Voice over IP

June 18th 2008

VoIP technology is using the Internet environment in order to establish audio and/or video communications, similar to those using the classic telephony (PSTN).

How it works

By using the IP protocol, VoIP technology is extremely versatile, the limits being imposed solely by the infrastructure architect and by the operator’s imagination..

Eurovoice VoIP system implemented by Join Telecom is an intelligent solution which enables an excellent quality communication in either corporate and residential domains.

SIP technology (Signaling Protocol) which form the basis of VoIP development is formed essentially of two distinct protocols:

  • SIP protocol, which signals events (call has been initiated, the phone was pulled, there is an ongoing call etc);
  • RTM (Real-Time Media) protocol, which allows for communication transportation (audio and/or video) who relies on its turn to different codecs havin different compression rates.

SIP allowed for the creation of dedicated hardware which includes all the components necessary to transfer voice and video from a microphone/video camera directly to the Internet and vice-versa. Thus, it is possible to use such equipment (generically calld voice gateways) with legacy telephones/videophones, which can still be used on classic PSTN lines. It is the ideal solution for those accustomed to use a regular phone at home.

SIP also allows for the implementation of software solutions, starting from the simplest ones (XLite, Zoiper) up to very complicated, transforming a regular computer into a true telephony switch, with tenths and even hundreds of connections, inter-connections, simultaneous calls, answer machines etc.

SIP can be transported to end-users directly using the home Internet line or, for corporate users, separately through fiber optics, using E1 line multiplexers (it is possible to transport 16 flows of 30 numbers using one optical fiber) in order to use these flows with smart telephony systems (such as smart Alcated systems).

Join Telecom Customisation

Join Telecom wished to have a very versatile system, having in mind that most of the users are corporate ones and flexibility demands are high.

Accordingly:

In order to achieve the highest call qualitym connections are realized in a centralized fashion by the means of metropolitan, high-speed lines. When a number outside Romania is called, the system uses telephony-dedicated external lines, which are not aggomerated with Internet traffic and which are monitored permanently. These dedicated lines are sending voice/video data to the most important Internet nodes in the world (Internet Exchanges), from where it is forwarded to the networks where call destinations users are located.

Happy Jointelecom clients are today on all continents (maybe except Anctarctica).

- call within Eurovoice and Join Telecom are completely free (subscribers already pay for internet access, other taxes would not be justified);

- subscriptions are symbolic for Join Telecom users (0-2 EUR/telephone number), this only covers operational expenses of the system;

- there is no notion of “free” or “included” minutes (any advertisement of this kind hides the real costs in the subscription cost), every call is delailed in the telephony interface, directly on the Internet, with detalied call duration, cost and other information.

- any telephone number can be set up on more than one equipment (or computers) simultaneously. When there is a call, all equipments start to ring. The equipment that realizes the call shuts down all other equipments. Equipments can be located everywhere on the Internet: one at home, one at the office, one on the mobile phone or in an other office abroad. This creates an enhanced portability, very useful for businessmen who do not have the time  to set up complicated solutions and who wish to be available on the same telephone number, wherever they are in the world;

- a telephone number ca be set up in a hunting system, useful for call-centers, where many people make simultaneous calls to the same telephone number and still more than one connection is desired.

- any telephone number can be forwarded to an other telephone number, according to several criteria::

  • unconditionnaly;
  • after a certain number of rings;
  • after a certain number of seconds, if not answered;

- finally, it is possible to iplement short numbers, of 3-4 digits, useable in a small group. The national trend is to have 10 digits for a telephone number, but Join Telecom goes to lower digits for small groups. The equipment set with a short number has a corresponding 10-digits number and hence can also be called from any other network using the regular numbering system.

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Manual de utilizare XLite

June 18th 2008

XLite este un program gratuit, simplu dar eficient, de instalare a unui post telefonic in tehnologie SIP direct pe calculator.

XLite utilizeaza internetul pentru transmiterea vocii catre alt post telefonic (SIP, PSTN, mobil) si sistemul audio al calculatorului pe care este instalat pentru microfon si difuzoare sau casti.

XLite utilizeaza adresa IP a calculatorului pe care este instalat pentru realizarea comunicatiilor de voce, asa incat, daca aveti internet pe calculator si XLite va functiona cu siguranta.

XLite permite stabilirea de conexiuni audio si video simultane, fiind astfel util si pentru videotelefonie.

XLite este limitat la utilizarea unui singur cont SIP. Un upgrade la versiunea comerciala permite si utilizarea a multor conturi SIP pe acelasi calculator.

Instalarea

Instalarea initiala a XLite este directa si nu necesita explicatii suplimentare.

Dupa instalare, XLite cere resetarea sistemului si apoi introducerea informatiilor despre contul SIP:

Adaugarea unui cont SIP in XLite

Apasati butonul Add si completati campurile ca in imaginea de mai jos:

Configurarea unui cont SIP in XLite

Fata de imaginea de mai sus, modificati:

  • CristiAcasa cu numele dumneavoastra, eventual cu locatia unde se afla calculatorul pe care este instalat numarul;
  • 317100177 cu numarul de telefon alocat de catre Join Telecom, omitand primul “0″ din fata numarului de telefon (in ambele locuri unde apare);
  • ….. cu parola care a fost alocata numarului dumneavoastra de telefon SIP;

In rest pastrati totul exact ca in imagine.

In cazul in care informatiile introduse au fost corecte, dupa apasarea butonului OK si inchiderea paginii cu contul SIP, veti vedea o fereastra similara cu cea de mai jos:

Cont SIP inregistrat in centrala

In cazul in care informatiile nu sunt corecte, nu veti vedea informatia “Ready” ci un mesaj de eroare.

Repetati introducerea informatiilor sau contactati un specialist Join Telecom pentru rezolvarea problemei.

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