The Devices to Perform Voice communication via IP-network

The Devices to Perform Voice communication via IP-network

Voice communication via IP-network can be carried out in various ways:


This option is not an example of IP-telephony, since the voice is transmitted only over a data network, without access to the telephone network. To organize the traffic transfer, the user acquires the necessary equipment and software and pays the provider for the operation of the communication channel. The advantage of this option is the maximum cost savings. The disadvantage is the minimal quality of communication.


In order to organize such connection, certain network devices and interaction mechanisms are necessary.

Voice traffic is transmitted through an IP network, usually by a dedicated expensive site. The devices that organize the interaction are the gateways that are connected to the public telephone network on the one hand, and to the IP network on the other. Voice communication in this mode, in comparison with the “computer-to-computer” option, is more expensive, but its quality is much higher and it is more convenient to use it. In order to use this service, you must call the gateway service provider, enter the code and number of the called party from the telephone and talk the same way as with ordinary telephone communication. All necessary operations for routing the call will be performed by the gateway.

“Computer – phone”

There are more usage opportunities for the corporate users, since the corporate network that serves calls from computers to the gateway is most often used, which is then transmitted over the public telephone network. Corporate solutions using the “computer-to-phone” connection can help save money, and the necessary equipment for this will be discussed below.

So, it is obvious that two basic elements are necessary for building an IP telephony network.

  • The first is a gateway that provides conversion functions between a packet-switched IP network and a public switched telephone network, analog-to-digital conversion, transmission format management, and VoIP call procedures. It is possible to use multiple gateways on the network.
  • The second main element is a gatekeeper, providing a number of functions to control access to the IP network and from the IP network, bandwidth and addressing. In addition, the control device monitors all gateways and terminals, performs the functions of the directory service, monitors user accounts.

The gateway can be supplied as a separate network device or installed on a personal computer. When using the gateway, the VoIP function is transparent to the user using a regular phone or fax machine. Let’s consider in more detail the basic functions of the gateway when transmitting voice over an IP network.

  1. Search function. When the outgoing IP gateway places a telephone call through the IP network, it receives the caller’s number and converts it to the destination gateway IP address, either from the table in the outgoing gateway or from the centralized server data. Viewing a table in an outbound gateway often takes less time than in a centralized server, and reduces the connection time from 4-5 seconds to 1-2 seconds.
  2. Communication function. The outbound gateway establishes a connection to the destination gateway, exchanging information about connection parameters and device compatibility.
  3. Digitization. Analog telephony signals are digitized by the gateway and are usually converted to 64 Kbit / s PCM (pulse code modulation) signal. This function requires the gateway to support a variety of analog telephone interfaces.

In many cases, it is also required to support a digital network with the integration of services and T1 / E1 interfaces. A digital network with service integration and T1 / E1 interfaces operate in PCM format, so no analog-to-digital conversion is required in this case. The digital network with the integration of BRI services has one or two PCM channels, T1 – up to 24 PCM channels and E1 – up to 30 PCM channels. A digital network with the integration of PRI services can have up to 24 or 30 PCM channels.

  1. Demodulation. Since some gateways can receive only a voice or only a facsimile signal, the trunk channels to the voice or fax processing modules must be pre-determined. More sophisticated gateways can handle both types of data, automatically determining whether a digital signal is sound or facsimile, and processing the signal depending on its type. The facsimile signal is demodulated by the signal processor (DSP) back to the digital format 2.4-14.4 Kbit / s, that is, to the original presentation before being sent from the fax machine (the fax machine represents the output signal in analog form). This demodulated signal is then placed in IP packets for transmission to the destination gateway (Figure 2).

The demodulated information is then again converted by the destination gateway to an analog fax signal for delivery to the fax machine.

Fax transmission can be performed using UDP / IP or TCP / IP protocols. UDP / IP, unlike TCP / IP, does not require the correction of errors that occur when transmitting packets.

  1. Compression. After it is determined that the signal is voice, it is usually compressed by the signal processor using one of the compression/decompression methods (CODEC) (Table 1) and placed in IP packets. It is important to ensure a good speech quality and low latency when digitizing the signal.

The audio packet is transmitted as a UDP / IP packet, not TCP / IP, to avoid the relatively large delays that occur when retransmitting TCP / IP packets. If FEC (Direct Error Correction) mode is used, the distorted or missing sound package can be restored based on the data of the previous sound package. If the FEC mechanism is not applied, then the distorted packet is simply discarded and the gateway uses the previous good package. This mechanism works invisibly for the user in the event of a low packet distortion/loss rate (<5%).

The data digitized by CODEC does not contain the IP packet address and control information (“header”) (Figure 3), which usually constitute an additional 7 kbps unless the IP router separately compresses the header, otherwise 2-3 Kbps.

The complexity of the implementation of CODEC determines the power of the required signal processor, measured in millions of operations per second (MIPS), to process the voice signal, excluding the functions of echo cancellation and silence suppression.

  1. Decompression / demodulation. The gateway, executing steps 1-4 described above, at the same time receives packets from other IP gateways and decompresses packets into a form understandable by appropriate analog telephony devices, a digital network with service integration or with T1 / E1 interfaces. The gateway also demodulates the digital facsimile signal to its original form, and then to the appropriate telephone interface.

In addition, the gateway can perform the functions of coordinating the initiators of the call initiator and the caller.

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Posted on: May 15, 2017Ana Nichols