Author Topic: Kiax?  (Read 6225 times)

newkirk

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Kiax?
« on: January 12, 2007, 04:56:45 pm »
I've been trying to cross-compile kiax and getting an error where 'as' can't find libopcodes - which is not referenced anywhere in the kiax code itself.

Anyway, I was hoping someone with a stable proven development system would be willing to build kiax for r121.  Qt-based, should be self-contained.  (the source tarfile includes IAX, GSM, iLBC, and Speex libraries)

http://sourceforge.net/project/downloading...tar.gz&97580772

IAX2 (Inter-Asterisk Exchange protocol) is SOO much easier to work with than SIP.  With SIP VOIP the SIP protocol (udp 5060) is used to set up/manage calls, but the audio goes via RTP over effectively random UDP ports - this makes it a major PITA to run SIP through NAT. (symptom - one-way audio or none at all)   With IAX2 (udp 4569) it encapsulates everything over a single connection & port, making it a breeze to NAT.  The Asterisk VOIP server software (duh) supports IAX, as do several soft- and hard-phone offerings, and several VOIP providers (like Teliax) directly support IAX. (and asterisk)   Depending on audio codec, IAX can handle a single call in as little as 28kbps, two calls in 40kbps.  I know I can achieve that over wifi without thinking about it, what I really want to test is whether I can push a call over my cellphone's unlimited GPRS/EDGE data service without undue trouble.  (I run my own asterisk server and have a line with Teliax, so this would be really useful)

j
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louigi600

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Kiax?
« Reply #1 on: January 13, 2007, 05:03:33 am »
While going trough the packages to install my c860 with r121  I noticed ekiga:
(formerly known as GnomeMeeting) an open source VoIP and Video Conferencing tool.
If you want more details look at
http://freshmeat.net/projects/ekiga/

Hope that solves your problem.
Regards
David

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jpmatrix

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Kiax?
« Reply #2 on: January 13, 2007, 10:12:08 am »
i've just tried Ekiga under pdaxii13 build5.2 : seg fault.........
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louigi600

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Kiax?
« Reply #3 on: January 13, 2007, 10:24:16 am »
Quote
i've just tried Ekiga under pdaxii13 build5.2 : seg fault.........

As far as I understand pdaxii13 is based on beta3 while r121 is a snapshot of cvs beta4.
I have not yet tryed myself because I've got other audio trouble on my c860 ... but as sso as I sort them out I'll try on r121 to see if it does any better than seg fault ;-)
Regards
David

SL-c760*  pdaXrom latest
SL-c860    pdaXrom latest ;-)
SL-c1000  pdaXrom Latest
Acer Aspire One running slackware and Clash
Toshiba AC100 running ARMedslack and Clash

*with some hardware problems but good for testing

newkirk

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Kiax?
« Reply #4 on: January 14, 2007, 02:36:21 am »
Quote
While going trough the packages to install my c860 with r121  I noticed ekiga:
(formerly known as GnomeMeeting) an open source VoIP and Video Conferencing tool.
If you want more details look at
http://freshmeat.net/projects/ekiga/

Hope that solves your problem.
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No, Ekiga is SIP and H323 only.  Both are a PITA to NAT.  I'm trying to get an IAX2-supporting VOIP client, like iaxcomm,  or (my preference) kiax.

IAX2 is easy to NAT, supports multiple calls in a single packet stream (SIP+RTP needs at least 3 ports for a single call), is more efficient bandwidth-wise (and usually processor-wise) than SIP+RTP, and offers encryption.

j
« Last Edit: January 14, 2007, 02:42:54 am by newkirk »
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adf

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Kiax?
« Reply #5 on: January 14, 2007, 03:10:17 am »
very interesting idea, actually. (the gprs thing) I've been curious about setting up asteris for a while.
 I'm guessing the old sharprom ziax phone won't do?

The obvious question is --have you found the kiax source and tried to compile it?  What are the difficulties? Is this something the OE folks (given the projected capabilities of the neo) might be motivated to solve? Is r121 easier to build for or on than beta3?  (I kinda thought it wasn't)
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newkirk

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Kiax?
« Reply #6 on: January 15, 2007, 11:22:22 pm »
Quote
very interesting idea, actually. (the gprs thing) I've been curious about setting up asteris for a while.
 I'm guessing the old sharprom ziax phone won't do?

The obvious question is --have you found the kiax source and tried to compile it?  What are the difficulties? Is this something the OE folks (given the projected capabilities of the neo) might be motivated to solve? Is r121 easier to build for or on than beta3?  (I kinda thought it wasn't)
[div align=\"right\"][a href=\"index.php?act=findpost&pid=151143\"][{POST_SNAPBACK}][/a][/div]
I've not tried the ziax package, I'll have to see if it can do anything under pdaXrom. (I'm on the road this week, not sure when I'll get the opportunity) When I try to build the kiax source (or the Qtiax source for that matter) it reaches a point where it's working on main.cpp and I get ".../armv5tel-cacko-linux/bin/as: error while loading shared libraries: libopcodes-2.16.1.so: cannot open shared object file: No such file or directory".  I posted about a week ago hoping for advice on that.  'libopcodes' doesn't appear anywhere in the kiax source itself.

Funny you should mention the Neo.  Three things have come together leading to me trying to compile an IAX softphone.  Desire for a nice softphone on my Z (I've not tried ekiga, am really looking for IAX - not even sure if I can route SIP+RTP over GPRS without VPN...), administering a few Asterisk servers, and recent discussion on the OpenMoko mailing list.  With Zaurus over bluetooth, or a phone environment like OpenMoko, it's possible to route VOIP calls over gprs to your own asterisk server at home, with a $40 FXS jacked into your home phone line.  You can set up Asterisk via Trixbox (centos-based install [centos is based on redhat enterprise] of asterisk and support packages - dedicated server but full linux so you can install services like webserver, mailserver, DNS cache, timeserver, all sorts of things useful in a modern SoHo network) in an hour and have it bridging calls from gprs to landline, handling voicemail, interactive-voice interface with the server, etc.  I know it varies wildly by location, but T-Mo in the US I have unlimited data for $20 a month, but $50 for 1000 minutes.  $20 is cheap for unlimited mobile data, but I still want to get my money's worth...

If you want to 'play' with asterisk, I recommend Trixbox installed under VMWare.  You'll need either an FXS (hardware linking computer to phone line) or a VOIP provider who allows freedom of choice in your client.  (Vonage will only allow a non-vonage 'device' to connect for certain classes of business plans or for a second line of service specifically for computer access instead of the provided Linksys FXO interface box - by contrast, Teliax offers cut'n'paste configs for various versions of Asterisk right in the customer support page)  The FXS can vary from a PCI card (like X100P) to an external box like a desktop router that supports a half dozen phone lines, and they can vary from $40 to $2000.  Asterisk has a deep well of potential, hence is (unnecessarily IMHO) somewhat complex to configure.  Configuring it via FreePBX web interface (provided with trixbox) has a bit of a learning curve both in VOIP/Telephony concepts as well as the software itself, but configuring asterisk by hand in /etc/asterisk (as I first learned it) can be a traumatic experience, especially if you want to achieve something unusual.

j
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adf

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Kiax?
« Reply #7 on: January 16, 2007, 02:28:08 am »
well have a look in vmaware... i had done a brief look at the configs and left it:)  you ideas make it seem more worthwhile.
**3100 Zubuntu Jaunty,(working on Cacko dualboot), 16G A-Data internal CF, 4G SD, Ambicom WL-1100C Cf, linksys usb ethernet,  BelkinF8T020 BT card, Belkin F8U1500-E Ir kbd, mini targus usb mouse, rechargeble AC/DC powered USB hub, psp cables and battery extenders.

**6000l  Tetsuized Sharprom, installed on internal flash only 1G sd, 2G cf

Kanuvar

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Kiax?
« Reply #8 on: April 01, 2008, 02:46:29 pm »
Hi!..
any news in this topic ?
I have been trying to compile any iax client in mi cross-compiler whit any luck.
There is some hope ?

P.D.: sorry for my bad inglish