U firmi koristimo Cisco IP telefoncice:

http://www.cisco.com/warp/public/cc/pd/tlhw/prodlit/7960_ds.htm

Mada, umesto hardware telefona mozes koristiti bilo koji SIP uredjaj na primer - pogledaj po netu, ima ih mali milion. Recimo Ahead SIPPS :) www.nero.com

- Moze li ta VOIP mreza primati pozive sa analognih brojeva kojim se vecina ljudi koristi, i koja je oprema potrebna za te pozive?
- Kakva je kvaliteta zvuka?
- Koliko korisnika odjednom moze koristiti VOIP?

Trenutno najkvalitetniji IP net phone (potpuno besplatan) je TAICHI 2...koliko cujem nemoguce ga je prisluskivati... ;-)

www.gsm-tech.de

TAICHI2

Features
• Free calls to other Taichi users
• Perfectly with any analog phone set or office PBX central system
• Easy set-up, easy use
• USB powered, no power adapter needed
• Global roaming number
• Crystal clear voice quality
• LED status indication


Taichi 2 allows you to use a regular telephone to place FREE
PC-to-PC calls over the Internet.
Simply installing the software and easily plugging Taichi2 into a USB port makes any PC into a powerful and intelligent VoIP terminal. Remember, wherever you are roaming over the Internet, with the unique Taichi number, communicating with other users has never be easier.


System Requirement
• Windows 98Second Edition/Me/2000/XP
• USB Port
• 32 MB RAM
• 300MHzCPU Speed
• Internet Connection (at least 10KB/s)
(ADSL, Cable Modem, Dial-Up Modem, Wireless LAN or PHS)
• Dynamic or fixed IP, real or virtual IP



This is a "must have" tool for people that have friends all around the world and don´t wanna pay money for communication-by-phone anymore. Each Taichi2 owner have a unique Taichi phone number that makes it possible to make free phonecalls through the internet as easy as with any conventional phone. Simply connect a regular, analog phone to the Taichi2 phone Adatpter(that is connected to a PC with Internet connection), dial your friends number and enjoy in the crystall clear voice quality.


www.gsm-tech.de

Ako ti je cilj da preneses glas izmedju dve tacke preko interneta,
mozda biti bila zanimljiva Cisco ATA 186 (10mbps, Ether.port, + prikljucak za dva obicna telefona). Najlepse od svega sto ti pored dve ate u minimalnoj konfiguracije ne treba nista vise. Izadje oko $200/komad.

Ako planiras da imas i izlaz/ulaz u fixnu/gsm javnu mrezu prica se komplikuje (citaj $$$$).

poz.

kako mu ne treba ništa sem ATA-e!? Kako će ATA #1 da zna kako da dođe do ATA #2? ;-)


Marko.

Pretpostavljam da ima neki oblik IP vidljivosti izmedju ATA.

ATA#1: IP:A.B.C.D ATA#2:IP: X.Y.Z.W

ATA#1 bira ATA#2 (DTMF telefon) : 'X*Y*Z*W#'. i obrnuto. Ako se ''X*Y*Z*W#' stavi u memoriju telefona (ili nedaj boze ispered kucne centrale), jako zgodno :)

iza tarabe se valjda stavlja UID (ID linije ate), ali ako nije iskonfifurisan zvoni prva linija.

Druga opcija je da ate stave u HOT LINE rezim, tj. da dizanjem slusalice jedne ate druga zvoni (prvi SW je ovo mogao, cuo sam (nisam probao) da novije verzije ne dozvoljavju ovo).

pozdrav.

Ah, nije mi palo na pamet da biraš direktno po IP-u :-). Meni je uvek lepše da okrenem broj, a da mreža ispod zna šta radi - to je po meni pravi VoIP :-)


Marko.

Ok, ima resenje i za pak K dolara, ali je ovo sasvim OK. ako firma ima dve, tri filijale, i u svakoj kucnu centralu, sa nekoliko ata moze sve jako fino da se napravi. Bolji PBX-ovi mogu da obezbede filtriranje brojeva (tj. davanje izlazne linije u skladu sa pocetkom broja), te je varijanta sa 'okretanje broja' i ovde moguca. Ima instalacija koje u unutardrzavnom saobracaju isplacuju same sebe za 15 dana. :)

Naravno, nemoj me pogresno shvatiti, FXO portovi i GK su dobrodosli u svakom slucaju :)

pozdrav.

A mi smo malo pobegli sa teme u prethodnim postovima , ali :

- Moze primati, za to su na uredjajima potrebni portovi koji gledaju ka javnoj mrezi (cisco ih zove FXO). Uredjaji se onda vode kao "Gateway-i".

- Kvalitet zvuka zavisi od vise stvari. Na prvom mestu kasnjenja (zbog kompresije na pocetku i na kraju, i ukupnog kasnjenja u prenosu). Ukupno kasnjenje u telefonskom saobracaju koje je vece od 300ms se smatra neupotrebljivim (ovo je opsti standard za telefoniju ne samo za voip). Kao test probaj ping na lokaciju na kojoj bi ti bio drugi kraj VoIP veze.

Na kvalitet utice i koder koji se koristi, ide u rasponu od 5.6kbps do 64kbps. 5.6 kbps je na UDP nivou, ukupan on wire protok je oko 20kbps.

- Koliko propusnog opsega, toliko i korisnika. Znaci na jednom modemskom kanalu (50kbps) bi verovatno mogao isterati dve veze.

pozdrav.

System Requirement
• Windows 98Second Edition/Me/2000/XP
• USB Port
• 32 MB RAM
• 300MHzCPU Speed
• Internet Connection (at least 10KB/s)
(ADSL, Cable Modem, Dial-Up Modem, Wireless LAN or PHS)
• Dynamic or fixed IP, real or virtual IP
Ovo pise da je potrebno da bi se koristila ona spravica za IP telefoniranje Taichi, e sad jel moze meni neko reci ko se razume, posto ja nisam bas siguran, sta je to "Dynamic or fixed IP, real or virtual IP "? i dali to ima svako ko ima internetsku konekciju?

Jedno jednostavno objasnjenje:

Za lokalnu upotrebu (na primer privatna FO mreza) VOIP je izvodljivo na vise (vec navedenih) nacina, sto isto zavisi od financiskih mogucnosti.

Koriscenje vise mreza za VOIP je skoro neizvodljivo - kao prvo iz finansiskih razloga, osim ako su sve mreze FO.

Pozdrav

Momci, kasnite, kasnite sa tehnickim novitetima. IP telefonija je vec u upotrebi. Mali adapter, ukopcass u ADSL liniju i navali ...

Phonecalls should be free !

Telio defies conventional wisdom on
cost of voice and videocommunications


"Phonecalls should be free" may seem like a suicidal motto for a service provider.....
.....but then again; Telio is not a traditional telecom operator !
As the first company in Norway, Telio is delivering pure IP-based, primary line telephony services to all Norwegian households that have, or that can have, a broadbandconnection. Our service went live in January 2004.

Because we are utilizing the best and most modern technology we can offer our customers...




National calls : FREE - all day - every day - all year

International calls* : FREE - all day - every day - all year



..... simply put : with Telio the cost of a phonecall to EVERY regular fixed line telephone is simply zero**.



Currently Telio is offering its services in Norway, but we are planning the launch of similar offerings in numerous European countries (including all of Scandinavia) throughout 2004.



Currently we only offer voice communications (albeit a lot cheaper). However, we are already testing and evaluating more advanced communication services; like plug and play IP-based videotelephony.



Telio is backed by a long range of VoIP industry veterans and experts. Our Board of Advisors include, but are not limited to, Jeff Pulver, Patrik Fältström, John Horrocks, Jiri Kuthan and SIP expert Rohan Mahy of Cisco Systems. Our CTO, Alan Duric, is internationally recognized as key contributor to the development of VoIP technology.





With Telio; European telecom industry will never be the same ............

______________________

* For International Calls, the first 100 minutes every month are free. Thereafter we must - at least for some time - charge a small fee per minute. This applies to nearly every country in Europe, North-America and Australia/New Zeeland. Calls to certain countries with less developed telecommunications infrastructure, and hence very high termination costs - are excluded.

** Applies for calls to fixed lines. Phonecalls to cellular phones are not yet free of charge, but we will strive hard to far cheaper that other telecomoperators also for such calls.