Introduction To Voice Over IP For Surpass

0BIntroduction to Voice over IP for SURPASS
Introduction to Voice over IP for SURPASS
Contents
1 1.1 1.2 1.3 2 2.1 2.2 Voice over IP Basics A Voice Network Data Network Development and Examples Gateways - Phone meets PC Why Voice over IP Advantages for network operators Advantages for users and subscribers 3 3 8 11 19 19 22
TG7017 VoIP for Surpass
1
1.1
Voice over IP Basics

A Voice Network Data Network
The communication networks nowadays represent one of the most important pillars of modern industrial societies. It is difficult to imagine that only a little more than a hundred years of development led to this now seemingly indispensable device of modern life. Before turning to more detailed discussion in the field of VoIP, using an IP data network, we want to give a short overview of the networks that represent the present state and the opportunities for improvement. Generally we can distinguish coarsely two types of communication networks, that of networks mainly used for voice transmission (namely the ISDN/PSTN Telephone Networks) and data transmission networks used to exchange information between computers These different types of networks have coexisted from the beginning of data communication Even in the 60ties of the 20th century, the early years of this new technique, data transmission via the already existing Public Switched Telephone Network (PSTN) turned out to be unsuitable. So already at that time installation of data networks began, and the most successful one was the ARPANET. It was the first step toward the enormous but at the beginning of the 1980ies mostly unnoticed communication network that is nowadays named IP-Network or The Internet (IP Internet Protocol).
Voice and Data Transmission Networks

Voice Requirements - real-time (critical) - small bandwidth (narrow band) - constant bit rate Data Requirements - non real-time - enormous bandwidth (broad band) - variable bit rate
Packet Switched Networks Circuit Switched Networks - no allocation of - fixed allocation of bandwidth bandwidth - flexible use of - fixed size of channels bandwidth (64 kbps) PSTN/ISDN- Network
Figure 1
IP/ATM/FR- Networks
Voiceband Communication and Circuit Switched Networks One of the most important requirements for networks with focus on voice communication is to transmit the signal in real time. In this light, circuit switched networks comply best with this key requirement, because at the very beginning of communication a circuit (channel) is set up to connect the subscribers to each other. In other words, the connection must be explicitly set up, before voice traffic can be transmitted. An appropriate bandwidth is allocated exclusively to such an established circuit as it has to be ensured, that the network is able to transmit the voice traffic within both small total delay times and small delay variations (jitter).
Features of Circuit Switched Networks

Appropriate for real time applications Small and predominantly constant delay times Connection-oriented Fixed allocation of bandwidth (n*64 kbit/s)
Figure 2
Circuit Switching
Phone 1
Switch
Phone 3
Switch
Switch Traffic between Phone 2 and Phone 3
Phone 2
Figure 3
Traffic between Phone 1 and Phone 4
Phone 4
Data Communication and Packet Switched Network In contrast to voice communication, for data communication less strict real-time requirements are needed. Data transmission times can jitter within larger limits and the acceptable total delay times can be longer. On the other hand the required bandwidth can be subject to considerable variation. For this reason packet switched networks such as the IP-Network were introduced. In spite of allocating circuits, packets are sent independent from each other throughout the network one at a time. In case there are too many packets in the network, congestions can appear in the network. Short-term overloads (traffic peaks) can be sustained by buffering the packets in order to forward them when the peak has passed. But if such an overload situation stays unchanged, packets even can be discarded. Voice over Packet (VoPacket) and in particular Voice over IP was initially regarded as not feasible because of the following set of problems: total delay time (caused by Packetizing and Routing) considerable variation of the delay risk of too much packet loss
Features of Packet Switched Networks

hardly appropriate for real-time applications highly variable delays (depending on the load level) for the most part connectionless flexible use of bandwidth
Figure 4
Packet Switching
PC Client 1
Router
Server a
Router
Router
Traffic bw. PC 2 and Server a
PC Client 2
Figure 5
Traffic bw. PC 1 and Server b
Server b
1.2
1.2.1
Development and Examples

A phone call from PC to PC
The more powerful both Personal Computers and Routers in the IP Network have become, the closer the networks have come to satisfying Voice over IP solutions. In February 1995 for the very first time software was brought on to the market, that allows voice communication with the help of two Personal Computers. As prerequisite, each PC just had to be equipped with sound card, loud speaker, microphone and of course a LAN card for the IP connection . This was the beginning of the story, even though the quality of the speech was not that high. But soon other companies launched their solutions for phone calls over IP networks and the question of compatibility of the different solutions soon appeared. In January 1996 the Internet Engineers Task Force IETF published the Real-Time Transport Protocol RTP which was designed to support the transmission of real-time critical information such as voice and video over the IP network. Also in the year 1996 the ITU-T (International Telecommunication Union Telecommunication Sector) published the H.323 standard, that includes the RTP as protocol used for transmission. Moreover, this standard covers and defines protocols needed for call set up.
Voice over IP - History

Februar 1995: first VoIP-solution on the market (Fa. VocalTec/IL) January 1996: IETF standardizes RTP (RFC1889) May 1996: ITU-T adopts H.323
Figure 6
VoIP PC-to-PC
VoIP PC
IP-Netz, z.B. LAN
VoIP PC
or IP-Phone
or IP-Phone
Figure 7
Nowadays this kind of communication is widely-spread for instance in local enterprise networks. In doing so, a company utilizes its LAN for both data and voice services. Often the functionality of software and hardware components is integrated in IP-Phones, which are directly connected to the IP-network. In this way the conventional enterprise telephone network can be abandoned. Utilizing the PC to PC communication more generally, it is even possible to bring about cheap long distance calls, as the access fee is independent from the distance of the called party and very often it is independent from the call duration (flat rate), too To avoid expensive international long distance-connections via the PSTN, it is possible to bypass it with a connection via the global internet, which is charged (in PSTN terms) like a local call only. By running appropriate software on the PC, the established connection, for instance between a PC in Germany and a PC in India, can be used to transmit voice. However, as everybody knows from surfing the internet, delays can occur during such a telephone call. In this way the coherence of an internet phone call can suffer, and in worse cases the received speech can become incomprehensible.
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Advantages of VoIP
A unique Network: less investment, less techniques less efforts for maintainence Cost-effective: Charging can be independent from the distance Charging can be independent from the duration
Figure 8
11
1.3
Gateways - Phone meets PC
At the beginning of the VoIP development the focus was on connections from PC to PC. For instance, on one hand this kind of solution allows the staff to call each other within the companys LAN domain, but on the other hand in the early days of VoIP no option to communicate with the PSTN was given. But this important problem was clear right from the very beginning. Finally the problem was solved by the development of VoIP Media Gateways. VoIP Media Gateways allow VoIP networks to connect to the PSTN network. Acting as a kind of interpreter, VoIP Media Gateways make phone calls possible between PCs located in a packet switched IP network and POTS/ISDN-telephones located in the circuit switched PSTN. They are able to forward voice traffic received from the circuit switched technique at the packet switched technique and vice versa. A very successful offer in this context was to start long-distance calls from the PC. Contrary to the PC to PC method, ordinary subscribers are available, too. For example, at home and abroad even friends without internet access can be contacted in a cheap and easy way. However, generally such a call might not have the same quality that is usually provided by the PSTN.
12
VoIP Media Gateways

allow access for subscribers connected to external networks act as a kind of interpreter, for example between IP and PSTN networks calls from the (cheaper) data network can be forwarded at the PSTN
Figure 9
VoIP PC-to-Phone
VoIP PC
IP-Network
PSTN-Network
MG
VoIP Media Gateway
PSTNTelephone
Figure 10
13
The introduction of the VoIP Media Gateways also paved the way for two PSTN telephones to be connected via an IP network (Phone to Phone). After the liberalization and deregulation of the telecommunication market across most of the world, IP network operators can, with little effort, enter this new market easily by placing at least two VoIP Media Gateways at the borders of their IP network. This VoIP scenario is mainly used for international calls from Phone to Phone. Typically, for international connections, a translation of the national signaling protocols used in the PSTN is necessary. Therefore special (that means expensive) switches must usually be used to connect two country specific PSTN networks. VoIP gives small operators the chance to bypass this huge investment, which would not become a cash cow for them anyway.
VoIP Phone-to-Phone
IP-Network
MG
PSTN-Network
MG
PSTN-Network
PSTN Telephone
Figure 11
PSTN Telephone
14
1.3.1
Gatekeeper - More Functionality
To establish a VoIP connection generally a minimum of two terminlsl is sufficient. This can be realized by two VoIP-PCs, two IP-phones, two Gateways or any combination of the above. However, by adding special servers, call set up and release can be controlled centrally and supplementary features and functionalities can be implemented. In terms of the H.323 standard such a server is named Gatekeeper, other protocol suites use different names. One of the most important tasks of a Gatekeeper is the address resolution. Telephone numbers or alias names must be converted into the IP address allocated to the called partys terminal equipment. Moreover charging can be performed and resources like the available bandwidth can be controlled. Features that are present in every private branch exchange (PBX) can be implemented in a Gatekeeper as additional features. After improving the functionality step by step, VoIP is finally able to compete with circuit switched solutions. Since 1998 complete solutions such as IP based call centers or IP PBX have been available on the market.
Gatekeeper
centralized control of call set up and release supplementary functionality: - address resolution - control of bandwidth resources - charging Features: - Anklopfen - Makeln - etc.
Figure 12
15
VoIP Gatekeeper
Gatekeeper
VoIP PC
VoIP PC
IP-Network, e.g. LAN
Figure 13
16
1.3.2
Quality of Service
As an essential but silent prerequisite, the VoIP solutions discussed previously require an IP network that is able to handle voice traffic in real-time. But this only can be true, if the distance for voice traffic to be transmitted is not too long the involved network elements work at high speed the required bandwidth is available any time In particular the last prerequisite is difficult to fulfill. Without taking additional steps usually this is only possible if there is considerably more bandwidth available in the network than is needed at any particular point in time. This means over-provisioning is a must. If not, the quality of a voice transmission will depend solely on the actual load in the network and good luck.
Real-time requirements for data network

short distances high-speed network elements (Router, Switches) necessary bandwidth must be available
Solutions: I) Over Provisioning II) QoS-Mechanisms (IntServ, DiffServ, MPLS)

Figure 14
17
Characteristic of Voice and Data Traffic

V o ic e t r a f f ic p r o f ile ( s c h e m a t ic )
1 ,2 5
Volume
0 ,7 5
0 ,5
0 ,2 5
0 1 5 9 13 17 21 25 29 33 37 41 45 49
T im e
D a ta tr a ffic p r o file (s c h e m a tic )

1 ,2 5
1 0 ,7 5
Volume
0 ,5 0 ,2 5
0 1 5 9 13 17 21 25 29 33 37 41 45 49
T im e
Figure 15
Over-provisioning is difficult. There can be considerable time dependent variations the demand for bandwidth in the IP network because of the data traffic. Apart from this, over-provisioning is very expensive, as over a long period of time the use of the network capacity is relatively low. In recent years, new techniques have been developed in order to make sure that enough bandwidth for voice traffic can be provided any time by the network without too much over-provisioning. These techniques provide what is called Quality of Service (QoS). The most important ones are RSVP (Resource Reservation Protocol), IntServ and DiffServ (Integrated and Differentiated Services) and last but not least MPLS (Multi Protocol Label Switching).
53
53
18
Voice and Data in one network

Profile of Voice Traffic (schematic) 1,25 1 0,75 0,5 0,25 0 10 13 16 19 22 25 28 31 34 37 40 43 46 49 52 55 1 4 7 Time (arbitrary units) Traffic Volume (arbitrary unit)
Traffic Volume (arbitrary units)
1,25 1 0,75 0,5 0,25 0 10 13 16 19 22 25 28 31 34 37 40 43 46 49 52 55 1 4 7
Data Traffic (schematic)
Time (arbitrary units)
Capacity limit overflow, Several Packets (Voice? Data?) get delay

Traffic Volume (arbitrary units)
1,25 1 0,75 0,5 0,25 0 1 4 7 10 13 16
Voice Traffic + Data Traffic
19
22
25
28
31
34
37
40
43
46
49
52
Time (arbitrary units)
Figure 16
The basic concept is true for all of them: In order to reduce interference with the realtime traffic, variation in the hardly determinable volume of data traffic must be (as far as possible) avoided . With the use of Quality of Service techniques a voice quality for VoIP scenarios can be achieved comparable to that realized in the PSTN network.
55
19
2
2.1
Why Voice over IP

Advantages for network operators
Throughout the world, circuit switched telephone networks facilitate highly reliable voice communication within good quality. In this context, certainly the question arises as to why operators would want to turn away from already installed and proven circuit switched techniques, and instead turn towards voice transmission via packet switched networks. A selection of reasons are given below:
Advantages of VoIP for network operators

more efficient use of the network - silence detection - compression low costs - a unique network (Administration, Operation, Maintainence) high innovation pace - open interfaces allow seamless and quick integration of innovations from different manufactureres (between Gatekeeper and transmission technique) Integration of voice and data services (Multimedia)
Figure 17
2.1.1
More efficient use of the networks
In circuit switched networks a fixed bandwidth is allocated to a connection. For example, in a PSTN network the voice-channel is transported as a bidirectional bitstream of 64kbit/s that is allocated to each connection regardless of whether the person is speaking or not. Thus as long as such a connection is held 128 kbit/s are in use, meaning 128 x 60=7680 kbit are transmitted per minute ( or 960 kiloByte per minute respectively). For a five minute call this piles up to 4800 kiloByte. Contrary to
20
the PSTN scenario, in a packet switched network this value can be reduced considerably: Silence detection Typical phone calls waste a lot of the allocated capacity, as usually one subscriber listens to what his counterpart him is talking to. In spite of the fact that bandwidth is allocated in both directions, information is more or less transmitted unidirectionally only. By the use of a voice activity detection, the bits containing information can be separated from the ones without any meaning (i.e. background noise). Because there is no need to transmit bits without actual information, the actual traffic in the network can be reduced by about the half. For the example of a five minute call that means the amount of voice traffic that must be sent through the network can be reduced from about 4.7 MB to about 2.4 MB. This estimation does not take into account the phases in which both parties are silent.
Compression Shortly after digitalization was introduced to the telephone networks, the bandwidth for a single connection was satndardised at 64 kbit/s for a single connection. The reason for that was the ancient contemporary state of the art in telecommunication technique that was not able to keep a good quality when using less bandwidth. After several years of intensive studies, newly developed algorithms and techniques were able to reduce the bandwidth needed for high quality voice call. However, the PSTN could not take an advantage of this improvements, since the whole system is strongly based on 64 kbit/s-channels. Only new networks like the GSM mobile network, utilize the newly developed compression algorithms (and even then, only on the radio interface e.g. 13kbit/s for voice). In a packet switched network, channels of predefined size do not exist at all. Therefore compression algorithms can save considerable bandwidth especially in an IP network environment.
Quicker introduction of Innovations In a VoIP network, different functions are implemented in diverse and separated network elements. For example, while the switching of the packets is executed by routers, the logical control of the call as well as additional services are centralized in servers (Gatekeeper, etc). Communication among the diverse equipment is realized via standardized interfaces. Thus they can be upgraded, modified, renewed or replaced independently from each other. For instance, to introduce a new service, the servers only must be modified, the routers are not touched at all. If routers are replaced by more powerful ones, then the servers and their services remian unaffected. Improvements can be implemented in a much quicker and cheaper way. Moreover, thanks to open interfaces, network elements from different vendors can be combined. Additionally, this fact leads to more competition among the vendors, which can ultimately results in lower prices.
21
2.1.2
New Opportunities
Operators of data services offer voice services With the introduction of VoIP, Operators which have previously specialized in data services can now offer voice services, too. VoIP opens the door to the lucrative telecommunication market without investing in a circuit switched network. Operators offering broadband access (xDSL, cable modem, powerline) to the IP network can offer in parallel VoIP for their subscribers even though they do not own a PSTN networks access infrastructure (Local Exchanges, Access Networks). Operators of voice services offer data services The volume of data communication has been increased rapidly in the last years. It is estimated that if this trend continues, the volume of data traffic will soon surpass that of voice traffic. Operators of voice services can prepare for these challenges by a step by migrating from circuit switched networks to packet switched networks. Integration of voice and data - multimedia In a convergent voice data network, apart from the pure voice and the pure data services, services combining voice and data can also be offered. And these are not just telephone conferences launched and controlled from an internet web page. In such a network multimedia services like video telephony, video conferencing, video streaming or tele-learning can be offered. Common to all these services is both the real time criteria and the enormous need for bandwidth.
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2.2
Advantages for users and subscribers
There are several advantages for users generated by a VoIP:
Advantages for users and subscribers

unique Infrastructure (one communication technique) only one network access new innovative services (Click-to-Call, etc.)
Figure 18
Unique infrastructure Private households and companies may ease internal networking by use of VoIP. Instead of operating a data network (LAN) and a telephone exchange in parallel, one network is sufficient for both: telephony and data networking. Only one network access If the access to the voice network is realised with VoIP, voice transmission uses same line then data transmission. A special cabling for access of PSTN is not necessary any longer. Furthermore maybe one provider can offer both services. New innovative services (Click to Call, etc.) With combining voice and data services within one single network, also completely new services can be offered. Eg. customers can administrate calls via the internet. They also can directly start their calls automatically via the internet (eg. wake-up calls from weather services). Companies, which present products in the internet may improve their service by placing a button on their website, which allows the customer directly to call the company via pressing this button (Click to Call).

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