Encoding, transmission and playback
Two groups have created standards for IP telephony: the International Telecommunications Union (ITU), which controls phone standards, and the Internet Engineering Task Force (IETF), which controls standards TCP/IP. After considering the conceptual components of an IP telephony system, the protocols proposed by each group will be reviewed. Fortunately, both comply with basic audio encoding and transmission rules:
- Audio is encoded using a well-known standard such as Pulse Code Modulation (PCM);
- Digitized audio is transmitted using Real-Time Transport Protocol.
- Each RTP message is encapsulated in a UDP datagram, which is also encapsulated in an IP datagram for transmission.
UDP is used for transport because TCP's long delay doesn't make sense for a phone call - the audio must be played as it arrives, and this execution cannot stop to wait for a packet to be relayed. If a packet is lost, the lost audio is played silently but continuously.
To understand why RTP is necessary, remember that UDP offers delivery through best-effort semantics. To address the problem of duplication, jitter, and out-of-order packet delivery, each RTP message has a sequence number and a real-time clock value. The receiver of the RTP message uses the two values to eliminate duplication and place the messages in the correct order and sequence relative to the runtime (playback). In particular, the real-time clock value allows the receiver to reconstruct the exact time sequence of the data. This way, if a packet is lost, the receiver knows the exact waiting time to start playing the next packet.
Because the IP telephony call involves two-way transfers, there are two independent RTP sessions. That is, an IP phone acts as an RTP sender to send data and an RTP receiver to receive said data.
A basic IP phone system
As shown in the figure below, the simplest IP telephone system uses two basic components interconnected by an IP internet:
- IP phone
- Media Gateway Controller
An IP phone is a device that uses IP for communication and allows humans to make or receive phone calls. An IP phone can be a standalone hardware unit, analogous to a conventional phone, or a lap-top with IP telephony software, speaker and microphone. The connection between an IP phone and the rest of the world can be a wired or wireless network (ie Ethernet or 802.11b).
VoIP without computer
Another way of using VoIP telephony is via ATAs (Analog Terminal Adapters) or VoIP Gateways (generally with more resources than an ATA). ATAs and GWs are devices that can be connected directly to broadband access (ADSL, Cable, etc.) and to a common telephone set or to a PABX in trunk positions (FXS) or extensions (FXO). With the service enabled, it is possible to have a phone anywhere in the world working 24 hours a day, without the need for a computer and headphones. This solution generally brings reductions in long-distance rates and, in the corporate case, the interconnection of branches at a lower cost.
Context
Currently, the increase in telephone operators has triggered a considerable drop in the prices of long-distance calls, and this has not happens only due to the competition established between these companies, but also due to the emergence of low-income communications alternatives. cost. However, a paradigm shift begins to take place. Today, data traffic is starting to grow more than telephone traffic, and voice traffic is increasing over the data network, radically altering voice transport. This article will deal specifically with VoIP, which stands for Voice over IP. With it, a significant reduction in telephony costs is expected, especially in long-distance calls. The initial idea is to enable low-cost communication within companies and outside them. According to the magazine “Foco Economia e Negócios”, of November 30, 2005, p. 44, call cost reductions of an average of 50% (if DDD) or more than 80% (if DDI) are expected.
The National Telecommunications Agency - ANATEL has as a definition contained in its glossary that VoIP (Voice via IP) is the technology that enables the use of IP networks as a means of voice transmission. The concept is simple and consists of converting analog voice packets into digital packets and making them travel over the internet.
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Advantages and Difficulties
Some of the advantages of deploying VoIP are in the cost/benefit ratio: significantly reduce the cost of telephony, perform long distance calls at local costs, integrating mobile and landline telephones, are just some of the advantages found in this technology. Another advantage is not having to create another structure just for voice traffic, thus being able to take advantage of the structure that already exists in the company. Due to this, the return on investment is faster.
If the network structure and equipment are old, the changes can cause a scare due to the high price in the acquisition of new equipment such as cabling, Hubs, Switches, Routers, IP Phones and labor specialized. Another difficulty encountered is related to the cost of equipment that is still quite high, for example, a One of the simplest IP phone models on sale today, it's being sold for around R$ 200,00.
There is also the limitation of IP networks to transmit voice, thus generating fixed transmission delays; one of the main causes of fixed delays are: compression, between-processes, transmission, network, buffer, decompression. Another would be the existing loss in the network, for a greater guarantee it would be necessary to increase the bandwidth, as well as an improvement in the increase in the processing time of the nodes.
regulation
There is still no regulatory consensus on VoIP in the world. Nowadays in Brazil there is still no discussion about VoIP regulation, but the increase of this technology leads us to think that soon we will have movements in this direction.
The agency responsible for regulating telephony in Brazil is the National Telecommunications Agency – ANATEL, which is managed by the General Telecommunications Law – LGT. Brazilian law does not classify Voip as a telecommunications service, but as a value-added service, when it uses part of the public telecommunications network. The VoIP service is governed by Art. 61. of the LGT. VoIP is governed by the Regulation of the SMC, Multimedia Communications Service, “which enables the offer the ability to transmit, issue and receive multimedia information using any means" (Article 3º). And it defines multimedia information as “signals of audio, video, data, voice and other sounds, images, texts and other information of any kind”. In other words: although it is a service that transports sounds and voice (phone) over great distances (telephone), VoIP is not telephony. And to make this clear, Art. 66 of the Regulation establishes that “In the provision of SCM, the offer of services with characteristics of a Fixed Telephone Service is not permitted.
References
EAT, D. AND. Internetworking with TCP/IP: Volume I – Principles, Protocols and Architecture, 4th edition, Prentice-Hall, 2000, Upper Saddle River, NJ.
EAT, D. AND. Computer networks and the Internet. Bookman Company. 2007.
DRIVER, H.; HOPEWELL, H.; IAQUINTO, J. How the Gateway Regulates Information Control, Data communications, September 1979.
Per Wendley Souza
Prof. computer engineering
Federal University of Ceará - UFC
Columnist Brazil School
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Voice Over IP - Protocols
The main protocols used in VOIP.
VOIP - Computing - Brazil School