Sip internet telephony. What is IP telephony. IP telephony: setup, providers, tariffs and reviews. About the cost of IP telephony

16.10.2023

VoIP technology implements tasks and solutions that would be more difficult or more expensive to implement using PSTN technology.

Ability to transfer more than one phone call within a high-speed telephone connection. Therefore, VoIP technology is used as an easy way to add an additional phone line to your home or office.

Properties such as

conference,
call forwarding,
automatic redialing,
calling number identification,

are provided for free or almost free, while traditional telecommunications companies usually charge for it.

Secure calls, with a standardized protocol (such as SRTP). Most of the difficulties to enable secure telephone connections over traditional telephone lines, such as signal digitization, digital signal transmission, have already been resolved within VoIP technology. It is only necessary to encrypt the signal and identify it for the existing data stream.

Location independent. You only need an Internet connection to connect to a VoIP provider. For example, call center operators using VoIP phones can work from any office where there is an efficient, fast and stable Internet connection.

Integration with others over the Internet is available, including video calling, messaging and data exchange during a call, audio conferencing, managing your address book, and getting information about whether other subscribers are available for a call.

Additional telephony features - such as call routing, pop-ups, alternative GSM roaming and IVR implementation - are easier and cheaper to implement and integrate. The fact that a telephone call is on the same data network as the user's personal computer opens the door to many new possibilities.

Additionally: the ability to connect direct numbers in any country in the world (DID).

Mobile numbers

Encoding introduces an additional delay of about 15-45 ms, which arises for the following reasons:

using a buffer to accumulate the signal and take into account the statistics of subsequent samples (algorithmic delay);
mathematical transformations performed on a speech signal require processor time (computational latency).

A similar delay occurs when decoding speech on the other side.

Codec latency must be taken into account when calculating end-to-end delays (see ). In addition, complex encoding/decoding algorithms require more computational resources of the system.

An analysis of the quality of speech data transmission over the Internet carried out in various research groups shows that the main source of distortion, reduction in the quality and intelligibility of synthesized speech is an interruption in the flow of speech data caused by:

loss of packets during transmission over the communication network;
exceeding the permissible delivery time of a packet with voice data.

This requires solving the problem of optimizing network delays and creating speech compression algorithms that are resistant to packet losses (recovering lost packets).

Codecs

The voice compression algorithms used when transmitting over an IP network are quite diverse. Some practically do not compress the voice, leaving it at the level of pulse-code modulation (that is, 64 kilobits per second), other codecs allow you to compress the digital voice stream by 8 times or more due to effective encoding algorithms. There are many good free codecs, the use of which does not require licensing. For others, it is necessary to achieve the appropriate licensing certification between the hardware (software) manufacturer and the authors of the compression method.

Codec	Transmission speed, kbit/s	Algorithmic delay, milliseconds	Occupied stream, kbit/s
Codec	Transmission speed, kbit/s	Algorithmic delay, milliseconds	IP packets	Ethernet frames
G.711	160	64	20	64,8	80
G.723.1 (6.3)	24	6,3	37,5	6,9	17,1
G.723.1 (5.3)	20	5,3	37,5	5,9	16
G.726-32	160	32	20	32,8	42,7
G.726-24	160	24	20	24,8	34,7
G.726-16	160	16	20	16,8	26,7
G.729 (8)	20	8	25	8,8	18,7
G.729 (6.4)	16	6,4	25	7,2	17,1

Network latency optimization

The main advantages of IP telephony are reduced bandwidth requirements, which is ensured by taking into account the statistical characteristics of voice traffic:

blocking the transmission of pauses (dialogue, syllabic, semantic, etc.), which can account for up to 40-50% of the time the transmission channel is occupied (VAD);
high redundancy of the speech signal and its compression (without loss of quality during restoration) to the level of 20-40% of the original signal (see audio codec).

At the same time, packet delays in the network are critical for VoIP, although the technology has some tolerance (resilience) to the loss of individual packets. Thus, the loss of up to 5% of packets does not lead to a deterioration in speech intelligibility.

When transmitting telephone traffic using VoIP technology, the strict requirements of the ISO 9000 standard for quality of services must be taken into account, characterizing:

quality of connection establishment, determined mainly by the speed of connection establishment,
connection quality, measured by end-to-end (user-perceived) delays and perceived speech quality.

The general acceptable delay according to the standard is no more than 250 milliseconds. The reasons for delays in the transmission of voice data over an IP network are largely related to the characteristics of packet transport. The TCP protocol provides packet delivery control, but is quite slow and therefore not used for voice transmission. UDP sends packets quickly, but recovery of lost data is not guaranteed, resulting in lost parts of the conversation when the audio is reconstructed (reconverted). Jitter (deviations in the period of packet arrival and reception), which appears when transmitting through a large number of nodes in a loaded IP network, brings considerable problems. Insufficiently high network bandwidth (for example, with simultaneous load by several users) seriously affects not only delays (that is, an increase in jitter), but also leads to large packet losses

To solve such problems, a set of measures is proposed:

use of algorithmic restoration of lost parts of the voice (averaging over neighboring data)
prioritizing traffic during transport in one network by marking IP packets in the Type of Service field
using a variable jitter buffer of the required length, which allows packets to be accumulated and issued again at normal intervals
disabling media data proxying at a network bottleneck, that is, achieving direct speech exchange between the calling and called subscriber node through intermediate servers only at the stage of establishing and ending the call
use of codecs with lower algorithmic latency (to reduce the load on the processor that implements the ADC and DAC)

Connection security

Most VoIP customers do not yet support cryptographic encryption, even though having a secure phone connection is much easier to implement with VoIP technology than with traditional phone lines. As a result, using a traffic analyzer, it is relatively easy to monitor VoIP calls, and with some tricks, even change their content.

An intruder using a network packet sniffer has the ability to intercept VoIP calls if the user is not within a secure VPN. This security vulnerability can lead to denial of service attacks on the user or someone whose number belongs to the same network. These denials of service can completely destroy a telephone network, overwhelming it with junk traffic, creating a constant busy signal and increasing the number of caller disconnections.

However, this problem also applies to traditional telephony, since there are no absolutely secure communication methods.

Consumers can secure their network by limiting access to the data VLAN, hiding their voice data network from users. If a consumer maintains a secure and properly configured gateway gateway with controlled access, it will protect itself from most hacker attacks. There are several open source solutions that analyze VoIP conversation traffic. Low levels of security are provided by proprietary audio codecs that cannot be found on open source lists, but this “security through obscurity” has not proven to be effective in other areas. Some vendors also use compression to make it more difficult to intercept information. It is believed that true network security requires full cryptographic encryption and cryptographic authentication, which are not available to the general consumer. However, in some respects, IP telephony outperforms traditional telephony in terms of security.

The current security standard SRTP and the new ZRTP protocol are available on some models of IP phones (Cisco, SNOM), analog telephone adapters (Analog Telephone Adapters, ATAs), gateways, as well as on various softphones. You can use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but it does use an encryption system that is transparent to the Skype provider.

The Voice VPN solution (which is a combination of VoIP and Virtual Private Network technology) provides the ability to create a secure voice connection for VoIP networks within a company by applying IPSec encryption to the digitized voice data stream.

It is also possible to perform multi-level encryption and complete anonymization of all VoIP traffic (voice, video, service information, etc.) using the I2P network, a router program for working with which can be installed on a PC, smartphone, netbook, laptop, etc. d. This network is a completely decentralized, anonymous data transmission medium, where each data packet is subject to four-level encryption using various encryption algorithms with maximum key sizes. An I2P network uses data tunneling, where incoming and outgoing traffic goes through different tunnels, each encrypted with different keys, and the tunnels are periodically rebuilt with changing encryption keys. All this leads to the inability to listen and analyze the passing stream by a third party. At the same time, tunneling and encryption do not affect streaming, since a library specially created for streaming services is used, so the data arrives strictly in the given order, without losses or duplications.

Caller ID

Caller ID support may vary between providers, although most VoIP providers now offer a named Caller ID service for outgoing calls. When a call is made to a local network number from a VoIP provider, the caller ID service is not supported.

In some cases, VoIP providers may allow a caller to spoof a caller ID that is not theirs, potentially allowing them to display an ID that is not actually the caller's number. Commercial VoIP equipment and software usually make it easy to change the caller ID information. While this service can provide enormous flexibility, it also opens up the potential for abuse.

Traffic statistics

Any VoIP connection has a number of parameters that are generally accepted as accurate indicators of assessing the quality of the connection. In addition, most existing IP telephony operators, when providing services, even allow you to select the node through which the call will pass, not only based on price, but also on additional statistical parameters characterizing the quality of communication:

ASR/ABR - the ratio of the number of calls served to the number of attempts to call as a percentage. Characterizes the best dialing.
ACD - average duration of calls through a node to a given direction; % - the percentage of completed calls with a duration of less than 30 seconds. Characterizes the most stable connection during a conversation.

Sometimes telecom operators also use other statistical parameters to evaluate the direction: Erlang, post-dial delay (PDD), percentage of packet loss (QoS), maximum increase in calls per second (Calls per seconds, CPS).

The IP telephony station/server records detailed information about each specific call in the form of CDR records (call detailed records). Each record contains the number of the caller (A-number) and the called one (B-number), subscribers, IP addresses (or domain names), time and duration of the call, as well as the initiator and reason for termination. Detailed records of calls (Call Detail Record) are often uploaded to the billing system for analysis and subsequent blocking of the caller's account, if call authorization is necessary (RADIUS). This verification method is usually typical for postpaid payment systems.

Online accounting in billing is also used through the Accounting procedure in the RADIUS protocol, which is convenient in prepaid payment systems.

Notes

Links

Monitoring and Debugging VoIP Networks Using a Network Analyzer
Attack on VoIP: Interception and Eavesdropping
“Chronic diseases” VoIP (online presentation, 16 min)

IP telephony – VoIP telephony – SIP telephony, what is the difference between them?

There is no difference between these three terms; they mean the same thing. When working with IP telephony, you can keep two other names in mind and use them if you want to demonstrate your rich vocabulary or avoid tautology.

Why is connecting to SIP telephony (VoIP telephony) significantly cheaper than connecting to traditional telephony?

SIP telephony does not require costly work associated with laying a telephone wire many kilometers from the operator’s PBX. Working with IP telephony is carried out very quickly on the local wired Internet.

How to choose equipment for IP telephony?

You can go to a website that sells equipment for IP telephony and choose the model you like. If the client has doubts about the correctness of his choice, then he should contact the system administrator for help. UIS sales specialists will also be able to help you during consultation and connection.

How to connect IP telephony?

Call UIS or leave a request on our website. We will connect everything ourselves in a short time and answer all your questions. We have an individual approach to each client, and we will definitely study the specifics of your business.

When connecting to the IP telephony service, the client will have access to all the functionality of the Virtual PBX, including the voice menu and voice mail. To do this, it is only necessary that the device to which the PBX is connected has access to the Internet and also receives calls via the SIP protocol.

Will I be able to receive incoming calls by activating the IP telephony service?

The IP telephony service provided by UIS allows you to both make and receive calls anywhere in Russia and around the world. Moreover, incoming calls to SIP numbers are free even on the starter tariff.

From which devices can I make calls?

Let's just say that an old Soviet landline phone is unlikely to be suitable for such calls. As for modern phones, any device that supports SIP protocols is suitable for making calls using the IP telephony service. You can make and receive calls both from a computer and through a mobile application.

IP telephony or Internet telephony is used by those who are looking for profitable communications and the advantages that new software technologies provide. In our company you can connect IP telephony - high-quality communication to optimize costs and develop your business.

IP telephony and telephone communication have a significant difference - the first requires only software and Internet access, while the second requires a telephone exchange and lines, which limits the number of connections or increases the costs of a company that wants to establish uninterrupted unlimited communication. In addition, IP telephony includes telephone communications, video calls, the ability to organize conferences, and much more.

In connection with this, IP telephony is being used more and more widely and has a number of advantages:

fast connection;
good audibility and high-quality video;
communication security;
possibility of “smart” redirection;
the ability to record a voice greeting;
recording conversations, etc.

Using IP telephony to make online calls allows you to implement many options easier and cheaper. Moreover, for communication you can use mobile phones, landlines, or install the program on a smartphone or PC, depending on what is more convenient.

The use of IP telephony provides the following business opportunities:

not a single call from your clients will be missed. Multichannel communication makes this a reality. However, the number of operators (employees) may not be enough to receive calls, and then voicemail will come to the rescue;
savings when using the Internet and IP telephony compared to conventional communications are about 50% or more;
you can take advantage of the additional opportunities that this type of communication provides;
you can choose any number (for example, known as a toll-free number 8-800 or a number with an area code (numbers with area codes for more than 40 Russian cities are available).

By connecting IP telephony for online communication and choosing a number, you get not only high-quality calls, prompt service and technical support, but also better opportunities in the future. We are constantly improving, finding and implementing the best technical solutions.

What is needed to connect IP telephony

The sequence of actions to connect to the following services is simple and consists of several steps:

Selecting a number and tariff;
Concluding an agreement with our company;
Determining numbers for call forwarding;
Our specialists will connect the number and help you configure the necessary functions.

Contact our company to get advice or help in choosing a number and the optimal tariff and connect IP telephony to expand opportunities for your business.

Long-distance and international calls over “traditional” telephone lines are a real luxury for the majority of residents of our country. The high price and low quality of communication are forcing more and more users to refuse to communicate with family and friends from other regions. Moreover, the unprotected nature of the telephone line makes it an easy target for attackers - with the proper skills, anyone can connect to it and talk at your expense.

If you do not want to expose yourself to the risk of being robbed, but still cannot refuse “international communication,” you should think about connecting to IP telephony. What are Internet calls, what are the advantages of the system and how to set up a connection - we’ll try to figure it out.

Digital calls

IP telephony is a combination of old habits and modern technologies. It is a set of communication protocols and methods that enable traditional dialing and two-way communication over the Internet and any other IP networks.

Unlike regular landline phones, in which the interlocutor’s voice is transmitted via an analog signal, in IP telephony the audio is encrypted in binary code and compressed. This improves communication quality and reduces network load. Other advantages of calling over IP telephony are:

Low cost of international and long-distance calls.
Independence from telephone lines.
Make calls anywhere.

As a final advantage, we should highlight the ability to block unwanted incoming calls at your expense using the figure-of-eight method. Using IP telephony is profitable and convenient, but, like everything new, you need to get used to it.

Types of IP telephony

Internet calls can be made from a regular landline phone, from special IP equipment, and even from a computer.

According to the type of device through which communication is carried out, there is a division of IP telephony for the home into types:

"Computer-computer". To communicate, subscribers need a PC with installed software and an Internet connection. The call in this case is similar to communicating on Skype. This type of connection is the least common.
Communication via card. To make a call, you need a regular landline telephone with tone dialing and an access card from your provider. To contact a friend, you first call the operator's number, enter your ID and PIN code in tone mode, and then the number of the called subscriber.
Communication via IP phone. A special IP phone is already configured for communication. All you need is an Internet connection. When you make a call, the phone automatically connects you with the provider, connects you to the proxy server and calls the subscriber.

Many people probably have a question: what is an IP phone? It is a regular device with a handset and a keyboard, works independently of a computer and can receive a call at any time.

Foreign operators in Russia

Choosing a provider is the first step towards communicating via IP telephony. The cost of calls will depend on the choice you make, so choose a service company, considering all the pros and cons. The largest representatives of IP telephony in our country are Sipnet and Comtube.

Sipnet is one of the first foreign companies to open its representative office in Russia. Its services are ideal for making calls within the network, that is, for communicating with intracity numbers - calls are completely free. For other directions, tariffs for IP telephony are as follows:

international calls - from 1.5 to 6 rubles/min;
intercity communication - up to 1 rub/min.

Reviews about the operator are positive. Some are confused by the fact that it is conducted in English.

Comtube is one of the youngest and most promising providers. It provides its clients with two sets of services - “Starter” and “Premium”. The first set will provide customers with basic capabilities, and the second, among other things, will provide a number of additional services. The cost of calls depends on the terms of the contract.

It is impossible to give reliable reviews about this operator - too few users are familiar with it. Some are pleased with the quality of communication and the fairly wide range of options in the “Start” package, others complain that all the service is focused on VIPs.

Domestic providers

Zebra Telecom is one of the most promising providers in Russia. Provides clients with the opportunity to make calls using an access card, as well as through a PC and IP phone. Calls from Zebra to Zebra are absolutely free. Long-distance calls cost from 50 kopecks/min, international calls from 1.5 rubles - it all depends on the subscriber’s country.

Among the positive qualities, it is worth noting the Russian-language localization of the software. According to customer reviews, this is one of the many advantages of the domestic provider.

IP telephony from Rostelecom will be a profitable solution for those who need constant communication with subscribers from other countries. Unlike other operators that charge calls by the minute, with Rostelecom you pay for packages of minutes once a month or as your traffic is used up.

So, the cost of a package of 100 “international” minutes will be about 250-300 rubles. It does not matter which country you are calling to. But for some clients it will be more prudent to use per-minute billing, especially if you talk to “abroad” for less than 50 minutes per month.

Necessary equipment

The set of equipment necessary for communication does not depend on the chosen telecom operator, but on the preferred type of IP telephony. So, for computer-to-computer calls you will need an Internet access point - a fiber optic cable, a USB modem and equipment for comfortable communication: a microphone, headphones, and for video calls - a webcam.

To call IP telephony numbers from a landline phone, you will need a SIP adapter and a computer or router with a built-in IP gateway. If you use a hardware IP phone, then you won’t need anything other than it.

Setting up communication via adapter

When choosing an operator, ask what equipment is needed to connect to a leased line, whether it is easy to buy and whether it is expensive.

Some providers offer their clients already fully configured adapters for communicating with other subscribers. In this case, the entire setup process comes down to sequentially performing several steps:

Follow the instructions to connect the adapter to the network.
Connect a landline phone to the LINE1 slot using a regular
Turn on the adapter by inserting the power supply into the socket, wait for the download to complete (2-3 minutes).
Pick up the phone and wait for a tone.

As soon as you hear a tone on the other side of the line, you know that you understand what IP telephony is and were able to connect the equipment correctly. Now there are no barriers to communication.

Setting up some softphones

Setting up IP phones has some special features. In particular, you will need to program your device and enter information about yourself into its database. Doing this is not as difficult as it seems.

Before you begin, make sure you have an ID and a passphrase or PIN. Let's give an example of the data that most phones require when connecting to the Sipnet operator.

Other operators should have a procedure similar to the one presented for setting up IP telephony. Customer reviews report that it is not difficult to deal with. In addition, if problems arise, you can contact the operator. It is their responsibility to help you with the setup.

Disadvantages of IP telephony

Now that you have studied the basic principles of connection, the advantages and technology of Internet telephony, you are ready to learn about some of the disadvantages of this method of communication. Such information will help you decide whether you need IP telephony or not.

The first and one of the most significant disadvantages is the dependence on the electrical grid. If you use a PC or landline phone to communicate, then “without light” it will be impossible to reach you, as well as you will not be able to make calls yourself. The exception is hardware IP phones.

When making a call for the first time, the interlocutor most likely will not recognize you. It's all about caller ID - the caller's display will show the number of the gateway you are connected to, not your own.

The last and one of the most important disadvantages is the high price of the equipment. Many clients are perplexed that such an IP phone can cost up to 3-4 thousand rubles, and this is without a subscription fee. However, having installed it in your apartment, you will no longer have to think about prices when communicating with subscribers from other countries.