VoIP, SIP and STUN

GungHo · Apprentice Joined: 27 Aug 2004 Posts: 254

Hi Folks,

hopefully this is the right forum for my question

I have the idea to make VoIP phonecalls over a DSL Uplink. Therefore I need a software, which is able to do signaling with SIP, transfer the digitized voicedata with STUN, and is able to use a headset connected to a soundblaster PCI soundcard for doing the actual audio IO.

Is there a software in portage, which is able to perform this ?

Linphone seems to come close to my requirement, but seems not to support STUN, but RDP. Kphone related I'm not sure, there is not much info regarding kphone floating around ?

Has anybody a working configuration of this kind, or can give me some hints, info, whatsoever regarding this topic ?

Thanks in advance

The Sentry · Posted: Mon Nov 01, 2004 5:58 pm Post subject:

I'm trying to get it to work for myself right now, I'm using kphone. But unfortunately it doesn't work correctly with aoss, so it is a bit tricky.

Kraymer · Guru Joined: 27 Aug 2003 Posts: 349 Location: Germany

Hi there..

I know this thread is quite a bit old.. But I wondered if any of you has figured out a working solution for voip..
These days I tried kphone, but quality is really bad..

Do you have another solution?

Sebastian

Hal9k · n00b Joined: 06 Aug 2003 Posts: 6

Linphone, at least the testing version in Gentoo, is an option. It's flaky though, especially with configuration, so I always restart it after a config change to make sure it "took." Unfortunately somewhat difficult to set up, but I've had it working with Free World Dialup (fwd.pulver.com). It is OSS only, but does fine with my ALSA OSS emulation.

SJPhone (http://www.sjlabs.com/sjp.html) appears to be the best option right now. It's an x86 binary and not part of Gentoo, but does offer H.323 and SIP signalling, and offers the upfront STUN connection so that a pathway can be worked out in case your device is on a private subnet. It is plain and tricky to configure, but is rock solid. Also OSS only, but not a problem here.

Gnomemeeting looks really promising. Doesn't yet offer SIP, but offers the older H.323 as well as STUN (version 1.20+) for NATted devices. It offers ALSA though I started getting soundcard errors from it, when the ALSA "default" device support was introduced. I'll check out this package once again when SIP comes out.

There's an echo test service reachable at sip:613@fwd.pulver.com if you can handle the ~90kbps "mu law"/uncompressed codec. An information service is available at sip:18005558355@tf.voipmich.com with either that codec or the much lower bitrate GSM codec.

Kraymer · Guru Joined: 27 Aug 2003 Posts: 349 Location: Germany

Hi!

Thanks for your answer..

I ordered my own headset yesterday, hoping it'll arrive soon. Don't want to beg my neighbor for it everytime

Extensive testing of different applications will continue then.

Yesterday, for an hour or so, I experimented with X-Lite (aka Sipphone). It's a windows/macosx application which - as it turns out - seems to work very fine under wine and has support for many codecs..
I tried kphone several times, it didn't meet my needs

Also, I can't see any active development from the webpage I found.
Sjphone.. I didn't spend much time, seemed kinda unintuitive to manage (for direct vs. sip calls, but I'll definitely get into it again later..).

I wonder why it seems so hard to code a good voip application. Skype seems to work for everyone but I don't want to take from that kazaa guy

We'll see. Last word isn't spoken yet.

Sebastian

Hal9k · n00b Joined: 06 Aug 2003 Posts: 6

Hello again. Glad to see you're having some luck, and nice to know that X-Lite under Wine is an option!
Skype looks to be popular, but I haven't gone that way as it doesn't appear to provide interconnects to other VoIP systems. Also I don't believe it uses the standard SIP protocol.
With SJPhone, you'll need to switch from the H.323 profile to the SIP profile. DON'T choose the "w/o STUN" one. Be sure to click "use" once a profile is selected. Once done, you'll be able to dial the test numbers listed previously. Basically, you'll be in direct client to client call mode, and your SIP address will be sip:<your Linux login>@<your public IP address/hostname>.
Later, you might want to grab a FreeWorldDialup (fwd.pulver.com) number. Then you'll create a new profile (SIP proxy). The proxy domain and user domain get set to "fwd.pulver.com". Check "register" and UNcheck "strict outbound." "Initialize" your profile (involves filling your FWD # and password) then "use" it.
By using SIP-based services, you'll later be able to buy a broadband router or dedicated box that lets you plug in a regular telephone. The box will auto-connect to your provider and permit you to use the regular phone while bypassing your computer entirely.
Enjoy this new world.

Kraymer · Guru Joined: 27 Aug 2003 Posts: 349 Location: Germany

Hal9k · n00b Joined: 06 Aug 2003 Posts: 6

Hello once again!
Well, I tried X-Lite under WINE just for kicks. It took me through the installation steps, a nice start, and did install, but possibly with an abort at the end. The application though just exits upon start. Fortunately GnomeMeeting now has SIP support in its CVS tree. I tried manually compiling, but it doesn't recognize my GNOME. Looking forward to its release.

Skype does look to be very user friendly, given that it doesn't have the complication of SIP (and NAT). I hope that they will peer with FreeWorldDialup and others to make VoIP as a whole more attractive. There's another NAT friendly protocol, one that is open but is not (I don't believe) a standard. It is IAX, the Inter-Asterisk Exchange protocol, used by the Asterisk software PBX systems. Asterisk is a Gentoo ebuild and allows you to run your own telephone "central office". There's also an IAX user client called IAXComm that seems to have good potential.

Here's my (very limited) understanding of how a SIP call works. We'll assume a FreeWorldDialup (FWD) subscriber is placing a call, is behind a NAT device, has no ports forwarded, and is using STUN (very helpful for NAT). The SIP client registers with FWD (SIP port 5060), so that FWD knows to be ready for a future outgoing call. Also allows FWD to deliver incoming calls. The client periodically sends update messages to let FWD know the registration is still active. Perhaps more importantly, it keeps the NAT device from closing down the pathway. (With the pathway down, incoming calls get dropped since we're not forwarding any ports. Everything is UDP, so there's no socket being kept up.)

Now let's place a call. The client goes to the STUN server (not necessarily part of FWD) on port 3478. It needs to learn its public IP address and source port as seen by the server. (Both the address and port get changed when passing through NAT, thus the client can't just check its connection table.) A new connection (coordinated over port 3478) is established to the STUN server on a "random" (but coordinated) target port. The client learns what's needed from the server and the session is concluded. Because this is UDP though, that "random" port pathway remains active in the NAT device.

We switch to the SIP protocol (port 5060) and place the call. If the call connects, then your client must announce how the callee routes voice packets to you. The voice itself is not part of SIP, but is part of the Realtime Protocol (RTP), and is a separate UDP stream. The voice packets go between caller and callee, bypassing FWD. To get RTP connected, the client must specify your address (your public address!!...which STUN earlier determined) as well as the port that your client is listening on. The client cannot directly mention a port of its choosing since ports get translated. Therefore, it mentions the source port that it learned from the STUN session. By using that one, it can capitalize on the fact that the pathway is still up, therefore permitting incoming packets to reach your client. I believe the callee's RTP source port must match the "random" target port used with the STUN session, as this is the other half of the pathway's specification. Basically, most NAT devices still honor the pathway even though we're no longer talking to the STUN server itself.

Did you sign up for sipgate.de? I notice they peer/interconnect with FWD, so we could have a test call one of these days. Would certainly be long distance as I'm in the U.S.

Harold

Kraymer · Guru Joined: 27 Aug 2003 Posts: 349 Location: Germany

fkryszon · n00b Joined: 01 Nov 2004 Posts: 64 Location: Belgium

found this through g00gle:
[url]
http://xten.com/apps/xprolinuxbeta/
[/url]

Sancho666 · n00b Joined: 29 Aug 2002 Posts: 12

Hi,

the german site www.onlinekosten.de postet news about the availability of xten's X-Lite for linux.

Download path: http://www.xten.com/index.php?menu=products&smenu=download

I've tested it with debian and its much better than kphone.

Sancho

oneeyedelf1 · Tux's lil' helper Joined: 04 Feb 2004 Posts: 124

ditto, couldnt get kphone working, x-lite, although the interface is huge was no trouble to setup

boatman · Posted: Sat Oct 29, 2005 11:06 pm Post subject: x-lite runtime error

This is the Linux version of x-lite.

./xtensoftphone
./xtensoftphone: error while loading shared libraries: libglade-2.0.so.0: cannot open shared object file: No such file or directory

I emerged libglade and can see libglade-2.0.so.0 in /usr/lib/libglade-2.0.so.0

How do I tell xtensoftphone where libglade is located?

Thanks

pharoh · Posted: Wed Nov 30, 2005 1:53 pm Post subject:

A little late but did you try running ldd against the binary to see where it EXPECTS the glade lib to be it is probably differant :-)

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fbcyborg · Advocate Joined: 16 Oct 2005 Posts: 3056 Location: ROMA

Hi!

I have the same problem here on x-lite-2.0.1105.
Here's my ldd /opt/bin/xtensoftphone about the libglade file:
libglade-2.0.so.0 => not found

It doesn't say where should it expects that library.
Anyway, libglade-2.0.so.0 is here: /usr/lib64/libglade-2.0.so.0
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pharoh · Posted: Fri Jul 31, 2009 12:43 pm Post subject:

I think x-lite is a 32bit app so it might need the lib in in /usr/lib32 but i don't think libglade has a compat package for 32bit?
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fbcyborg · Advocate Joined: 16 Oct 2005 Posts: 3056 Location: ROMA

I tried to use linux32 without success, but I guess a 32 bit chroot would be better.
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pharoh · Posted: Fri Jul 31, 2009 12:49 pm Post subject:

have you tried Ekiga? I did not care much for x-lite last I tried it.
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fbcyborg · Advocate Joined: 16 Oct 2005 Posts: 3056 Location: ROMA

No, I didn't try it yet!
I try it now!

Thanks a lot!
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boerKrelis · Posted: Fri Jul 31, 2009 5:32 pm Post subject:

doesn't net-im/twinkle fit the bill? I've been using that one a lot and I like it.

fbcyborg · Advocate Joined: 16 Oct 2005 Posts: 3056 Location: ROMA

Using ekiga, I'm not able to connect to the SIP provider. Using x-lite from windows, yes!

I will try twinkle as soon as possible, thank you!
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