Frequently Asked Questions

What is VoIP Phone?

VoIP phone is a SIP phone or phone program. It allows users to call any telephony program, mobile phone or landline using IP (VoIP). Thus, the sound is carried over the internet instead of the traditional PSTN system. It can also be a VoIP phone, a simple software-based telephony program or a hardware tool such as a regular phone. Some common VoIP phone features are: Caller ID, park the call, transfer the call, and hold the call.

 

 

What is DID-Direct Internal Call?

DID-Direct Internal Call (also known as DDI in Europe) is a feature that phone companies offer to their subscribers. This feature is used with PBX systems owned by subscribers. Telephone companies assign their customers a number field associated with one or more telephone lines through the DID system. The purpose of this is to enable a company to give special phone numbers to all employees without allocating a separate phone line to each employee. In this way, telephone traffic can be divided and managed more easily. To take advantage of the DID feature, you must purchase an ISDN or digital line and ask the telephone company to assign you a number field. You will then need DID compatible BRI, E1 or T1 cards or gateway equipment for use in your own business.

 

What is a STUN server?

A STUN (Simple Migration of User Datagram Protocol [UDP] over Network Address translators [NATs]) server allows NAT clients (such as computers behind a firewall) to forward a phone call to a VOIP service provider outside the local network. The STUN server allows clients to find their IP addresses, the type of NAT they are behind, and the side Internet port associated with a particular local port by NAT. This information is used to establish UDP communication between the client and the VOIP service provider, thereby initiating a phone call. The STUN protocol is defined in RFC 3489. The STUN server is communicated on the UDP 3478 port, but the server will tell the clients to try the other IP and port number (STUN servers have two IP addresses).

 

What is a SIP server?

SIP server is the main part of IP PBX. A SIP server enables the creation of all SIP calls on the network. SIP server is also known as SIP Proxy or Registrar Server.

 

What is SIP (Session Initiation Protocol)?

SIP, the Session Initiation Protocol, is an IP telephony signaling protocol used to create, set up and end VOIP phone calls. SIP was developed by IETF (Internet Engineering Task Force) and published as RFC 3261. SIP identifies the communication required to create a phone call. SIP has become very popular in the VOIP world. This protocol, similar to the HTTP protocol, is text-based, fairly open and flexible. Therefore, it has largely replaced the H323 standard.

 

What is RTP - Real Time Transport Protocol?

 

RTP - An abbreviation for Real Time Transport Protocol; Refers to the standard packet format used for transmitting audio and video over the Internet. RTP is defined in RFC 1889. This protocol was developed by the Audio Video Transmission group of employees and was first introduced to the general public in 1996. RTP and RTCP are highly interconnected; the data itself is transmitted by RTP, and RTCP is used to provide feedback on the quality of this service.

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