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Useful Terminology of VoIP telephony sphere

If you’re newcomer in IP-telephony sphere, this information would be extremely useful for you. Here you could see all terms concerning VoIP technologies and learn more information about them, its main functions and usage. Without this knowledge it’s hard to orientate in IP-telephony services and features presented by Freezvon. Maybe you cannot remember all of them, but this article will be your memorial service for further using of VoIP telephony with our company. We always want to improve conditions of using products we offer for our subscribers.

We’ve chosen the most important and frequently used notions that will make you a professional in IP-telephony. All technical terminology presented in this section explains principal services, products and processes that Freezvon provides for customers. According to your demands, it’s possible to find really useful piece of information when something is not quite clear or unknown for you. IP-telephony is reliable helper for all your beginnings!

Terminology that could help to understand functioning of IP-telephony

Virtual number that is a number comprising a code of appropriate state and its city or network operator. It has the same quantity of digits like other ordinary numbers and doesn’t have territory limits because it is used without SIM-cards or telephone lines. This number is designated at receiving and making calls, sending and getting SMS, and receiving Fax. It can be multichannel or Toll-free.

DID (Direct Inward Dialing) – that’s a virtual direct number for appropriate country or city, which works without telephone line and can be used out of region where it functions.

Direct Dial In (DDI) – that is a number that works thanks to forwarding of calls, SMS and fax. It can have more trunk lines, with the help of these trunks calls go to other devices available from provider.

Toll free number – that’s telephone number with code of needed city, it’s multichannel one. Calls for subscribers to this number are free. All calls must be paid by its user.

SIP Protocol for data communication for internal network permitting exchange of multimedia files. It is widespread for making calls to foreign countries to mobile numbers of subscribers.

IP-PACKAGES: DATA ARRAY, WHICH TRANSMITTED in INNER NETWORK

TCP (Transmission Control Protocol): Protocol of transported level providing data transmission with previous connection and guarantees vividness of data. When data is lost it make new request, avoid duplicating with two copies of one package.

Real IP-address: that’s IP-address, which can be transmitted via Internet. Unreal address is considered to be designated at local network and without routing to global network.

SIP-number (SIP ID): number from SIP network given by provider, which is used for calls.

SIP-address: full address in SIP network known as SIP_ID@domain. It is used for calls via external SIP-network.

UDP (User Datagram Protocol): one of the easiest protocol permitting to transmit the data (datagram) without installing connection. It cannot provide confirmation and accuracy of data transmission or sending package secondly, that’s why it doesn’t use too much traffic.

RTP (Real-time Transport Protocol): – it transmits media data in real time regime. Data required for renewal of voice or video content in take-up unit is transmitted in title. RTP can also transmit information with description of coding data type.

SIP Proxy server: it gets and handle users’ requests of various types, provides cooperation of SIP appliances. Its name in settings depends on the type of connected appliance. For example, SIP-server or Proxy.

RTP-port: it serves for receiving and removal of media-data. Port of accepting and transmission of packages coincide in one side (if SIP-protocol is used).

SIP-port: this is port of receiving and sending information describing control session. Standard assignment – 5060. Port for receiving and package transmission coincide in one side.

SIP-signalizing: it’s a scheme of special data embracing settings and command of SIP-protocol making method of data transmission. SIP-port is used for transmission of signalizing while call making.

Firewall: internetwork display which main task is the protection of computer or network from external unauthorized danger. Firewall blocks data passing from doubtful IP-addresses.

NAT (Address translator of network): it retransmits internal IP-addresses and port to external addresses and ports. Such translator is used by most providers and individual users in order to solve the problem of IP addresses lack and guarantees safety connected with Internet.

Proxying: particular service (or server) in networks, which gives a possibility to make indirect requests to other services in network. For, example, SIP Proxy server is used for SIP-signalizing and media-packages transmission. Main task of proxy-server is a cooperation of equipment and SIP-apps, which do not have real IP-address and is situated out of NAT translator.

SIP-commands (known also directives and requests): a range of directions, which determine actions for SIP-equipment during connection, authorization, phone toots and other call comprising.

LIST OF MAIN COMMANDS USED FOR MAKING CALLS

INVITE – call user to communication session.

INFO – transmits the data, which do not change state of flowing session.

АСК – testifies about answer accepting for request 200OK for installing connection.

MESSAGE – command of sending instant messages with the help of SIP.

REINVITE – user redirection in flowing session.

BYE – command permitting to finish a session.

CANCEL – disposition about cancellation of requests, which were transmitted recently. If request is handled, the direction could not influence on it

REGISTER – it removes information about address of user registration on server.

In rare cases other commands could be used, for example, PRACK – this request has the same duty as ASK, but it is used for previous or SUBSCRIBE for entering electronic address to a list of active boxes of server.

Answers for SIP requests – answer of proxy-server for accepted request with connection confirmation, information about troubles or other content.

FOLLOWING TYPE OF ANSWERS TO SIP-REQUESTS EXIST

1ХХ – informative answers displaying step of request processing. For example, 180 Ringing.

2ХХ – answers consisting conclusion about successfulness of request processing. This could be an agreement of calling appliance to installation of communication session, confirmation of successful registration etc.

3ХХ – it transmits information to appliance of calling subscriber about location of called one. For example, they can comprise information about user being on other address and indicate it in section “Contact”.

4ХХ – summarizing answers indicating on mistake during request process and user should not make repeated request without previous updating. Such mistakes appeared because of syntax mistakes during writing of request or because requests of this address do not served anymore.

5ХХ – concluding answers, which indicate on impossibility of request processing because of server denial. Reasons of denial are internal mistake on server or absence of necessary functions for request processing.

6ХХ – resulting answers about impossibility of connecting because called user doesn’t exist or doesn’t want to accept the call.

Destination – this is appliance or account in SIP or Skype programs, where readdressing of calls are made from virtual number. Destination can be also cell of landline telephone, SIP or Skype – subscriber determines it on his/her own.

Forwarding – a process pf calls, fax and SMS transmission from virtual number to chosen destination by subscriber.

Terminating lock – appliance designated for installation of connection between SIP networks and landline or mobile networks (PSTN).

Coder – this is a program used for reduction of voice or video data quantity for their further transmission on network.

DTMF (Dual-Tone Multi-Frequency) – multi-frequency signal for telephone number dialing which has two tonality. It is also used in work of interactive systems (for example, IVR menu).

CLI (Caller ID, АОН) – number attached to SIP account for displaying as number with digits during calling process.

In order to check this terminology in practice, you’d better to order any of products, services. If you want to become a great user of VoIP technologies, read the information below.

For receiving appropriate service, it is better to contact our technical experts. They are always waiting for you online in live help chat, in Skype. If you don’t like to talk just write us a letter with described case or problem and we will solve it in several minutes immediately. Our consumers are serious people, so we are going to save their telecommunication abilities in correct and high-qualified state. Rely on us and click here Contact us.

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