VoIP, also know as Internet Telephony, is the technology that make it possible to have a telephone conversation between two parties over the internet or a dedicated IP network instead of voice transmission lines. Early Voice over IP system requires the dedicated user PC to be eqqupied with the sound capability and a software application.
This software application convert speech signal into digital packet using a specified type of protocol as the transport medium. It also translates into free or very low cost long-distance calling.
The basic scheme of IP telephony starts with Pulse Code Modulation(PCM). The encoded data is transmitted as packets over packet switched network. Onces it reaches the receiver side, the data is decoded and convert back to analog form which travel through the handset. The process of digitization, compression and packetization is handle by Codec - Codec/Decoder.
In VoIP, voice are transported in packets form through the internet and usually an IP phone or software is used between two parties. These devices will convert the analog signal into digital format, followed by compression. The purpose of compression is to save bandwidth and the more it compressed, the lesser quality of the voice. After that, the binary value is input to a packetizer which consists of IP packets and transmit through the internet.
At the receiving end, the packets will input to a depacketizer that will only keep the payload. After that the payload(digital signal) is converted to analog signal using DAC and input to a standard.
In short, to transmit voice over the IP network, the three steps must be followed:
1. Digitization
2. Compression
3. Packetization
In VoIP, voice are transported in packets form through the internet and usually an IP phone or software is used between two parties. These devices will convert the analog signal into digital format, followed by compression. The purpose of compression is to save bandwidth and the more it compressed, the lesser quality of the voice. After that, the binary value is input to a packetizer which consists of IP packets and transmit through the internet.
At the receiving end, the packets will input to a depacketizer that will only keep the payload. After that the payload(digital signal) is converted to analog signal using DAC and input to a standard.
In short, to transmit voice over the IP network, the three steps must be followed:
1. Digitization
2. Compression
3. Packetization
There are three methods of connecting to a VoIP network:
- Using a VoIP telephone
- Using a "normal" telephone with a VoIP adapter
- Using a computer with speakers and a microphone
VoIP telephone calls can be placed either to other VoIP devices, or to normal telephones on the PSTN (Public Switched Telephone Network).
There are two types of PSTN to VoIP services: Direct inward dialing and access numbers.
Direct Inward Dialing(DID) will connect a caller directly to the VoIP user ,while the access numbers need a caller to provide an extension number for the called VoIP user.
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