Voice over IP – Regulatory, technological and market considerations

Internet and IP-based communication protocols were born because of the need for computer systems communication that, in the beginning, could tolerate broad parameters for many of the variables regarding the nature itself of the applications to be run on such systems.

Parameters, such as transmission cadence delay and variation (Known as jitter), which is fundamental to closely control for voice services, were not critical in the data applications that existed at the time .

This led to a natural development of IP-based networks in which, in the beginning, in the their original design base structure, were not considered parameters that would enable proper end-to-end control over all of the variables that real-time communication demands, which is precisely the case of telephone services.

Building additional facilities on the existing base, alternate technologies that have been applied, and later developments made on IP finally enabled practically solving on these networks the technical problem of convergence .

General description of voice transmission problems and general technical concepts

Traditional telephony systems, which have existed for more than one hundred years, have evolved gradually from manual switch systems using human operators, passing through electromechanical systems, analogous electronic systems, and finally digital systems, to finally end up as convergent systems or soft switch systems in which the IP protocol can be fully applied.

All of them have given us several problems to solve, which arise in the transmission process of any type of information in a system of networks, and, in particular, when transmitting the human voice over a distance. We can categorize these problems into four fundamental aspects in all telecommunications networks.

Switching, Multiplexing, Signaling, and Routing

Why is it necessary to solve these problems? They arise when passing from a point-to-point communication to a multiple point-to-multiple point communication, when configuring networks with universal access, where any subscriber to such networks must be able to reach any other subscriber. So, we could say that communications networks, in general, must provide the infrastructure that carries information of any nature between remote points. Here it is important to remember the definition of telecommunication in the ITU recommendations: Recommendation B.13. “All transmissions, broadcasting or reception of signs, signals, written material, images, sounds or information of any nature by threads, radioelectricity, optic media or other electromagnetic systems”. Three elements are clear in this definition: 1) the process run, that is to say, the broadcasting, transmission, and reception. The existence of a channel: medium, element or system, and finally what is carried: that is to say, information of any nature.

We can deduce a fundamental problem from this concept. It is seeking economic efficiency, with a proper cost-benefit ratio, to enable offering the service to more subscribers, as the value of a telecommunications network is intimately related to the number of users connected to it , it is then desirable to be able to extend the service to as many users as possible. That implies that, to the extent possible, the elements proper to each stage of a telecommunication must be designed seeking scale efficiency and scale economy. Therefore, it is desirable to share communication channels among several users (switching process), activating said channels on a jointly determined medium (multiplexing process), ensuring proper process control (signaling), and seeking in the network the most technically and economically efficient route (routing). These are the main functions that we find in any telecommunications network. Here we would like to mention “Metcalfe’s Law”, named in honor of Bob Metcalfe, the inventor of Ethernet. This law status that the value of a network increases geometrically based on its number of users.

So, let’s take a look at how each of these aspects has been approached for traditional voice telecommunications.

Comparison between traditional voice switching technologies and data transmission services

Just the opposite of traditional voice networks, the world of computer systems that started a vertiginous development with the onset of microelectronics and digital systems, at a certain point in time understood the need for these systems to be able to share information and interact. This gave birth to the need to establish communication mechanisms for computer systems.

At first, such systems had some communication requirements that were quite different challenges from those seen in voice communication. Among them we mention the ones below.

1. Less probability of error in transmissions
2. Low sensitivity to delay
3. Low sensitivity to jitter
4. Possibility of splitting information
5. raffic behavior in bursts
6. Communications not necessarily connection-oriented
7. Possibility of sending information from various sources on one sole channel
8. Multipoint-to-multipoint communication schemas
9. Possibility of dynamic routing
10. Seeking efficiency in communication costs

Below we draw the conclusions from this first chapter.

• Every telecommunications network must cover the basic functions of switching, channel multiplexing, signaling, and routing.

• Traditional voice services evolved into telephone networks from mechanical and electromechanical technologies with analogous transmission, to networks with analogous electronic switching equipment, and finally, to digital circuit switching networks.

• The voice digitalization process opened the way for the gradual development of multiservice networks based on digital standards.

• Data networks that were born and evolve during the second half of the 20th century, gave a base that, although designed at first for transactional environments and batch file transfers , improved and evolved into networks with multiservice capacities, where it is feasible to send voice contents in real time.

• The main problems to solve in data networks for voice services that converge on them are the problems of loss of information packets, transmission delays and jitter or the change of cadence in the information flow.

• Among the techniques applied to solve these problems, we find the development of protocols and mechanisms for reserving band width in packet networks, voice compression, suppressing silences in the transmission, and handling buffers and specialized queues that enable prioritizing voice traffic over data traffic, as well as regenerating the signal without jitter effects. In addition, the problem of the loss of packets in voice transmission due to congestion, errors or extreme delay can be solved by using predictive algorithms and signal intrapolation and extrapolation.

• During the gradual development of data networks, we find protocols, the most important being X.25, Frame Relay, ATM, and finally the IP protocol family.

• For voice handling, TDM maintains a low delay and error rate but low efficiency in using the media. Frame Relay and ATM improve efficiency, although the former introduces some voice delay.

• On the other hand, IP introduces delays and generate more probability of transmission error but it is the protocol most used and with the greatest scope.

• The IP protocol family includes not only the base IP protocol but also diverse protocols for communication control such as TCP and UDP, routing protocols such as RIP, OSPF and BGP, service quality reservation protocols and mechanisms such as RSVP and MPLS, and specialized protocols for particular applications, among which we find those applicable to VoIP services such as SIP, those under the standard H.323, and MEGACO.

• The base for IP network handling is the routers, switching elements with a high degree of versatility and processing capacity, which are the heart of IP networks and of Internet.

• Service quality and band width reservation protocols are commonly found in private carrier or company closed IP networks. To the contrary, their use is not so common in Internet, so VoIP in the Internet network is still subject to the rule of the “best effort”. However, to the extent that available band width increases, its cost drops and accesses to broad band become massified, the effect of congestion will be less impacting and the probability of handing service quality at all times will increase.

• The advantages that traditional voice networks still have are mainly the preliminary reservation of band width, which enables predictable handling of parameters such as delay, jitter, and network congestion during a call, thus eliminating their harmful effects.

• Although IP-based voice networks still suffer from the deficient handling of the above-mentioned quality parameters, they enable more efficient use of communication channels in a convergent environment, separating the access provider from the service provider, and integrating these networks in multimedia applications.

• Below we add some considerations regarding the most relevant quality parameters to be taken into account in voice services on IP. They are taken from Table 3 “Quality Parameters in IP Telephony” in Number 3.1 in the CITEL document “Folder Technique: Structure of the study on voice characteristics de la voice based in networks that use IP”, which in turn uses the Gartner Group, Inc. as its source.

Colombian Association of Engineers (ACIEM)