# voice conversation across the network

## tnt

is there any ebuild to enable voice communication in the LAN or even across the internet?

 :Question: 

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## Jonbond

dont know about an ebuild but i know of a few voice programs

unfortunately dont have their urls saved but  try googling 

ventrillo  or  Teamspeak

we use ventrillo for a voice chat while playing bf1942 / cs / dod / ffxi.

it works great and the sound / voice is execellent plus its freeware.

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## befortin

From Slashdot.org, on Thursday March 04

http://developers.slashdot.org/article.pl?sid=04/03/04/038241&mode=thread

============================================

Howard Vanbel writes "Apparently the developers of GnomeMeeting have released the final v1.0 version of the videoconferencing/VoIP software. GnomeMeeting started as a final studies work at the Department of Computing Science and Engineering of the Universite Catholique de Louvain and after 3 years of development, GnomeMeeting 1.00 is ready! GnomeMeeting is the most advanced Open Source VoIP and videoconferencing software available - there's more info in the project FAQ."

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## tnt

i'll try to find something - thank you...

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## Chris W

That should be Ventrilo - no double L.   Use a double L and you get a whole world of squatting search engines etc.

http://www.ventrilo.com/

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## blackdeath

I downloaded the Windows binary of Ventrilo and ran that with Wine, the problem being after the usual font matrices bit the window for the install just closed.  Anyone had this issue before that could offer insight into possible reasons and could help me remember where that log file is in Wine   :Rolling Eyes: 

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## smart

there are plenty of native programs, and ebuilds. try this:

emerge -s phone

the good thing with these phones, they are running SIP protocol. which is a more and more supported by IP phone companies and other tools like the linux software PBXs (mosta dvanced currently i think is asterisk).

but SIP is e.g. iuncompatible with M$ netmeeting which prefers h323.

for h323 there is a tool we have and that i'd say is more worked out than those above. thats:

emerge gnomemeeting

so what i'd like to see is gnomeeting to support SIP or one of the others to become as good as gnomemeeting.

all in all, you've got plenty of choice of doing your voicecomm and for some, videoconferencing is delivered in the same box.

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## Shiryou

Does this mean Ventrilio actually works with the bog standard free wine? I've heard it aint so

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## m.b.j.

Im new on sip, voip ... i have general questions about this topics:

1. I is it possibel to run  voip software in a privat network? (can 192.168.1.101 phone 192.168.1.102?)

2. Can Linux-Users call Window-Users?

3. Software i have to use?

4. Does voip software use a client-server construktion or  p2p?

5. Is the name resulution linke dns?

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## smart

1) It's possible. You do not need a special "service provider" locally or externally per se. You CAN make use of such though on your own, externally or both. Think of the parts like this: a) You have phones on the end (on you mchine) that are the network interface to you. These are already fully capable of communication and can chat to each other. b) You add a central service provder, aka telephone system, that offers you things like call routing, voice mail boxes, conferences, call queues and whatnot c) there can be bridges to other voice networks, either just to different VoIP standards or to the actual telehpone network.

2) Yes, as long as they both use the same kind of, aka standard in VoIP software. That doesn't make it as unlikely as usually, since in principle there are no more than two, plus proprietary but those are actually not from M$ this time  :Smile: . The two main standards are H.323 and SIP, which actually focus on the negotiation part, that is, how to set up calls. Netmeeting and Gnomemeeting for example set on the H.323 standard, but this standard is more or less a bit outdated. Things seem to move towards the newer and more capable SIP standard. As such, the *meeting pieces of software are quite well polished, whereas sth. like kphone is more future proof. Even for Windows there are free Softphones, and they are apparently quite solid as well, like SJphone, the on from firefly and others. Teamspeak and Ventrilo AFAIK are a completely differnt thing. They are not for general purpose telephone kind of use, but more personal. So you can set up your chatterpartys with them, but they are not meant to run inside a bigger network comprised of things like mentionned in 1)

3) Yes, Software you'll have to use  :Smile:  There's a lot, go figure. If you have specific interest/request, repeat that... otherwise lists would be hughe...

4) Somewhat both, but in the end, it's not of interest to you. I guess your point comes down to "Do i need a paid provider at some time based on the principles of the technology ?" and the answer is no.

5) You can go direct if you know the data of the one to be called. Other than that there are different brands of telephone book systems. Some based on registration on a kind of provider, like firefly (but that ones free) or more general approaches like ENUM using DNS.

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## Hypnos

You could just use GnomeMeeting, which is in Portage.  I'm in No. California, and I've had decent conversations with people as far as Poland.

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## m.b.j.

Thx for your posts,

a lot of my questions are solved at this time, im currently emerging gnomemeeting...

additional information about my software question:

All this Senarios are yust for 5-6 clients so my adsl connection (786/128) can provide it. 

Senario 1:  (Homenetwork)

If client1 will to phone client2, client1 sends a query to a local server and recieves a ip address, then client1calls client2 direct, not over a server, the server just does a type of name resulution. This should work in my home network.

Senario  2 (optimal): (Home Network and Internet connectet)

 A client specification by ip addresses is not usefull, in a case where more then one person is behind a router,  just one person could be called. So there has to be an other naming system (maybe like email). A voip session would be opend from an internet-client to my server , my server opens a new connection to the privat-network client like a proxy. If an internal Client whants to phone an external client the connection goes over the server/router/gateway. 

Im sorry for my bad englich!

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## smart

Since you decided for gnomemeeting (which is a good client) and thus H.323 and i cnnto give you direct advice on what software to use for the other components, i'm settled with SIP myself.

Scenario 1:

You would not even need a "server" if in your Homenetwork you can rely on known addresses to the users computers. If  that's not the case, you need a registration server. This can probably be done by the same software that provides your proxy for scenario 2.

Scenario 2:

Absolutely correct, to traverse the NAT for more than 1 client you need a proxy.

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## m.b.j.

I did not decided to use gnomemeeting, i have just emerged it 

```
# emerge unmerge gnomemeeting
```

could solve this, gnomemeeting is just a test, i wanted to see the interface, the options, and the general usage but it does not run on my amd64... (i have got serveral problems with gnome) but this is not the place to put them.

I want to use sip too, could you give me the names of these programs? Is linphone one of those progs?

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## smart

Linphone is SIP based, yes, but admittedly as said before, in my opinion the end user software is not as far in their development as the earlier started h.323 like gnomemeeting. nonetheless, backend stuff to me seems to have a better and wider choice. as for a pure proxy partysip is a good choice in my opinion.

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## m.b.j.

But what software i can use for proxy?

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## smart

partysip  or SER for examples

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## KenTI

ok i'm trying to setup VOIP, i think i'll try kphone and SER (partysip fails to compile)

is there any howto explaining all the steps to set this up?

i have a 640/256 adsl line with a router, i wanted to try it first within my homenetwork and then from the outside

and do i need any special hardware or the microphone and speakers are just fine? i'm using the builtin nforce audio card on an ASUS A7N8X-E deluxe

another thing, is ALSA supported?

thanks

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## m.b.j.

I can not explain it to you (i self a noob to this), but look for your self: http://www.iptel.org/ser/admin.html

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## EliasP

Have you tried net-im/skype ??

Greetings

Elias P.

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## smart

any audio equipment should be fine. ALSA is your best bet even, with some few exceptions.

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## STEDevil

 *smart wrote:*   

> Linphone is SIP based, yes, but admittedly as said before, in my opinion the end user software is not as far in their development 

 

Amen to that. I would go as far as say it's horribly bad and next to unusable as a phone. The other option in portage for SIP software, kphone, is if possible even worse.

Anyone looking for a SIP VOIP software should probably take a look at eg http://www.sjlabs.com/ instead. MUCH better then the two mentioned before. Apart from giving feedback of what is actually going on it also lists things like made calls, recived calls, missed calls, etc and even allows for multiple profiles.

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## Stolz

 *KenTI wrote:*   

> ok i'm trying to setup VOIP, i think i'll try kphone and SER (partysip fails to compile)
> 
> 

 

KenTI, did you get it to work?

I want to make VoIP calls between 2 boxes on the same LAN (no router). I've installed SER with the default values. As you can see in the readme: *Quote:*   

> default configuration is very simple in order to be easily installable. It provides minimum features. Particularly, authentication is by default disabled, which means anyone can register using any name with the server. (This is on purpose to avoid installation dependencies on MySQL which is needed for storing user credentials.)

 

So it's suposed I can register because there is no authentication at all. The problem is Kphone always give me a "Registration failed.Too many Hops" error. Any kinf of advise is appreciated.

Thanks.

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## R!tman

 *STEDevil wrote:*   

>  *smart wrote:*   Linphone is SIP based, yes, but admittedly as said before, in my opinion the end user software is not as far in their development  
> 
> Amen to that. I would go as far as say it's horribly bad and next to unusable as a phone. The other option in portage for SIP software, kphone, is if possible even worse.
> 
> Anyone looking for a SIP VOIP software should probably take a look at eg http://www.sjlabs.com/ instead. MUCH better then the two mentioned before. Apart from giving feedback of what is actually going on it also lists things like made calls, recived calls, missed calls, etc and even allows for multiple profiles.

 

I agree with you, STEDevil. linphone really is not as good as sjlabs. I cannot find colorful words like you did, your statement comes alot closer to what linphone is than mine. 

But...although sjlabs is pretty easy to install, I'd rather have an ebuild for it. Even if it is a binary one. Maybe someone can do that.

This may not be so hard, maybe I can try to get that rolling.

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