Possibilities of ip telephony. Installation and configuration of IP (SIP, VoIP) telephony

It is difficult to imagine a successful company without high-quality telephone communication. Any business involves constant communication with customers, partners and subordinates. The cost of installing telephones for a large enterprise is quite tangible, because you have to buy equipment, pay for connection and maintenance of lines. You can easily reduce costs if you adopt modern technologies and install IP telephony!

Organization and installation of IP telephony

Today, IP telephony is the most important telecommunication technology, which allows not only to save money, but also to raise such key characteristics of communication systems as reliability and fault tolerance to a new level. It is able to provide high quality communication and becomes available even for organizations with low incomes. The organization of IP telephony allows you to conduct telephone conversations, send / receive faxes via the Internet. The use of broadband access provides simultaneous data transmission over multiple channels in real time, which fully satisfies the needs of corporate communications.

* These prices on the site are preliminary. Call us or leave a request on the website to contact and clarify the final cost of the work.

Procedure for configuring IP telephony

Setting up IP telephony is a simple and quick task, especially for specialists with extensive experience in such work.

To do so, you will need:

• a dedicated Internet channel that meets certain requirements necessary to ensure high quality telephone communications;
• telephone sets or softphones with the necessary set of functions;
• office PBX, which can be hardware, open source software and so-called virtual;
• multi-channel numbers for incoming telephone communication for organization in the office of incoming communication.

Alefo offers a service for setting up IP telephony, which will allow you to:

1. Save your money, especially if you have to regularly make calls to another city or country.
2. Obtain absolute confidentiality of communication, which is ensured thanks to the relevant protocols of the operators' obligations.
3. Increase the number of lines at no additional cost.
4. High quality connection, which does not depend on the distance.
5. Get access to any statistical data, determine the scope of costs and set restrictions on connection with certain subscribers.

Setting up SIP telephony

Setting up SIP telephony is carried out by connecting the SIP phone to the Internet and registering on the server of the SIP provider. After that, the device is ready to perform all the functions laid down in it by the manufacturer. In addition to the usual voice communication options, many SIP devices have audio communication capabilities and support video calls. SIP phones are also equipped with codecs that transmit the voices of interlocutors without distortion and freezes.

It is also worth noting that such telephones display more detailed information than conventional ones. At the end of the conversation, a notification about the balance is displayed on the display of the SIP phone, for internal calls the name and surname of the interlocutor are displayed, and additional light indication helps to track the lines of other subscribers, as well as intercept intended calls for them (for example, if a colleague is not in place).

Setting up VoIP telephony

VoIP-phones appeared on the Russian market relatively recently, however, despite this, they have already managed to earn authority and recognition among domestic consumers. This is largely due to the good workmanship of the devices themselves, the reliability of their operation and ease of setup. And most importantly - all this at a reasonable price.

Setting up VoIP telephony allows you to create a single network infrastructure that helps reduce costs by combining enterprise departments and reducing the number of employees. However, the advantages of this telecommunication system are not limited to this.

Other benefits also include:

• No hardware restrictions. This makes it possible to install and connect as many VoIP phones to one number as you may need. If necessary, you can increase the number of numbers and lines, which is especially important for dynamically developing organizations.
• Versatility. VoIP phones allow you to create a single telephone network between company offices located in different locations. This eliminates the need to purchase multiple PBXs.
• Mobility. When the office moves to another location, the number and the entire structure remain unchanged, which means that you do not have to notify clients about the change of number.
• Wide functionality. VoIP phones help you stay in touch with the user, even if he called after hours. His call will be handled by the virtual secretary.

A modern communication standard that involves the transmission of voice and messages over the Internet. It is independent of analogue lines and mobile channels. To organize corporate communications, you only need an Internet connection and IP-telephony from UIS.

Benefits of IP telephony

1. Minimum expenses:

you can connect additional numbers without laying a cable;

no need to buy automatic telephone exchanges and other expensive equipment;

SIP telephony is supported by the provider, which reduces the load on the client's IT service.

2. Scalability of IP telephony:

when opening new divisions, the same contacts of the company are retained;

to expand the call-center does not need to buy additional equipment;

when holding promotions and other events, special telephone numbers are connected in 10-15 minutes.

3. Efficiency:

It is possible to organize communication in the office in one day. Setting up external communications takes a little more time;

when the company moves, SIP lines continue to function, so business processes are not interrupted even for a minute;

automatic forwarding of calls from the company's unified numbers directly to the necessary departments and employees saves a lot of time, which improves the attitude of customers towards the company.

4. Extended functionality of the Virtual PBX for processing customer calls:

optimization of call distribution and communication quality control tools allow you to increase the conversion of calls into sales;

dynamic call tracking, event notifications and automatic callbacks on passes provide a lower cost of leads;

bringing call processing schemes in line with business processes - for example, integrating telephony with CRM - leads to an optimization of the sales process as a whole.

How IP telephony works

VoIP communication is different from classic analog telephony. It converts a person's voice into digital packets, not electrical signals. And this gives businesses the flexibility to handle requests.

The company receives multichannel IP numbers from the provider. A virtual PBX is capable of receiving up to 100 calls to such a number at the same time. She chooses the recipient of the call, taking into account various factors: from the location of the client to previous communication experience. The simplest example: if a call comes from an unknown number, the IP equipment will forward it to the sales manager. If the call comes from a partner of the company, then IP-telephony in the office will connect it with a specialist "leading" a certain direction.

By connecting IP telephony, you can listen to all conversations of employees and identify errors. The service quality-based reward system increases conversion and improves the company's business reputation. And in cooperation with a remote call center (outsourced), control of office telephony will help to establish a fair cost of services.

Other possibilities of IP telephony for business: :

connection of electronic tools: feedback systems, automatic call forms, virtual chats, bots and others;

compatibility of IP technologies with corporate planning, accounting and management systems, including CRM;

easy integration of work with calls into ready-made solutions and individually developed programs.

A multi-channel virtual number also allows you to add other participants to the conversation, delimiting their communication. It looks complicated, but such a SIP telephony feature is due to the real needs of companies working with customer calls. If problems arise in a conversation with a client, the process is controlled by the head of the department or the trainer. Only the manager of the company will hear his comments and recommendations. SIP lines allow you to control communication in real time, preventing conflicts and increasing sales efficiency.

Equipment for IP telephony

Corporate telephony may include different types of equipment. For digital calls, an IP phone is usually used. It is compatible with various routing systems and network architectures, making it easy to connect. Alternatives to it are computers, smartphones and workstations with special applications for calls. IP telephony easily connects to popular messengers including Skype, WhatsApp and Viber, making it convenient for small businesses.

Instead of a digital IP phone, you can also use a classic analog device. It is connected through an IP gateway that converts electrical signals into digital packets. This solution simplifies the upgrade of existing communications.

Sometimes an IP channel serves as an intermediary between two communication systems. A good example is call forwarding to an employee who works on the road or is on a business trip. In this case, the conversation via IP is not sent to a stationary device, but to a mobile phone. At the same time, the client is not even aware of the complex path that the signal overcomes. The connection is established in a fraction of a second.

Favorable cost of communications from UIS

Our company is a reliable provider of IP-telephony. We have been working in the field of digital communications for over 18 years and are in the TOP-3 of the Russian virtual PBX market (according to IKSMEDIA). The UIS network stability rate of 99.97% makes IP data transmission fast and high quality. In a highly competitive environment, instant connection with the client and clear speech transmission is the key to a successful sale.

Using the latest software and hardware systems allows us to set up telephony in the office in less than a day. We cooperate with various enterprises: from small start-ups to financial institutions and industrial complexes and online stores of an all-Russian scale.

By connecting IP-telephony from UIS, you get a convenient and affordable solution for business. The cost of connecting a virtual PBX starts from 590 rubles per month.

IP telephony is called voice communication, which is carried out over networks designed for data transmission, more often over IP networks (the term IP stands for "Internet Protocol"). At present, communication using IP-telephony has begun to replace traditional telephone networks due to its low cost of calls, ease of deployment, high quality connection and communication, their relative security, and ease of configuration. In this article, the presentation of the material will be conducted from the data link and physical layers to the data layers, adhering to the principles of the OSI model (stands for the Open System Interconnection basic reference model).

The principle of operation of IP-telephony

When calling between IP telephony subscribers, the voice signals broadcast between them are converted into compressed data packets. This process will be covered in more detail in later chapters on PCM and codecs. After the data packets are compressed, they are sent over packet-switched IP networks. As soon as the data packets reach their destination, they are again converted into voice signals. All these processes are implemented through the use of a large number of auxiliary protocols, some of which will be discussed below.

Considering data transfer protocols in this context, we can call them a kind of language that allows subscribers to establish voice communication with each other and ensure the transfer of the data necessary for this between communication points.

Distinctive features of IP-telephony from traditional communication

The establishment of a traditional telephony connection is carried out via the telephone exchange and is performed only for the purpose of making a conversation. Signals between subscribers are transmitted through a dedicated connection over telephone lines. When using IP telephony, data packets pass through a local or global network, and they also have a specific address, on the basis of which they are transmitted over it. In this case, one cannot do without the use of IP-addressing with its features such as routing.

A more profitable solution in terms of the cost of conducting calls for the operator and the subscriber will be IP-telephony due to its following features:
- to access the global network today, almost every computer user can do with minimal costs or even do without them;
— making calls over the local network is possible when using an internal server without the help of an external PBX;
- if there is excess performance in traditional telephone networks, then in IP telephony, thanks to packet compression technology, it is possible to fully use the capacity of telephone lines.

With the above advantages, IP-telephony also allows you to improve the quality of communication due to three main factors:
— Owners of private networks have full control over hardware and software, which allows them to change and configure network parameters such as the number of line subscribers and bandwidth, as a result of which the delay is reduced;
- the constant improvement of telephone servers, together with the improvement of their operation algorithms, makes communication more resistant to delays and other problems in IP networks;
- the development of packet-switched networks and the annual introduction of new protocols and technologies that improve the quality of communication sessions (an example is the RSVP protocol, designed to reserve bandwidth);
- IP-telephony elegantly solves the problem of a busy line - the implementation of call forwarding or transfer to standby mode is performed by entering several commands into the configuration file on the PBX.

1. Physical layer

The physical layer of data transmission is characterized by the transmission of streams of bits through the corresponding interface over the physical medium. And in this, IP-telephony uses almost completely the existing infrastructure of network connections. To transmit information, as a rule, a fifth category twisted pair (UTP5), coaxial cable or multimode optical fiber is used. Such borrowing implements the principle of convergence of network telecommunications in full.

PoE

In the context of considering the physical layer of data transmission, it will be interesting to consider the PoE technology (stands for "Power Over Ethernet"), which operates according to the IEEE802.3 af-2003 standards, as well as IEEE 802.3 at-2009. The essence of the technology lies in its ability to provide power to the device using a standard twisted pair cable. Modern IP phones, such as Cisco's Unified IP Phones 7900 Series, support PoE. According to the 2009 standard, IP telephony devices can be powered up to a maximum of 25.5 watts.

Two of the four twisted pairs of the 100Base-TX cable are used to supply power to the device, however, manufacturers can use all pairs by increasing the power transmission power to 51 watts. PoE technology will not require modifications to cable networks already in operation, as well as Cat 5 cables themselves.

To determine the device's ability to be powered (which is indicated by the marking PD - powereddevice), a voltage of 1.8 to 10 V is applied to its cable. This way, the input resistance of the connected device can be calculated. When determining the resistance within 19-26.5 kOhm, the second operation is performed, otherwise the test will continue with an interval of 2 ms or more. The essence of the second operation is to find the power range of the tested device. The search is carried out by applying an increasing voltage to the input, followed by a measurement in the current line. After that, a voltage of 48 V is applied to the power line. In the process of powering the device, constant monitoring of power supply overloads is carried out.

2. Data Link Layer

Specification Terms IEEE 802 divide the link layer into 2 sublayers:
1 – MAC(stands for "Media Access Control"), which provides interaction with the physical level;
2 – LLC(stands for Logic Link Control), which serves the network layer.

The data link layer uses switches designed to provide interconnection between several nodes in a network of computers, as well as to distribute frames between hosts based on physical addressing (MAC).

It is worth writing about virtual local area networks (English Virtual Local Area Network - VLAN). VLAN technology allows you to create a logical network topology, regardless of the physical characteristics of the latter. This is achieved using traffic tagging, which can be found in more detail in the description of the IEEE 802.1Q standard.

Voice VLAN technology is widely used to isolate voice traffic generated by IP phones from other data. It is advisable to use the capabilities of this technology for the following reasons:
— Improving the quality of data transmission. It is implemented by the VLAN mechanism to set an increased priority for voice data packets, as a result of which the quality of communication increases.
- Security. By creating a separate voice VLAN, you can reduce the likelihood of intercepting and analyzing voice packets by unauthorized persons.

3. Network layer (English Network Layer)

Given that the network layer is designed to implement flow routing, it is customary to consider routers as its main devices. These devices determine the path that data takes to reach a destination that has a specific IP address.

IP (Internet Protocol) is used as the main routed protocol. On its basis, both IP-telephony and the World Wide Web function. In addition to the main one, there are many dynamic protocols for routing, the most popular of which can be called the internal protocol OSPF (Open Shortest Path First).

Along with conventional gateways, there are also special VoIP gateways (eng. Voice Over IPGateway) that provide connection to the IP network for ordinary phones. They usually have a built-in router that keeps track of traffic, authorizes users, automatically distributes IP addresses, and manages bandwidth.

Some of the standard features of VoIP gateways are:
— support for facsimile communication;
- support for SIP (Session Initiation Protocol) and H.323 protocols;
- support for voice mail;
- functions to improve security (authorization, creation of lists of users with access).

To avoid the delays that occur when transmitting data over IP, it is necessary to use additional tools along with it, for example, queuing protocols that eliminate the problem of competing voice data with regular ones. To achieve this goal, routers use weighted queuing based on CBWFQ (Class-Bassed Weighted Queuing) or low-latency LLQ (Low-Latencyqueuing) queuing. Marking schemes will also be needed to prioritize voice data as the most important in the overall transmission flow.

4. Transport Layer

The transport layer provides:

• end-to-end connection;
• segmentation of application data from the top layer;
• data reliability.

The transport layer uses the following protocols as the main ones:

• UDP (User Datagram Protocol);
• TCP (Transmission Control Protocol);
• RTP (Real-time Transport Protocol).

IP telephony directly uses the RTP and UDP protocols, which mainly differ from TCP in that they do not provide reliability in data delivery. For IP telephony, this feature is more acceptable than using TCP with its delivery control, because telephone communication is very dependent on data transfer delays, but packet loss is not critical for it.

UDP protocol

UDP is based on the IP network protocol, and its functions are limited to providing transport services to application processes. The main difference between the UDP and TCP protocols is the provision of non-guaranteed delivery first (when sending and after receiving data, UDP does not request any confirmations). When sending data via the UDP protocol, it is not necessary to establish a logical connection between the source and the destination.

RTP protocol

Although RTP is considered to be a transport protocol, it usually works over UDP. The RTP capabilities implement work with timestamps, recognition of the type of passing traffic, numbering of the sequence of packets and control of their transmission.

The main purpose of the RTP protocol is to assign time stamps to all outgoing packets, which are subsequently processed by the receiving side. Thanks to this, it becomes possible to receive information in the order in which it was sent, the influence of uneven time intervals of packets in the network is reduced, and synchronization between video and audio data is restored.

5. Data Layers

The last three layers of the OSI model can be considered together. It is permissible to combine them in this description due to the fact that the processes occurring in them are closely related to each other, and it would be less logical to describe them separately.

H.323

The H.323 protocol stack was developed back in 1996. This standard contains descriptions of network services, terminal devices and equipment designed to implement video and audio communications in networks with the presence of packet switching (the Internet). Any H.323 device must support the exchange of voice information.

According to H.323 recommendations, the equipment it regulates must contain:
— standard encodings of analog data;
- platform independence;
— flexibility and compatibility;
- the ability to control the bandwidth.
In this context, it is worth noting an important fact: the recommendations do not specify the transport protocol, network interface, or physical transmission medium. This ambiguity allows all devices that support the H.323 standard to work with any network available today and working with packet switching.

According to the H.323 standard, the main 4 components for VoIP connections are:

SIP protocol (stands for Session Initiation Protocol)

The SIP signaling protocol is designed to organize communication sessions, modify and terminate them. Despite the fact that SIP does not depend on transport technologies, it is desirable to use UDP when establishing it. At the same time, it is recommended to use RTP for the transmission of video and voice information, and the possibility of using other protocols is not excluded.

The SIP protocol defines 2 types of signaling messages, referred to as request and response. It also implemented the work of six procedures:
- INVITE (invitation) - serves to initialize a new connection, that is, invites the user to a communication session; this procedure may have additional parameters used for negotiation;
- BYE (disconnection) - serves to terminate the connection created earlier by two users;
— OPTIONS (options) – the procedure is used if it is necessary to transfer information about supported characteristics (the transfer can be sent both to another user's agent and through an intermediary SIP server);
- ACK (acknowledgment) - the procedure is used to be able to acknowledge the receipt of a message or to receive a positive response to the sent INVITE command;
- CANCEL (cancellation) - used to stop the search for the user;
- REGISTER (registration) - using the procedure, you can transfer information about the user's location to the SIP server, which in turn can broadcast the received data to the address server (Location Server).

Codecs

An audio codec is an algorithm or program that compresses or decompresses audio type data, thus reducing the bandwidth requirements for data transmission channels. Today, in IP telephony, the G.729 and G.711 codecs are more common than others, which perform data conversion and compression according to the A (alaw) and u (ulaw) laws.

G.729

The G.729 codec compresses the received file with the loss of its data. The main idea underlying the codec is to transmit not the digitized signal itself, but only its parameters (spectral characteristic, number of transitions through the zero mark), which are sufficient for their subsequent synthesis by the receiving party. After decompressing an audio file, its main characteristics (timbre, amplitude, and others) are not lost.

The G.729 codec is designed for a channel bandwidth of 8 kbps. The length of the frame it processes is 10 ms, and the sampling rate is 8 kHz. Each processed frame is redefined into a mathematical model in the form of a code, which is transmitted to the channel.

Using G.729 encoding causes a delay of 15 ms, with 5 of them being spent filling the pre-buffer. It is also worth noting that this codec is demanding on processor resources.

G.711 codec

The G.711 voice codec does not imply data compression, except for companding - reducing effects in a channel that has a limited dynamic range. The method is based on the principle of reducing the signal quantization levels of areas with high volume, while the sound quality does not decrease. There are two companding schemes commonly used in telephony, called alaw and ulaw.

The signal flow in this codec is 64 kbps. It transmits 8000 frames per second, each with 8 bits. In a subjective comparison, the voice quality after processing it with this codec is better than after G.729.

Alaw and ulaw

A-law (alaw) is a compression algorithm that compresses audio data, but removes some information from it. It is used mainly in Russia and Europe. U-law (ulaw) as well as A-law is designed for audio data compression, in which it loses part of the data from the file. U-law is used primarily in North America and Japan.

Pulse code modulation PCM (eng. Pulse Code Modulation)

Pulse code modulation can be described as the transmission of a continuous function, which has the form of successive pulses.

To obtain a modulated signal on the input communication channel, it is necessary to measure the carrier signal using an ADC after a certain period of time. In this case, the sampling rate (described as the number of digitized values ​​per second of time) must be greater than or equal to twice the maximum frequency from the analog signal spectrum. The resulting values ​​are then rounded up to the level specified in advance in the program. It is worth noting that all levels must be a multiple of a power of two. After determining the number of levels, it becomes possible to determine the number of bits that encode the signal.

During demodulation, a sequence of 0s and 1s is acquired by a demodulator as a copy in the form of pulses. In this case, the quantization level of the demodulator is equal to the quantization level of the modulator. Further, with the help of the DAC, the signal is restored, and the smoothing filter removes the last inaccuracies.

Modern telephony must have at least one hundred levels of quantization, in other words, the minimum number of bits for encoding a signal must be at least seven.

IP telephony: quality of service

Networks built on the basis of TCP / IP protocols are not able to provide high quality service to telephone subscribers, as they introduce unacceptable delays into data transmission. The TCP protocol guarantees reliable delivery of information, while its transfer by default can be carried out with various delays. The UDP protocol is characterized by minimizing such delays, while the guarantee of correct delivery is not provided.

As you know, the quality factor of the transmission of speech signals is very dependent on the quality of their transmission. Networks that cannot implement mechanisms that guarantee the desired quality do not meet the requirements of IP telephony users.

Quality of service can be expressed in terms of such basic indicators as transmission delay and network throughput. The delay is defined as the time elapsed from the moment a packet was sent to the moment it was received. In addition to the main ones, additional characteristics can be distinguished, such as network reliability and availability. They can be evaluated after a long time based on the results of the control of the service level or by the utilization factor.

To improve the quality of communication, the following mechanisms can be used:
— communication channel resources are reserved for the duration of the entire connection;
- rerouting, with the help of which data is delivered using backup routes if the main channel is overloaded;
- traffic prioritization, which allows you to mark the importance of packets and further serve them in accordance with these markings.

As already mentioned, voice data traffic is very dependent on transmission delays. The maximum delay value must be less than 400 ms, which includes the duration of packet processing at the receiving stations. Delays can be divided into two main types:
1) Delay introduced by the transmission network. It can be reduced by improving the network infrastructure, namely, by using high-speed channels and reducing routers.
2) Information delay in the terminal equipment or when it is encoded in voice gateways. It can be reduced by improving the voice conversion and processing algorithms in use.

Jitter

A phenomenon characteristic of IP telephony is a random delay in the propagation of a packet, called jitter. Jitter can be caused by three factors:

• thermal noise;
• high delay in signal propagation;
• limited bandwidth or incorrect operation of operated network devices.

Often, to combat jitter, a jitter buffer is used to combat jitter, which stores the number of packets specified by the program. The buffer length is usually dynamically tuned by tuning during the entire connection session. Heuristic algorithms can be used to find the best length.

jitter buffer

To compensate for uneven packet arrival rates, the receiving side creates a temporary storage for packets called a jitter buffer. The task of this buffer is to collect incoming packets in the correct order, corresponding to timestamps, and issue them to the codec at the correct intervals and order.

The size of the jitter buffer can be specified in the settings forcibly or calculated during sessions. This decision is based on the impossibility of calculating the optimal value of the buffer size, since a large value of it will cause an increase in transport delay, and a small value can cause packet loss if delays in the IP network increase unexpectedly.

The size of the jitter buffer causes controversy between users and IP telephony providers. With a small buffer size on the user's side, not all packets sent by the provider can reach the user's codec, while the provider will state the delivery of all packets without exception. From a practical point of view, more than 1% of lost data will cause discomfort during a conversation, and at 2% it will already be difficult. A loss value of 4% can make conversation nearly impossible.

The jitter buffer size is made larger than the network transit time jitter value. If for a dozen packets the transit time fluctuates between 5 and 10 ms, then the buffer should be up to 8 ms in size in order not to lose a single packet. If the buffer has a size of 12 ms, then it will also be able to re-request lost packets.

Software and hardware for the deployment and use of the telephone network

Asterisk

The Asterisk software PBX is capable of switching VoIP calls between traditional telephone network subscribers and IP phone users.

Asterisk PBX supports UNIStim, H.323, IAX, SIP, Skinny protocols. Among the codecs supported are: G.222, G.223, G.729, G711 (alaw and ulaw), LPC-10, iLBC, Speex, GSM.

The Asterisk software is open to third parties, dynamically evolving, and can be installed without the hassle of licensing. This feature makes software PBX a profitable solution for medium and small businesses. The number of subscribers served by it can be up to 2,000, and only the capacity of the server serves as a limitation.

The second advantage of Asterisk is the possibility of its flexible configuration. The functions necessary for full-fledged work are already implemented in it, and auxiliary ones can be added independently without tangible monetary and time costs. The principle of the program contributes to this: one program module is used for one task.

If we compare Asterisk with the products of such vendors as Avaya or Cisco, then it also attracts with the cost of its deployment. All costs for it are reduced only to the purchase of telephone sets, as well as a server that could cope with the necessary load on the network. The program itself is free.

Cisco Call Manager

The CallManager hardware and software complex is primarily designed for networks with up to 30,000 subscribers. The complex is able to provide reliable operation and allows you to configure many settings, such as voice menu or call forwarding. Lightweight express version of the complex is intended for small offices.

The advantage of Cisco CallManager is Cisco's renowned technical support. With an appropriate level of service contract, any problem related to questions about setting up a hardware or software environment or equipment breakdown is solved almost instantly. This quality of the CallManager complex will come in handy for those companies that are willing to pay considerable expenses, receiving the highest quality of customer service.

Avaya IP Office

The IP Office Appliance Solution is a good choice for a medium-sized telephone network. The limit on the number of subscribers here is connected not only with the capacity of the server, but also with the purchased licenses. Licenses are imposed on almost every detail of the complex, such as the applications used and expansion boards. The equipment is configured through various programs, the most popular of which, and, moreover, easy to use is IP OfficeManager from the same Avaya company. You can also manage IP Office settings through the console using the Avaya Terminal Emulator tool.

Avaya also produces other products besides IP Office, and having merged with another well-known manufacturer Nortel in 2009, it has become a recognized leader among companies selling equipment for IP telephony.

Do you want to know more about ? Contact ITERANET — we have been implementing complex communication projects for more than 15 years, we are engaged in the infrastructure of facilities. The list of our services includes a list of high-tech solutions from more than 100 items.