common Internet protocols. Of course, that's because it preceded many of these protocols.
Currently, the number of H.323 based solutions exceed the number of SIP based solutions for VoIP, but the tide appears to be shifting.
There is also an extension to SIP called SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions). SIMPLE is somewhat of a competitor to the XMPP protocol that Jabber uses. SIMPLE has the advantage of being able to more easily leverage a SIP connection, but XMPP is much more mature and feature rich.
you're thinking of H323.
SIP has specific support for NAT. H323 does not.
My guess is they don't.
It's a schoolbook usage of UDP, but all VoIP works that way. You should have no more delay than on a ping.
http:/www.peerio.com
Nope. No linux ventrilo.
I have been using Oh-Phone http://xmeeting.sf.net/ on MACOSX it works well with OpenPhone http://openh323.org/ and Gnomemeeting. God bless standards
There's project called PPhone that supports win32 and unix platforms, comes in open source and offers textual communication in ICQ and IRC style as well. Is able to use multiple codecs, one connection can use different codecs (one for sending, one for receiving). During call initiation codecs are elected automatically each side offers set of codecs for sending and receiving. Client works well on i486-50MHz. Among codecs are: CELP, MELP, GSM-HR, GSM, .. Conference calls are possible.
Ever since my boss got friendly with the Asterisk developers, my company's internal telephone network is now almost entirely VOIP. We have a server running Asterisk, with a Zaptel line card (needed a 3V3, 6MHz, 32-bit expansion slot; something you apparently only find on high-end mobo's, as most of the low-cost ones are 33MHz and/or 5V) plugged into an E1 line giving 30 ISDN lines. But you only need this to connect to POTS phones -- connecting to other VOIP phones is just done over the internet. The Asterisk machine also currently runs our intranet, though I'm ordering a new server for all the non-telephonical functions as something keeps crashing (not often enough to be serious, but we need to narrow it down).
..... we use dedicated hardware telephones. The Grandstream BudgeTone 101 was the first we evaluated, be aware that this comes with a Continental-style mains adaptor so you may need to get a new power pack (regulated 5 volts 400mA DC + --o)-- - polarity). This works lovely as a SIP telephone but doesn't as standard allow for a headset, which we kind of need in a call centre. The handset does use a standard RJ01 connector, but there seemed no easy way to deal with the receiver switch. We also evaluated every softphone we could get our hands on. In general they seem to be a bitch to get to compile; I had the best result with Linphone, it wasn't as polished as KPhone but it seemed to crash less often; and got absolutely gnowhere with Gnophone. Bear in mind also that a telephone headset will reveal the limitations of the sound chipsets on modern mobo's: you will require a real SoundBlaster-compatiable if you want to be able to understand what anybody is saying. I am running Debian Sid, my boss is running some perversion of Mandrake with a load of stuff from Cooker, and all our workstations run Mandrake 9.2 (hackerish systems are fine for us hackers, but it's more important to have Stuff That Just Works for the masses). We also got a softphone client from Zultys, called LIPZ; which looked stunning but was problematic in practice. It seems to bogart memory and CPU cycles. And when I came to do some hacking on it, I found the real kicker: it doesn't include the source code, so who knows what the hell it's really doing? In the end, we wound up using Zultys ZIP4X4 hardware SIP telephones. These are very expensive for "just" a telephone, but they are stuffed with features, all known codecs, 4 virtual lines, even an integral 5 port (one for the phone, four brought out on RJ45 jacks) 100Mb/s switch, and they are hardware -- my favourite programming language is still 63% tin, 37% lead.
As for phone clients
My honest recommendation would be go for something like the Grandstream, which does everything an "ordinary" phone should do and, being hardware is truly cross-platform. But note, it doesn't have any integral switch so you will take up an extra jack on your ADSL router.
Je fume. Tu fumes. Nous fûmes!
Speak Freely by Brian C. Wiles & John Walker has been around for ages, is open source and thus available for any platform (http://www.speakfreely.org/). It's also very tweakable - you can use it on a 14.4kbps modem with OK quality - and has integration with ICQ and some other messaging programs.
I have been using the sip softphone from xten.com on a Mac - also on windows. They had some claims about linux support some time soon. Tell them to do linux - no one will do it if no one asks for it.
When you use asterisk on a server, you can communicate with it via either hardware IP-phones, or a variety of softphones using either H323, SIP, or IAX (asterisk's own protocol).
For IAX, see iaxclient and some of the phones made from it. There are iaxclient-based soft-phones for all three platforms mentioned.
Yes, that seems so. However, the other VoIP programs listed here are not equivalent solutions, so the need is great. For example, Skype works over port 80, all the other software I've seen requires opening ports in your firewall. Not good.
Maybe there could be a kind of streaming Ogg Vorbis? Maybe part of the work is already finished open source.
Flash and the Flash Communication Server make Live Audio and Video Broadcating over IP easy, easy...
iaxComm runs on Win32, OSX and Linux. It uses Asterisk's native IAX2 protocol. You can use it peer to peer, or with an asterisk server.
h tm l
http://iaxclient.sourceforge.net/iaxcomm/index.
I've spent the last few weeks putting together a home Asterisk box. VoIP with Asterisk is amazing; the fact that I have a fully functional IP-PBX sitting in my living room running on hardware I found at the dump is mind boggling.
The IAX protocol, which is a Asterisk-specific VoIP protocol, is great behind my IPCop box since it effortlessly works with NAT, requiring not a STUN server or any other kind of help. I've bought pre-paid VoicePulse Connect service for long distance calls to PSTN, and since I don't do much long distance its really a cost saver since I don't have to pay SBC all that money. For local calls, I have a clone-Wildcard PCI card I found off Slashdot. This isn't really a requirement, as you can get local numbers in most areas. Just not mine.
Bottom line: If you want to get serious about VoIP, start tinkering with Asterisk. Its going to be the Apache of VoIP. No doubt.
--- Kicking the Cheat since late 2002
Yes! You should look at http://www.openh323.org which is open source VOIP with a wide variety of personal and private applications built on it. It also has several cross platform ports done around it. BSD, Windows, Mac OSX, and embedded linux.
Applications include Gateway, Gatekeper, Conferencing server, firewall penetration, and vaiour clients.
Well actually Grandstream does have a two port version (which I use) and this way you just go in series with your computer.
The real problem lies in that it only works on 10Mb, which when in use will degrade your whole LAN. Best solution if you use GS is to have a seperate VoIP LAN, which is more secure anyways...
well I was just guessing.. I didn't know it was so low though. Thanks for the info.
It is amazing what they are/can be doing with it.
...because it sounds like you're looking for Vonage. If you decide to switch, please let me know, I'd like the referral bonux.
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you're probably right. I imagine they use UDP to avoid the overhead of sending TCP packets... and actually, I think you're definitely on to something with the 'sub-audible noise' thing... in the periods of silence between conversation, I notice... well, it's not sub-audible, because I can hear it... just the tiniest bit of white noise.
Standing at the very edge of my imagination, I peered into the inky void and realised -- I couldn't think up a new sig.
http://mysip.ch/
SIP Software from Siemens Switzerland.
see http://spf.pobox.com/slides/thenewspf/
While these are not open-source, free solutions, most of them offer a free try-out and a very cost-effective fee for their use. Please consider:
1) Marratech This is a fully secure collaboration space that supports both VoIP and videoconferencing. You may want to try immediately by using one of the several available test rooms. Kolabora.com Direct access from here or here (Win, Mac, Linux)
2) Flash Communication Server -FCS offers full web-based VoIP and videoconferencing across mulitiple platforms. To see how easy it is to use go ahead and test it from any major OS on anyone of these online services:
2a. Uvault Demo Rooms
2b. Megameeting
2c. E-boardroom
3) iVisit - This is a very cost-effective audio/video application which integrates the ability to create private spaces. (Win & Mac only)
4) Convoq ASAP - This is a web conferencing and live presentation solution. While the presenter needs to be on a PC, attendees can join from any major OS. Convoq integrates VoIP and videoconferencing using FCS.
Here is a good source for all things VOIP http://www.voip-info.org/tiki-index.php
If you're not cheating you're not trying.
Try X-Lite from Xten.com. It's free and runs on Windows and MAC OS X, Linux coming soon.
I'd be totally happy having VOIP with or without the possibility of
having PSTN (traditional phone network) connectivity and a PSTN number.
I'd be pretty happy using a cell phone for most or all of my voice traffic,
DSL for internet, and MAYBE VOIP/VODSL for enhanced voice capabilities
from my residence if I didn't want to use the cell phone from home.
The VOIP/VODSL is pretty optional if I'm paying for cellular, though.
What is, however, desirable is to get rid of the expensive and useless POTS
"local/long distance" services at home and to keep the DSL!
I'd have and want to keep DSL internet service on which I'm getting from a
Verizon CO and routing through a third party independent data-onlu ISP.
As far as I've heard it's "impossible" by "policy/capability" for them
to sell you a DSL circuit without having functional "phone service" and
an non-disconnected "phone number" associated with the line.
I frankly find that to be likely (a) BS and (b) shocking given the regulatory
nonsense and questionable anti-trust violations with forcing users to do business
with a *unrelated* division of a telecom company (the "local phone" service
provider) just to have access to the network's DSL circuit offering.
I can justify paying some for a cell phone, DSL broadband, and VOIP is in many
ways preferable to POTS service, though it's hard to justify ALSO paying
$40+ a month BEYOND that for POTS "local/long distance telephone service" and
its related fees / surgharges that I'm paying *doubly* or *triply* given my
cellular and VOIP "telephone network access" services.
So has ANYONE figured out how to successfully have
CELLULAR + VOIP + DSL + ISP and NOT POTS and still have the same flexibility
of independent VOIP and ISP providers that one would have if one had
DSL + local telco's POTS + arbitrary ISP + arbitrary long distance IXC +
arbitrary cellular?
I'm sure in some places there are VOIP prodivers / VODSL providers that offer
POTSless VODSL + DSL, and that would be BETTER than what I'm stuck with now
(POTS and cellular and DSL), though I'd think the options would be considerably
more limited if one 'had' to get the ISP/VODSL/DSL services through one carrier
vs. having some choices as to providers.
I like having a choice of ISPs because I run personal use servers for SMTP,
DNS, WWW, et. al. and don't tolerate port blocking, traffic filtering,
use policies against running servers, et. al. I have multiple static IPs,
etc. I like the idea of having choices of VOIP and VOIP-to-PSTN providers
for purely features / standards compliance / trustworthiness / competitive
issues. And of course I like having as attractively priced DSL circuit
access fees for bandwidth as possible.
Anyone have any experiences with achieving this?