VoIP is becoming more common and practical. Essentially it relies upon transmitting audio data packets with low latency across broadband networks. Most of the VoIP services use the SIP protocol.
Kphone/Pi is a port of KPhone to the Zaurus kphone setup guide
TkcPhone is another Zaurus VoiP program tkcphone homepage
A third application that might be worth trying is Linphone, which also has a gui front end; this may only work for sharp-based (non-OZ) ROMS, however
KPhone SI is a SIP User Agent for Linux. It implements the functionallity of a VoIP Softphone but is not restricted to this. It is licensed under the GNU General Public License. KPhone is written in C++ and uses the Qt-Toolkit.
KPhone SI supports as well Proxy Agents as direct communication between User Agents. The latter may be not possible if firewalls and NATs restrict point-to-point communication. Kphone SourceForge Homepage
SL-C1000 Pdaxii13 Quickstart Guide by Raj Prakash
1. Confirm that your microphone/speakers (i.e. 2.5mm mono headphones) works by recording something and playing it back. I confirmed I can record via my 2.5mm mono headphones through my left earbud using KhdRecord and I can playback the recording via XMMS in the right earbud.
Note: You may have to play with the volume settings. Pdaxii13 comes with OSS Mixer by default, so play around with it until you get your desired voice record volume and playback volumes.
2. Get an account with a supported SIP provider. I used sipphone.com. Confirm that your account is active and have enough SIP minutes (if you need them) to make outgoing landline calls.
3. Install the package on the media of your choice from here.
4. Unfortunately the preferences screen is larger than the Z's viewing area however you can find the config file in your home directory under ~/.qt/kphonesirc
Note: You can edit the preferences via the software interface but when you try and hit "enter" to save the changes, the software seg faults. Not to worry! It did in fact save your settings to the location mentioned above.
5. Configure Kphone/Si for your sip account. For sipphone.com accounts, you will enter your SIP # as the "User part of SIP URL", your SIP # as the "Authentication Username", SIP Domain as the "Host Part of SIP URL". As an example, here is the config for my account:
Full Name: Raj Prakash
User Part of SIP URL : 189384924770
Host Part of SIP URL : proxy01.sipphone.com
Outbound Proxy : proxy01.sipphone.com:5060
Authentication Username (optional) : 189384924770
If your network setup requires it, also go into Preferences menu and enter in the STUN Server information.
6. Dial sipphone.com's test numbers as specified in your sign-up confirmation email and make sure you hear the recordings. If you can't hear the standard recordings, something went wrong. Make sure that you entered all your authentication information and that the bottom left corner of the Kphone/Si window shows you are connected via a connected icon. This means your auth info was accepted by sipphone.com and you are "registered".
7. Here was the kicker for me. Kphone/SI has 5 available codecs, ulaw, alaw, gsm, ilbc, and speex. By default, Kphone/SI tries to use them in that order unless you disable that particular codec by using a -1 in the config dialog box (or the config file). I disbled speex, ilbc, alaw, and ulaw as I read some of them are very resource intensive codecs and the Zaurus is not exactly a processing power house. Enabling them one by one, I found my optimal settings are gsm as "0", alaw as "1", and the rest disabled via "-1". Now I can make calls out to landlines and mobile lines no problem.
If you've gotten this far, you can now make calls via Kphone/Si using a sipphone.com account. I've used this account from the US to various places in the world. I've used this setup from Paris, France in hotels and at offices. I've also used it while on vacation in Italy, Germany, CZ, Austria, Switzerland, etc.
The 750,760,860,1000,3x00 have a dual-purpose headphone/microphone socket. When the audio driver is used for audio-in one channel is switched to a microphone input.
It is thus possible to use a universal headset (usually designed for home cordless telephones) with a 2.5mm jack plug with three connection rings, with a 2.5 to 3.5 adaptor, and plug it into the Z to VOIP calls.
The 6000 has a built-in microphone and speaker, so can be used directly, quality suffers due to echo, so a headset is ideal; in this case the 6000 has a 2.5mm jack. I think that the same universal headset should work.
Note that some headsets with 2.5mm jack plugs have four connector rings on the tube, these are designed for mobile phones, with the extra connector being for a push button to answer or dial. These might not be suitable for the Zaurus, but they are useful if you have an Audiovox RTM8000 gsm/grps compact flash modem card (it might also work with an Enfora, but I am not sure)
Original pages, will be removed when the page is mature enough
|VoIP software||There are at least two VoIP apps available|
|VoIP hardware||Headsets and connectivity options|