This chapter aims to provide an introduction to satellite internetworking with terrestrial networks in terms of related access and transit transmission networks. It also provides an introduction to network traffic related to user plane, control plane and management plane, network hypothetical reference connection, traffic engineering, network signalling schemes, performance objectives, SDH over satellite, and internetworking between the satellite and mobile ad hoc networks. When you have completed this chapter, you should be able to:
Telecommunication networks were originally designed, developed and optimised with respect to the speech transmission quality of narrow-band 3.1 kHz real-time telephony services.
In the early generation of data networks in wide area, people tried to fully utilise the 3.1 kHz for data communications without the additional costs of a network infrastructure. At that time, the transmission speed of the data terminals was relatively low. In addition to telephony services, the networks can also support the transmission of non-voice signals such as fax and modem transmission, and wholly digital data transmission. To some extent, the telecommunication networks could meet the transmission demand of data communications.
Because of the development of computers as network terminals, high-speed data networks had to be developed to meet the demand of data communications. This led to the development of different types of networks for different services. Traffic on data networks is becoming larger and larger, and the same applies to network capacity. The increase in traffic generated the opportunity to transport telephony voice services over data networks. High-capacity user terminals and network technologies enable the convergence of telephony services and data services, and also broadcasting services. A new type of network, broadband networks, has been developed to support the convergence of services and networks.
All these developments are great for new services and applications, but also bring great challenges to internetwork between these different types of networks. Due to economic reasons, new networks are ‘forced’ to interface with legacy networks.
It is even more of a challenge for satellite networks to interwork with all these different types of networks. One of the great problems in telephony networks is that the terminals and networks are so well engineered that any change in one party would be restricted in the other party. Modern networks try to separate the functions of user terminals from the networks so that the user terminals provide services without concerning too much about how the traffic is transported over the networks, and the networks provide different types of transport schemes with little concern about how the terminals are going to process the traffic.
We will follow the same principle to discuss satellite internetworking with terrestrial networks, that is, what the requirements are from the terrestrial networks, and how the satellite networks will be able to meet these requirement for internetworking purposes.
Medium and large private networks consist of several interconnected multi-line telephony systems (MLTS). The terms ‘corporate network’ or ‘enterprise network’ are sometimes used to describe a large private network; in some countries these terms are used in a legal sense for a group of interconnected private networks. From the point of view of networking, there is no difference between a large private network and several smaller interconnected networks. Therefore, only the term ‘private network’ will be used to refer to this type of networks.
A private network can be a terminating network (one to which terminal equipment is connected). It can also provide transit connections between other networks. We will emphasise the case for terminating networks, as the transit network case is quite similar to public networks.
We will focus more on the principles of all kinds of intra- or internetwork connections rather than detailed implementations, regardless of the number of public or private networks involved, or the specific configuration in which they are interconnected.
Therefore, there is neither restriction on the network with respect to size, configuration, hierarchy, technology used, nor on the components of the network.
Today, all communications networks in the world are now digital. Radio resource management in the frequency domain still uses the same principle as analogue networks; and in the time domain all are in digital. Naturally we will focus our discussion more on digital networks because of the prevalence of digital signal transmission media and digital signal handling in switching equipment.
Before going into details, we will explain the definitions of a number of terms pertinent to the relevant concepts:
The term ‘private network’ is used to describe a network which provides features only to a restricted user group, in contrast to the public network available to the general public. In general, a private network is a terminating network and consists of several interconnected nodes (local switch, routers, gateways), with interconnections to other networks via mainly public networks.
A private network has the following characteristics:
The term ‘public network’ refers to networks providing transmission, switching and routing functions as well as features which are available to the general public, and are not restricted to a specific user group. In this context, the word ‘public’ does not imply any relation to the legal status of the network operator.
In some cases, a public network may provide a limited set of features only. In a competitive environment, a public network may be restricted to serve a limited number of customers, or restricted to specific features or functions. Generally, public networks provide access points to other networks or terminals only within a specific geographical area.
From the point of view of an end-to-end connection, a public network can function either as a ‘transit network’ (a link between two other networks) or as a combination of ‘transit and terminating network’ in cases where the public network provides connections to terminal equipment such as telephone sets, local switches, computers, routers or gateways.
In telephony networks, quality aspects take into consideration both the telephone set and different components in the network. The perception of speech transmission quality during a telephone conversation is primarily a ‘subjective’ judgement. The concept of ‘quality’ may not be considered as a unique discrete quantity, but may vary, depending on the user's expectation of sufficient ‘speech transmission quality’ for a 3.1 kHz telephony call for the terminal mode (e.g. handset) as well as the particular service (e.g. wireless). An end-to-end consideration is taken into account from one human's mouth to another human's ear.
For the judgement of the quality in a given configuration, and the performance of ‘subjective tests’, the ITU-T has developed several methods. One of the most common methods is to perform laboratory tests (e.g. ‘listening-only tests’), wherein the test subjects are requested to classify the perceived quality into categories. For example, a ‘quality rating’ can be graded on a 1–5 grade scale as bad, poor, fair, good and excellent.
The scores are used to calculate the average value of the judgement of several test subjects for the same test configuration. The result is the so-called ‘mean opinion score’ (MOS), which may, theoretically, range between 1 and 5. An assessment about the speech transmission quality can also be obtained by calculating the percentage of all test persons rating the configuration as ‘good or better’ or as ‘poor or worse’. For a given connection these results are expressed as ‘percentage good or better’ (%GoB) and ‘percentage poor or worse’ (%PoW).
Therefore, it is a complicated task to evaluate quality of services in telephony networks, and involves the collection of the necessary information on the various network components in the configuration investigated and their contribution of transmission impairments that impact the end-to-end connection speech transmission quality. The ITU-T has developed several methods and tools to evaluate QoS over telephony networks.
In digital networks, the impairment in any part of the network does not propagate from one part to the other part. Therefore, the quality of services can be evaluated for each element separately. For example, the modern network terminals are capable of buffering digitised voice or putting the voice into memory before playing out. The terminals should be given the freedom of how long to keep and how much to buffer the voice. Similarly, modern networks process the digitised voice in terms of frames or packets, and should also be given the freedom of how much time to process and what the sizes of frames or packet should be.
As delay is one of the main factor affecting the quality of the telephony service quality, any delay or delay variation should be minimised; but for data service, loss should be minimised as data service is more sensitive to loss than delay.
IP based networks have been developed based on the Internet protocols transmitted over different types of network technologies including LAN, WAN and wireless and satellite networks. From a board gateway protocol (BGP) router point of view, the world consists of autonomous systems (AS) and lines connecting them. Two AS are considered connected if there is a line between boarder routers in each one.
The network can be grouped into three categories. The first category is the stub networks, which have only one BGP router connecting to the outside, hence cannot be used for transit traffic. The second category is the multi-connection networks, which can be used for transit traffic except that they refuse to carry the transit traffic. Finally, the third category is the transit networks, which are willing to handle third parties, possibly with some restrictions, and usually charge for their services.
Each AS has a similar structure. The stub network sending traffic to and receiving traffic from backbone networks, and backbone networks to transport the traffic between the AS. Typical networks include:
They consist of internal routers and edge routers (e.g. between LAN and WAN). The telecommunication network can be used to link the routers together and link IP terminals to ISPs.
IP based networks rely on the Internet protocol (IP) and provide packet based transport of data. Thus, a digitised speech signal will be divided into small segments for the real-time transport protocol (RTP) in the application layer, the user datagram protocol (UDP) at the transport layer protocol and then the Internet protocol (IP) at the network layer. The header of these protocol layers in general contains the following data:
Finally, at the receiving side, the speech segments are used to construct the original continuous digital speech signal. For non-real-time data services, transmission control protocol (TCP) is used at the transport layer.
Network components in an end-to-end connection can be categorised into three main groups: network terminals, network connections and network nodes.
With respect to speech transmission, terminals are all types of telephone sets, digital or analogue, wired, cordless or mobile, including the acoustical interfaces to the user's mouth and ear. These components are characterised by their send loudness rating (SLR) and receive loudness rating (RLR), which contribute to the overall loudness rating (OLR) of a connection. Other parameters, such as the side tone masking rating (STMR), the listener side tone rating (LSTR), the design of the handset (D-factor), and the frequency response in send and receive directions and the noise floor, also contribute to the end-to-end connection rating of speech transmission quality.
In the case of wireless or IP based systems, additional distortions and delay may be added, depending on the coding and modulation algorithms used in such interfaces. However, with packet networks, there are great advantages in the terminal with memory and processing power overcoming the problems of telephony networks.
With respect to data transmission, terminals include all types of computer, smartphones and even smart TVs.
Network nodes are all types of switching equipment, such as local switches and main switches in telephony networks and routers in the Internet. These nodes may use analogue or digital switching or packet-switching technology. The main impairment contributions of analogue systems are loss and noise in telecommunication networks. Where four-wire to two-wire conversions take place within or between switching equipment interfaces, signal reflections contribute to impairments as a source for echo effects. Digital switching systems contribute to the end-to-end delay, due to signal processing, and also to the amount of quantisation distortion associated with digital pads and code conversion. Packet based routers contribute, in addition, to delay and delay variation due to data buffering and packet loss due to transmission errors or buffer overflows.
Network connections use all kinds of media as the facility between network nodes and between nodes and network terminals. The physical media of these connections may be metallic (copper), fibre optics or radio. The signal form is either analogue or digital. Impairments associated with analogue signal transmission include propagation time (generally proportional to distance), loss, frequency response and noise (mainly due to longitudinal interference). Impairments due to frequency response and noise can usually be neglected for short and medium line lengths.
For digital transmission, the main impairment is caused by the propagation time via metallic, optic and radio media. For wireless sections, additional delay is introduced, depending on the coding and modulation algorithm used. Where the connection includes analogue-to-digital conversion, loss and distortion are additional impairment factors.
Multiplexing is generally used to transport several channels via one single physical media. A variety of multiplexing systems are in use in the existing networks:
In telephony networks, connections support either 64 kbit/s pulse code modulation (PCM), or the more recently introduced compression techniques based on low bit-rate codecs. In broadband networks, the connections will be able to support traffic at a much higher speed of video and data up to as high as one gigabit per second, in addition to the telephony voice services.
An end-to-end path between two user terminals can be as near as next door or as far as the other side of the world. The path may just involve a private network or a local exchange, or a private network and a local exchange, a long-distance connection in public networks and international connections.
In telephony networks, the predominance of incoming and outgoing calls are originated or terminated only within a local calling area. We can divide traffic into local calls, national long-distance calls and international calls. Therefore, the large number of user terminals can be supported by a small number of national long-distance connections. Similarly, we can use smaller number of international connections to support more national calls, as only a small proportion of the calls are routed to the international connection.
The end-to-end connection may also involve different type of network technologies including cable, optical, terrestrial wireless or satellite networks. All the technologies contribute in different ways to the performance and quality of service (QoS) of the network. Trade-off has to be made between different types of technologies so that end-to-end path quality is acceptable to users.
For example, for an acceptable level of telephony quality, one may reasonably expect that the impairment of the connection should not affect or disturb the normal communication caused by delay, noise, echo or other disruptive factors. However, the same level of quality may not be acceptable for listening to music, or causing transmission errors for data services.
The level of acceptable quality varies also depending on the types of services and considerations of economic, technical and advantage factors. In terms of economic factors, it may be concerned with the cost of use and implementation, for technical with the limit of technologies, and for advantage that people may accept lower quality for mobile, long distance and satellite network use, if it would otherwise be unavailable.
Reference configurations provide an overview of the considered end-to-end paths and to the identification of all terminals, nodes and connections, which contribute impairments to the end-to-end QoS and performance.
Due to the variety of hierarchy, structure, routing, number and types of network technologies in a network, different networking technologies (wireless, cable and satellite) may play different roles in the reference configuration. Here we try to identify some typical reference configurations, which can be used for evaluation of QoS and performance of networks with different technologies and their roles in providing network services.
Figure 4.1 shows a basic reference configuration of a telephony network. It is generalised to include international scenarios, the public network, the private network and therefore the entire connection.
Figure 4.1 Basic configuration of access and transit networks
It is assumed that the impairment allowance between the access points for calls within the national public network is allocated symmetrically with reference to the international connection, which can be considered as the virtual centre of the public network for international calls. For connections not involving an international connection, the equivalent virtual centre can be assumed to be within the portion of the highest-ranking network shown as the public network in Figure 4.1.
The private network normally connects to a local exchange or router, usually the lowest hierarchy and the common connection point in a public network. It is also possible to connect the private network directly to a higher hierarchy level, for example an international connection, bypassing the local switch. In some cases, especially for larger private networks, bypass may permit more allocation of specific transmission parameters (e.g. delay) to the private network.
A virtual private network (VPN) although provided by the public network operator, should be considered as part of the private network. The same is valid for leased lines interconnecting private networks usually provided by public network carriers. The private network with leased lines and VPN connections has some implications on end-to-end QoS and performance.
Internetworking involves the following types of traffic: user traffic, signalling traffic and management traffic. User traffic is generated and consumed directly at user terminals. Signalling traffic conveys the intelligence for subscribers to interconnect with the others across the networks. Management traffic provides information in the networks for effective control of the user traffic and network resources dynamically to meet the QoS requirement of the user traffic. User traffic belongs to application layer, which consumes the major amount of network resources (such as bandwidth). The management traffic also consumes a significant amount of resources. Figure 4.2 illustrates the relationships between user, signalling and management functions.
Figure 4.2 Relationships between user, signalling and management functions
User traffic is generated by a range of user services. Satellite networks can support a wide range of telecommunication services including telephony, broadband access, TV broadcasting and so on. Figure 4.3 illustrates some typical network connection and interfaces.
Figure 4.3 Example of network connections and interfaces
Telephony, fax and various low-bit-rate data transmission services were originally based on analogue transmission. Nowadays, they are systematically implemented and developed based on digital technologies. In analogue transmission, network bandwidth is allocated in the frequency domain for the duration of network connection. In the digital domain, network bandwidth is allocated in the time domain. The use of time division multiplexed digital carriers, especially when combined with technologies such as adaptive differential pulse code modulation (ADPCM), low bit-rate encoding and digital speech interpolation (DSI) with digital circuit multiplication equipment (DCME), can provide increased traffic capacity in terms of a large numbers of channels on such carriers.
In addition to digital voice, today broadband networks can support videoconference, high-speed data transmission and high-quality audio or sound programme channels and packet-switched data services. It can also support integration of voice, video and data or combinations of these as multimedia services.
Satellite usage must take into account the end-to-end customer requirements as well as signalling/routing constraints of a particular network configuration. The requirements of these services may also differ depending on the capacity of the satellite links.
Traditionally, telephony networks classified signalling generally into subscriber signalling and inter-switch signalling and functionally into audible-visual signalling, supervisory signalling and address signalling.
Subscriber signalling tells the local switch that a subscriber wishes to contact another subscriber by dialling the number identifying the distance subscriber. Inter-switch signalling provides information allowing switches to route the call properly. It also provides supervision of the call along its path. Signalling provides information for the network operator to charge for the use of network services.
The audible-visual signalling provides alerting (such as ring, paging and off-hook warning) and progress of the call (such as dial tone, busy tone and ring back). Supervisory signalling provides forward control from user terminal to local switch to seize, hold or release a connection and backward status including idle, busy and disconnect. Address signalling is generated from the user terminal by dialling a number used by the network to route the call.
Two trade-off factors are the signalling delay after the user dialled the number and signalling cost for setting up the call, as the network needs to reserve resources link by link until the call is set up successfully or has failed.
In telephony networks, in-band signalling refers to signalling systems using an audio tone, or tones inside the conventional voice channel, to convey signalling information. It is broken down into three signalling categories: single frequency (SF), two frequency (TF), and multi-frequency (MF). As conventional voice channel occupies the frequency band at 300–3400 Hz, SF and TF signalling systems utilise the 2000–3000 Hz band where less speech energy is concentrated.
SF signalling is used almost exclusively for supervision. The most commonly used frequency is 2600 Hz, particularly in North America. On two-wire trucks 2600 Hz is used in one direction and 2400 Hz in the other. Figure 4.4a illustrates the concept of in-band signalling of 2600 Hz within the frequency band, and Figure 4.4b illustrates two out-of-band signalling of 3700 Hz used in North America or 3825 Hz for ITU. Similarly in digital networks, there can also be in-band signalling and out-of-band signalling, as shown in Figure 4.5.
Figure 4.4 Analogue network in-band signalling and out-of-band signalling
Figure 4.5 Digital network in-band signalling and out-of-band signalling
A two-frequency signal is used for both supervision (line signalling) and address signalling. SF and TF signalling systems are often associated with carrier (FDM) operation. In supervision line signalling ‘idle’ refers to the on-hook condition while ‘busy’ refers to the off-hook condition. Thus, for such types of line signalling there are two audio tones, of which SF and TF are typical, for ‘tone on when idle’ and ‘tone on when busy’.
You may have noticed that a major problem with in-band signal is the possibility of ‘talk down’ which refers to the premature activation or deactivation of supervisory equipment by an inadvertent sequence of voice tones through the normal use of the channel. Such tones could simulate the SF tone, forcing a channel dropout (i.e., the supervisory equipment would return the channel to idle state). Chances of simulating TF tone set are less likely. To avoid the possibility of talk-down on an SF circuit, a time delay circuit or slot filters may be used to bypass the signalling tone. Such filters can cause some degradation to speech unless they are switched off during conversation. They must be switched off if the circuit is used for data transmission. Therefore, TF or MF signalling systems overcome the problem of SF. TF signalling is widely used for addressing signalling.
Multi-frequency (MF) signalling is widely used for addressing signalling between switches. It is an in-band method utilising five or six tone frequencies, two at a time, of which each have four different frequencies, forming the typical signalling of 16 buttons in the telephone set.
With out-of-band signalling, supervisory information is transmitted above 3400 Hz of the conventional voice band. In all cases, it is a single frequency system. The advantage of out-of-band signalling is that either system ‘tone on’ or ‘tone off’ may be used when idle. Talk-down cannot occur because all supervisory information is passed out of band away from the speech information portion of the channel. The preferred out-of-band frequency is 3825 Hz, whereas 3700 Hz is commonly used in the United States (see Figure 4.4b). Out-of-band signalling is attractive, but one drawback is that when channel patching is required, signalling leads have to be patched as well.
Traditionally, signalling goes along with the traffic on the same channel it is associated with on the same media. This signalling may or may not go on the same media or path. Most often, this type of signalling is transported on a separate channel in order to control a group of channels. A typical example is the European PCM E1 where one separate digital channel supports all supervisory signalling for 30 traffic channels. It is still associated channel signalling if it travels on the same media and path as its associated traffic channels.
If the separated signalling channel follows a different path using perhaps different media, it is called disassociated signalling (see Figure 4.6). ITU-T Signalling System No. 7 (ITU-T SS7) always uses separated channels, but can be associated and disassociated. Disassociated channel signalling is also called non-associated channel signalling.
Figure 4.6 Associated and separate signalling
In the OSI reference model, there are five categories for network management functions defined as the following:
Configuration and name management comprise a set of functions and tools to identify and manage network objects. The functions include the ability to change the configuration of objects, assign names to objects, collect state information from objects (regularly and in emergencies) and control states of objects.
Performance management comprises a set of functions and tools to support planning and improve system performance, including mechanisms to monitor and analyse network performance and QoS parameters, and control and tune the network.
Maintenance management comprises a set of functions and tools to locate and deal with abnormal operation of the network, including functions and mechanisms to collect fault reports, run diagnostics, locate the sources of faults and take corrective actions.
Accounting management comprises a set of functions and tools to support billing for the use of network resources, including functions and mechanisms to inform users of costs incurred, limit use of resources by setting a cost limit, combine costs when several network resources are used and calculate the bills for customers.
Security management comprises a set of functions and tools to support management functions and to protect managed objects, including authentication, authorisation, access control and encipherment and decipherment and security logging. Please note that security management is more to provide security for the network than user information.
Network management is implemented in network operation systems including user specific functions and common functions; the later are further subdivided into infrastructure functions and user generic functions.
Infrastructure functions provide underlying computer-related capabilities which support a wide range of processes. These include such services as physical communications and message passing, data storage and retrieval and human–machine interface (such as in a workstation computer with windows).
User-generic functions are general utilities in the network operation systems (NOS). They can support a number of user-specific functions. Some of the generic functions are listed in the following as examples:
Network operation systems (NOS) involve four layers of management functions: business management, service management, network management and element management with business at the top of the layers and element at the bottom as shown in Figure 4.7.
Figure 4.7 Layers of management functions in network operation systems (NOS)
The mediation function (MF) acts on information passing between network element functions and the operation systems functions (OSF) to achieve smooth and efficient communication. It has functions including communication control, protocol conversion and data handling, and communication primitive functions. It also includes data storage and processing involving decision making.
According to ITU-T recommendation Y.101, an access network is defined as an implementation comprising those entities (such as cable plant, transmission facilities, etc.) which provide the required transport bearer capabilities for the provision of telecommunications services between the network and user equipment. A transit network can be considered as a set of nodes and links that provide connections between two or more defined points to facilitate telecommunication between them.
The interface has to be well defined in terms of capacity and functionality to allow independent evolutions of user equipment and the network, and new interfaces have to be developed to accommodate new user equipment with large capacity and new functionality. The evolution of access and transit networks can be seen from analogue transmission of telephone networks, to digital transmission telephony networks and to packet and broadband networks.
Although almost all of today's networks are digital, the connections from many residential homes to the local exchanges are still in analogue transmission. They are gradually fading away with the installation of broadband access networks such as asymmetric digital subscriber line (ADSL). ADSL is a modem technology that converts twisted-pair telephone lines into access paths for multimedia and high-speed data communications. The bit rates transmitted in both directions are different with a typical ratio of 1 to 8 between user terminal and local switch.
We discuss analogue telephony networks not because the technology itself is important for the future, but because the principles of design, implementation, control, management and operation developed with the network have been used for many years, are still relevant to us today, and will still be relevant in the future. Of course these principles have to be used and developed in the new network context.
The telephony networks were well designed, well engineered and optimised for telephony services. In the context of available technologies and knowledge, the user service was telephony, the network resource was channel, and bandwidth of 4 kHz was allocated to each channel to support good acceptable quality of service.
The networks were dimensioned to provide a telephony service to a large number of homes and offices with 4 kHz channels, taking into account factors of economics such as user demands and costs of the network to meet the demands. There were well-developed theories to model user traffic, network resource and performance of the network and grade of service.
There are well-established mathematical theories to deal with these factors in classical scenarios in terms of call arrivals and holding-time distribution, number of traffic sources, availability of circuits and handling of lost calls. Some of the mathematical formulas are simple and useful and can be summarised as the following:
where n is the number of circuits available and A is the mean of the traffic offered (in Erlang). The formula assumes an infinite number of sources, equal traffic density per source and traffic lost call cleared.
The formula assumes an infinite number of sources, equal traffic density per source and lost calls held.
The formula assumes an infinite number of sources, lost call delayed, exponential holding times and calls served in order of arrival.
The formula assumes a finite number of sources , equal traffic density per source and lost calls held.
To transmit the telephony channel over satellites, a carrier has to be generated which is suitable for satellite radio transmission on the allocated frequency band and channel signal modulating the carrier can be transmitted over satellites. At the receiving side, the demodulating process can separate the channel signal from the carrier; hence the receiver can get back the original telephony signal to be sent to a user terminal or to a network which can route the signal to the user terminal.
If a single channel modulates the carrier, we call it single carrier per channel (SCPC), that is, each carrier carries only a single channel. This is used normally for user terminals to be connected to the network or other terminals as an access network. It is also possible to use this as a thin route to connect a local exchange to the network where the traffic density is low.
If a group of channels modulate the carrier, we call it multi channel per carrier (MCPC). This is normally used for interconnect between networks as a transit network or local exchange to the access network.
If all connections between earth stations used single global beam coverage, there would be no need to have any switching functions on-board satellite. If multiple spot beams are used, there are great advantages to using on-board switching, since it allows the earth stations to transmit multiple channels to several spot beams at the same time without separating these channels on the transmitting earth stations. Therefore, on-board switching will give satellite networks great flexibility and potentially save bandwidth resources.
Figure 4.8 illustrates the concept of on-board switching with two spot beams. If there is no on-board switching function, the two transmissions have to be separated at the transmission earth station by using two different bent-pipes, one of which is for connection within the spot beam and the other is for connection between the spot beams. If the same signal is to be transmitted to both spot beams, it will require two separate transmissions of the same signal; hence it will need twice the bandwidth at the uplink transmissions. It is also possible to reuse the same bandwidth in different spot beams.
Figure 4.8 Illustration of on-board circuit switching
By using on-board switching, all the channels can be transmitted together and will be switched on-board satellite to their destination earth stations in the different spot beams. Potentially, if the same signal is to be sent to different spot beams, the on-board switch may be able to duplicate the same signal to be sent to the spot beams without multiple transmissions at the transmitting earth station. The same frequency band can be used in the two spot beams by taking appropriate measures to avoid possible interferences.
In the early 1970s, digital transmission systems began to appear, utilising the pulse code modulation (PCM) method first proposed in 1937. PCM allowed analogue waveforms, such as the human voice, to be represented in binary form (digital). It was possible to represent a standard 4 kHz analogue telephone signal as a 64 kbit/s digital bit stream. The potential with digital processing allowed more cost-effective transmission systems by combining several PCM channels and transmitting them down the same copper twisted pair as had previously been occupied by a single analogue signal.
In Europe, and subsequently in many other parts of the world, a standard time division multiplexing (TDM) scheme was adopted whereby 30 64 kbit/s channels were combined, together with two additional channels carrying control information including signalling and synchronisation, to produce a channel with a bit rate of 2.048 Mbit/s.
As demand for voice telephony increased, and levels of traffic in the network grew ever higher, it became clear that the standard 2.048 Mbit/s signal was not sufficient to cope with the traffic loads occurring in the trunk network. In order to avoid having to use excessively large numbers of 2.048 Mbit/s links, it was decided to create a further level of multiplexing. The standard adopted in Europe involved the combination of four 2.048 Mbit/s channels to produce a single 8.448 Mbit/s channel. This level of multiplexing differed slightly from the previous in that the incoming signals were combined one bit at a time instead of one byte at a time, that is, bit interleaving was used as opposed to byte interleaving. As the need arose, further levels of multiplexing were added to the standard at 34.368 Mbit/s, 139.246 Mbit/s and even higher speeds to produce a multiplexing hierarchy, as shown in Figure 4.9.
Figure 4.9 Example of traffic multiplexing and capacity requirement for satellite links
In North America and Japan, a different multiplexing hierarchy is used but with the same principles.
Digital signals can be processed in the time domain. Therefore, in addition to sharing bandwidth resources in the frequency domain, earth stations can also share bandwidth in the time domain. Time division multiplexing can be used for satellite transmission at any level of the transmission hierarchy, as shown in Figure 4.9. Concerning on-board switching, a time-switching technique can be used often working together with circuit switching (or space switching).
The multiplexing hierarchy appears simple enough in principle but there are complications. When multiplexing a number of 2 Mbit/s channels they are likely to have been created by different pieces of equipment, each generating a slightly different bit rate. Thus, before these 2 Mbit/s channels can be bit interleaved they must all be brought up to the same bit rate adding ‘dummy’ information bits, or ‘justification bits’. The justification bits are recognised as de-multiplexing occurs, and are discarded, leaving the original signal. This process is known as plesiochronous operation, meaning in Greek ‘almost synchronous’, as illustrated in Figure 4.10.
Figure 4.10 Illustration of the concept of plesiochronous digital hierarchy (PDH)
The same problems with synchronisation, as described above, occur at every level of the multiplexing hierarchy, so justification bits are added at each stage. The use of plesiochronous operation throughout the hierarchy has led to adoption of the term plesiochronous digital hierarchy (PDH).
It seems simple and straightforward to multiplex and de-multiplex low bit streams to higher bit-rate streams, but in practice it is not so flexible and not so simple. The use of justification bits at each level in PDH means that identifying the exact location of the low bit-rate stream in a high bit-rate stream is impossible. For example, to access a single E1 2.048 Mbit/s stream in an E4 139.246 Mbit/s stream, the E4 must be completely de-multiplexed via E3 34.368 and E2 8.448 Mbit/s, as shown in Figure 4.11.
Figure 4.11 Multiplexing and de-multiplexing to insert a network node in PDH network
Once the required E1 line has been identified and extracted, the channels must then be multiplexed back up to the E4 line. Obviously this problem with the ‘drop and insert’ of channels does not make for very flexible connection patterns or rapid provisioning of services, while the ‘multiplexer mountains’ required are extremely expensive.
Another problem associated with the huge amount of multiplexing equipment in the network is one of control. On its way through the network, an E1 line may have travelled via a number of possible switches. The only way to ensure it follows the correct path is to keep careful records of the interconnection of the equipment. As the amount of reconnection activity in the network increases it becomes more difficult to keep records current and the possibility of mistakes increases. Such mistakes are likely to affect not only the connection being established but also to disrupt existing connections carrying live traffic.
Another limitation of the PDH is its lack of performance-monitoring capability. Operators are coming under increasing pressure to provide business customers with improved availability and error performance, and there is insufficient provision for network management within the PDH frame format for them to do this.
PDH reached a point where it was no longer sufficiently flexible or efficient to meet the demands of users and operators. As a result, synchronous transmissions were developed to overcome the problems associated with plesiochronous transmission, in particular the inability of PDH to extract individual circuits from high-capacity systems without having to de-multiplex the whole system as shown in Figure 4.12.
Figure 4.12 Add and drop function to insert a network node in SDH network
Synchronous transmission can be seen as the next logical stage in the evolution of the transmission hierarchy. Concerted standardisation efforts were involved in its development. The opportunity of defining the new standard was also used to address a number of other problems. Among these were network management capability within the hierarchy, the need to define standard interfaces between equipment and international standard transmission hierarchies.
The development of the SDH standards represents a significant advance in technology. Services such as videoconferencing, remote database access and multimedia file transfer require a flexible network with the availability (on demand) of virtually unlimited bandwidth. SDH overcomes the complexity of the plesiochronous transmission systems.
Using essentially the same optical fibre, a synchronous network is able to significantly increase available bandwidth while reducing the amount of equipment in the network. In addition, the provision within the SDH for sophisticated network management introduces significantly more flexibility into the network.
Deployment of synchronous transmission systems is straightforward due to their ability to interwork with existing plesiochronous systems. The SDH defines a structure which enables plesiochronous signals to be combined together and encapsulated within a standard SDH signal. This is called backward compatible, that is, new technology is able to interwork with legacy technology.
The sophisticated network management capabilities of a synchronous network give an improved control of transmission networks, improved network restoration and reconfiguration capabilities and availability.
This standards work culminated in ITU-T recommendations G.707, G.708, and G.709 covering the synchronous digital hierarchy. These were published in the ITU-T Blue Book in 1989 and replaced by G.707/Y1322 in January 2007. In addition to the three main ITU-T recommendations, a number of working groups were set up to draft further recommendations covering other aspects of the SDH, such as the requirements for standard optical interfaces and standard OAM functions.
The ITU-T recommendations define a number of basic transmission rates within the SDH. The first of these is 155.520 Mbit/s, normally referred to as synchronous transport module level 1 (STM-1). Figure 4.13 shows the STM-1 frame. Higher transmission rates of STM-4 and STM-16 (622 Mbit/s and 2.4 Gbit/s, respectively) are also defined, with further levels proposed for study.
Figure 4.13 STM-1 frame of the SDH network
The recommendations also define a multiplexing structure whereby an STM-1 signal can carry a number of lower rate signals as payload, thus allowing existing PDH signals to be carried over a synchronous network as shown in Figure 4.14.
Figure 4.14 Mapping from PDH to SDH
All plesiochronous signals at1.5–140 Mbit/s are accommodated, with the ways in which they can be combined to form an STM-1 signal defined in Recommendation G.709.
SDH defines a number of ‘containers’, each corresponding to an existing plesiochronous rate. Information from a plesiochronous signal is mapped into the relevant container. Each container then has some control information known as the path overhead (POH) added to it. Together the container and the POH form a ‘virtual container’ (VC).
In a synchronous network, all equipment is synchronised to an overall network clock. It is important to note, however, that the delay associated with a transmission link may vary slightly with time. As a result, the location of virtual containers within an STM-1 frame may not be fixed. These variations are accommodated by associating a pointer with each VC. The pointer indicates the position of the beginning of the VC in relation to the STM-1 frame. It can be increased or decreased as necessary to accommodate the position of the VC.
G.709 defines different combinations of virtual containers which can be used to fill up the payload area of an STM-1 frame. The process of loading containers and attaching overhead is repeated at several levels in the SDH, resulting in the ‘nesting’ of smaller VCs within larger ones. This process is repeated until the largest size of VC is filled, and this is then loaded into the payload of the STM-1 frame (referring to Figure 4.14).
When the payload area of the STM-1 frame is full, some more control information bytes are added to the frame to form the ‘section overhead’. The section overhead bytes are so-called because they remain with the payload for the fibre section between two synchronous multiplexers. Their purpose is to provide communication channels for functions such as OAM, facilities and alignment.
When a higher transmission rate than 155 Mbit/s of STM-1 is required in the synchronous network, it is achieved by using a relatively straightforward byte-interleaved multiplexing scheme. In this way, rates of 622 Mbit/s (STM-4), 2.4 Gbit/s (STM-16), 9.6 Gbit/s and 38.5 Gbit/s can be achieved.
One of the main benefits in the SDH network is the network simplification brought about through the use of synchronous equipment. A single synchronous multiplexer can perform the function of an entire plesiochronous ‘multiplexer mountain’, leading to significant reductions in the amount of equipment used. The more efficient ‘drop and insert’ of channels offered by an SDH network, together with its powerful network management capabilities can ease the provisioning of high bandwidth lines for new multimedia services, as well as provide ubiquitous access to those services.
The network management capability of the synchronous network enables immediate identification of link and node failure. Using self-healing ring architectures, the network will be automatically reconfigured with traffic instantly rerouted until the faulty equipment has been repaired.
The SDH standards allow transmission equipment from different manufacturers to interwork on the same link. The ability to achieve this so-called ‘mid-fibre meet’ has come about as a result of standards, which define fibre-to-fibre interfaces at the physical (photon) level. They determine the optical line rate, wavelength, power levels, pulse shapes and coding. Frame structure, overhead and payload mappings are also defined. SDH standards also facilitate interworking between North American and European transmission hierarchies.
The basic element of the STM signal consists of a group of bytes allocated to carry the transmission rates defined in G.707 (i.e. 1.5 Mbit/s and 2.0 Mbit/s transmission hierarchies). The following describe each level of the transmission hierarchy in SDH.
Figure 4.15 Section overhead (SOH) of the STM-1 frame
Within an STM-1 frame, information type repeats every 270 bytes. Thus, the STM-1 frame is often considered as a (270 byte 9 lines) structure. The first nine columns of this structure constitute the SOH area, while the remaining 261 columns are the ‘payload’ area.
The SOH bytes are used for communication between adjacent pieces of synchronous equipment. As well as being used for frame synchronisation, they perform a variety of management and administration facilities. The purpose of individual bytes is detailed below:
The path overhead (POH) of the VC-4 (as shown in Figure 4.13) consists of the following bytes:
Synchronous transfer module level N (STM-N) is constructed by combining lower level STM signals using byte interleaving. The basic transmission rate defined in the SDH standards is 155.520 Mbit/s (STM-1). Given that an STM-1 frame consists of 2430 eight-bit bytes, this corresponds to frame duration of 125 ms. Four higher bit rates are also defined: 622.080 Mbit/s (STM-4), 2488.320 Mbit/s (STM-16), 9953.280 Mbit/s (STM-64) and 39813.120 Mbit/s (STM-256).
Once the STM-1 payload area is filled by the largest unit available, a pointer is generated which indicates the position of the unit in relation to the STM-1 frame. This is known as the AU pointer. It forms part of the section overhead area of the frame. The use of pointers in the STM-1 frame structure means that plesiochronous signals can be accommodated within the synchronous network without the use of buffers. This is because the signal can be packaged into a VC and inserted into the frame at any point at time. The pointer then indicates its position. Use of the pointer method was made possible by defining synchronous virtual containers as slightly larger than the payload they carry. This allows the payload to slip in time relative to the STM-1 frame in which it is contained.
Adjustment of the pointers is also possible where slight changes of frequency and phase occur as a result of variations in propagation delay and the like. The result of this is that in any data stream, it is possible to identify individual tributary channels, and drop or insert information, thus overcoming one of the main drawbacks of PDH.
In North America ANSI published its SONET standards, which were developed in the same period of time using the same principles as SDH, and can be thought of as a subset of the worldwide SDH standards, however, there are some differences.
The basic module in SONET is synchronous transport signal level 1 (STS-1), which is three times smaller than the STM-1 in terms of bit rate and frame size. It has the same bit rate of 51.840 Mbit/s as the optical carrier level 1 (OC-1). The STS-1 frame consists of (9 × 90) bytes with a frame duration of 125 ms, of which three columns are used as transport overhead and 87 columns as STS-1 payload called envelope capacity.
In term of data rates OC-3/STS-3 is the same as the STM-1 with payload bandwidth of 150.336 Mbit/s or line rate of 155.520 Mbit/s.
ITU-T and ITU-R standards bodies together with Intelsat and its signatories developed a series of SDH compatible network configurations with satellite forming part of the transmission link. The ITU-R Study Group 4 (SG 4) was responsible for the satellite services and studying the applicability of the ITU-T recommendations to satellite communication networks.
SDH was not designed for the transmission of basic rate signals. Because it is a great challenge to implement and operate a satellite network at a bit rate of 155.520 Mbit/s, various network configurations were studied to allow relevant SDH elements to operate at lower bit rate whenever there is a need to transport SDH signals over satellite. These network configurations were referred as ‘scenarios’. These scenarios defined different options to support SDH over satellite, summarised as follows:
Since the bit rate of IDR is capable of supporting a range of PDH signals at a much lower bit rate than STM-1, it can be implemented with minimal rearrangement of the transponder band plans, with the possibility of mixing PDH- and SDH-compatible IDR carriers. Development work was carried out to modify existing IDR modems to be compatible with SHD at lower rates, rather than more expensive options of developing new modems (for example, for the STM-1 and STM-R options). This option is widely used in the satellite network operations.
Due to the availability of satellite networks, it is the natural to make use of satellite networks to extend the digital network for a global coverage. Though the digital network does not restrict using any particular transmission systems, it is important from a satellite radio engineering point of view to investigate how satellite transmission systems differ from the traditional systems required to support digital networks, how satellite transmission error affects the network performance, and how propagation delay via satellite link affects the network operations. It is the responsibility of the ITU-R SG 4 to define relevant requirements on conditions and performance for satellite links to carry digital channels and translate the ITU-T standards in terms that are significant for the satellite portion of the overall digital connections.
The hypothetical reference connection (HRX) is defined in the ITU-T G.821 and G.826 recommendations. It is used to specify the performance requirement of the major transmission segments of the overall end-to-end connection. The distance for reference of the overall end-to-end connections is 27 500 km, which is the longest possible connection along the earth surface between subscribers (at reference point T).
Three basic segments are identified with distances that are expected to be typical distances of the portion in the overall end-to-end connections in the context of IRX, which are allocated allowable performance degradation of 30%, 30% and 40% to low-, medium- and high-grade segments.
The 30% for the low-grade segment is shared by two sides of the connections from user terminal to local exchange. Similarly, there are two medium-grade segments from local exchange to international exchange sharing the 30%. Satellite links of fixed satellite service should be equivalent to half of the high-grade segment as 20%, if used in the end-to-end connection.
In terms of distance, the high-grade segment counts for 25 000 km and the low and medium segments on one side of the connection count for 1250 km and on the other side 1250 km. Satellite link counts for 12 500 km, if used in the end-to-end connection.
ITU-R defined the hypothetical reference digital path (HRDP) in ITU-R S.521 to study the use of a fixed satellite link in a part of the hypothetical reference connection (HRX) defined by ITU-T. As shown in Figures 4.16 and 4.17, the HRDP should consist of one earth–satellite–earth link with possibly one or more inter-satellite links in the space segment and interface with the terrestrial network appropriate to the HRDP.
Figure 4.16 Hypothetical reference digital path (HRDP)
Figure 4.17 HRDP in ITU-T HRX at 64 kbit/s
Additionally, the earth stations should include facilities to compensate for the effects of satellite link transmission time variation introduced by satellite movements, which are of particular significance in digital transmission in the time domain such as PDH.
ITU-R HRDP uses 12 500 km from the HRX to develop performance and availability objectives. The distance has been defined by taking into account various satellite network configurations with a maximum single hop covering an equivalent terrestrial distance of approximately 16 000 km. Consequently, in the majority of cases satellite is used in international segments of the connection with two landing points usually less than 1000 km from the users. In practice, satellite network landing points should be designed as close as possible to user terminals.
Satellite networks to support ISDH should allow end-to-end-connections to meet the performance objectives defined by the ITU-T. The ITU-R has developed recommendations for satellite to achieve the performance objectives in the end-to-end connections:
Table 4.1 Quality objectives for digital telephony at 64 Kbit/s
Measurement conditions | Digital (PCM) telephony (S.522) | 64 Kbit/s ISDN (S.522) |
Bit error rate (BER) | Bit error rate (BER) | |
20% of any month (mean value 10 min) | ![]() |
— |
10% of any month (mean value 10 min) | — | ![]() |
2% of any month (mean value 10 min) | — | ![]() |
0.3% of any month (mean value 1 min) | ![]() |
— |
0.05% of any month (mean value 1 s) | ![]() |
— |
0.03% of any month (mean value 1 s) | — | ![]() |
Table 4.2 Overall end-to-end and satellite HRDP error performance objectives for international ISDN connections
Performance classification | Definition | End-to-end objective | Satellite HRDP objectives |
Degraded seconds | Minutes intervals with ![]() |
<10% | <2% |
Severely errored seconds | Minutes intervals with ![]() |
<0.2% | <0.03% |
Errored seconds | Minutes intervals with one or more errors | <0.03% | <1.6% |
Table 4.3 Overall end-to-end and satellite HRDP error performance objectives for digital connection at primary rate or above
Performance classification | Definition | End-to-end objective | Satellite HRDP objectives | ||
Bit rate | — | 1.5 to 15 Mbit/s | 15 to 55 Mbit/s | 1.5 to 15 Mbit/s | 15 to 55 Mbit/s |
Bit per block | — | 2000–8000 | 4000–20000 | 2000–8000 | 4000–20000 |
Errored seconds (ES) ratio (ESR) | ES/t:
|
0.04 | 0.0075 | 0.014 | 0.0262 |
Severe errored seconds (SES) ratio (SESR) | ES/t:
|
0.002 | 0.002 | 0.007 | 0.007 |
Background block errored (BBE) ratio (BBER) | BBE/b
|
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Note: Higher possible rates can also be found in ITU-R S.1062, including 55–160 Mbit/s and 160–3500 Mbit/s.
Mobile wireless ad hoc networks (MANET) has been developed in the recent years. The application of MANET to areas such as emergency rescue, environmental observation, scientific investigations, commercial environments, home and enterprise networking, educational applications, entertainment, military operations and location-aware services holds great promises. The automatic adaptation of the network topology to the field context provided by MANET, make these a great asset in providing local connectivity. However, these applications often happen in infrastructure-less or remote regions where remote connectivity to the outside world has to be provided by some other means.
Satellite is one of the solutions to provide the remote connectivity and sometimes the only option for the MANET to communicate with other parts of the world. Both fixed satellite services (FSS) and mobile satellite services (MSS) can be used to provide broadband connections to the Internet, enabling the exchange of voice, data and video in the field, as well as the relay of control messages and service data from service centres to MANET. Currently, satellite is considered not only as a component of an alternative routing path but also as part of integrated networks. Convergence of satellite and terrestrial networks is becoming a key factor in forming the foundation for efficient global information infrastructures.
On one hand, ad hoc networks are characterised by dynamic topologies, limited bandwidth, energy consumption constraints, limited physical security and distributed management. From the network point of views, the following are the main issues to be considered:
On the other hand, broadband communications via satellite are useful in many different scenarios, especially:
Examples of the benefits of satellite include IP via satellite in regions with no access to terrestrial DSL, new architectures for near-video on demand, satellite networks integrated with WiFi, LTE or TETRA, or satellite networks backhauling collectively mobile networks in ships, trains or airplanes.
The concept of internetworking satellite and MANET networks is therefore a natural evolution to provide local and remote connectivity in a highly mobile, dynamic and often remote environment.
This combination raises, nonetheless, significant challenges in terms of optimising network resources, link availability, providing quality of service (QoS) and quality of experience (QoE; the subjective measure of a customer's experiences with a service supplier) and minimising costs and energy.
The challenges issues include re-organisation of MANET to connect to satellite access points, re-organisation of the satellite access points, selection of which satellite access points to use, the use of satellite as a relay between two MANET, the adjustment of routing in accordance with the current network situation and the exchange of cross layer information to improve resource management.
The concept of internetworking of satellite network and MANET is illustrated in Figure 4.18.
Figure 4.18 Hybrid MANET–satellite network concept
The selected scenarios can be used to explain the concepts of the satellite and MANET networks, associated challenges and potential as shown in figures below. In order to understand the complexity of these heterogeneous, dynamic and distributed environments, we can look into the protocol, functional and network architectures using complementary top-down and bottom-up approaches.
The stochastic movement of the nodes that form a mobile ad-hoc network makes it likely that some partitions may occur in the wireless network without connectivity among them. Both geostationary/non-geostationary satellites can be envisaged as a “range extension” network.
Considerations include the connectivities between nodes in the same ad hoc cluster; between nodes belonging to different ad-hoc networks with the added possibility of nodes using different equipment or technologies. This can be achieved by re-organisation of the MANET to connect to the access points on one hand, or the re-organisation of the access points themselves on the other hand.
For multiple satellites and fixed backbones for connections of the MANET, the trade-off needs to consider of providing higher QoE, minimize communication costs, minimize energy consumption to ensure higher network life become important ones. These are closely related to routing (both unicast and multicast), the current network status (network topology, link quality, node positions, etc.), the availability and characteristics of internet links (upstream/downstream, cost, etc.), and the network usage requirements (priority, QoS, speed, etc.)
Figure 4.19 represents a self-organisation solution by selecting satellite access points either because one access point drops out of the MANET, or so that the transmitting node can choose from several to provide optimal connection. Figure 4.20 represents a solution to keep the connectivity when the initial network is broken down into two MANETs.
Figure 4.19 Hybrid MANET–satellite network challenge: selecting satellite access points
Figure 4.20 Hybrid MANET–satellite challenge: satellite as relay between two MANETs
In addition to the standard routing algorithms for MANET, the satellite and MANET internetworks also consider the access point information, nodes positions and QoS mechanisms. Providing location information may be useful in implementing advanced routing techniques, able to support QoS. Geographical routing protocols can make use of satellite positioning systems, like GPS and Galileo for the nodes to know their own location, so as to share it with the other nodes in the network. There are some MANET routing protocols, such as the ad hoc on-demand distance vector (AODV), and optimised link state routing protocol (OLSR) dynamic source routing (DSR) routing protocols and so on, defined by the Internet Engineering Task Force (IETF).
The complexity of network management will increase between the providers of network connectivity. The autonomous functions operate at the network control layer, facilitating the negotiation of agreements between networks as well as their efficient verification and enforcement. Network management of a MANET including satellite links needs investigation.
Generally, ad hoc network node resources support network formation and management activities, in addition to data communication. The lack of infrastructure in the MANET often dictates the uses of distributed network management. However, the use of the satellite access links may require a centralised management entity that manages the access points and their organisation and hand-over between them. Therefore, the investigation of centralised versus distributed or mixed network management and re-organisation as well as determination of decision mechanisms (manual or automatic) is a challenge issue.
The ideas such as the connection/disconnection of satellite access points (e.g. to save energy or to increase throughput) as well as the choice of satellite access points (horizontal handover) are illustrated in Figure 4.21.
Figure 4.21 Satellite and MANET network: resource management—automatic access point activation to increase throughput
Applications can be significant differences in terms of services required which imply diverse impacts on the network organisation and usage of resources. Data services (such as file transfers, web services, email, etc.) impose low constraints on delay and latency but have high requirements in terms of reliability and integrity.
Voice and video on the other hand have serious requirements in terms of latency and delay (e.g. the high latencies associated with GEO satellites may hamper communications between emergency teams leading to misunderstandings or late reactions resulting in loss of property or life) but are relatively lax in terms of transmission errors (a few errors in the transmission will result in glitches in video or garbled speech) as the human brain has the power to fill in the gaps. Basically, video and voice require real-time exchanges to be useful. The provision of different services in terms of QoS and QoE is differentiated by prioritisation.
Other interesting issues include the impact of GEO versus LEO satellites on the composite network. The latencies of GEO satellites (approximately 240–270 ms, depending on the location of the satellite terminals) may raise difficulties in voice services under certain situations. The LEO delays of 30 ms are in principle more amenable to the requirements of voice and video. The network must be able to perform trade-offs and choose the appropriate satellite service (if both GEO and LEO services are available) according to the desired service.
The network optimisation should consider such as operation and services costs, energy consumption, QoS, availability and so on. This is closely related to the application scenario and is strongly influenced by the application chosen.
Satellites offer a variety of means to accommodate different transmission rates including basic rate, primary rate and high-speed intermediate data rate (IDR). Therefore, there is a significant difference between the interconnections of different networks.
Satellite networks can be used as thin routes between pairs of earth stations, as access networks to provide basic rate and primary rate and as transit networks to interconnect main networks with a capacity measured in thousands of connections.
Transmission bit rate is the physical layer feature of the networks and is only one aspect of interworking issues. There are also high layer protocol issues. Often interworking units have to be introduced to deal with differences of functions at higher layer protocols when interconnecting different type of networks. Here we will discuss some general issues concerning interworking with heterogeneous networks.
Different services are available in heterogeneous networks. For example, video telephony service is supported by the digital network and ordinary telephony is supported by a plain telephony network. If a call is made from one to the other, the video information must be left out for the connection to be successful. Voice services in the video telephony terminal should be able to work as an ordinary telephone terminal because the ordinary telephony service is only a subset of video telephony. Another example is the conversion between email and fax where the interworking functions are more complicated because different terminals providing different services are involved.
Some services do not always need to be able to internetwork with each other such as file transfer, while some services may not be able to internetwork together at all. Normally service level internetwork defines the functional requirement to implement internetworking functions for heterogeneous networks.
In heterogeneous networks, addressing is an important issue to be considered. We have to try to maintain the independent different networking schemes. Each address used to identify a terminal in the network must be unique.
Each internetworking unit of two networks should have two addresses, one of which is used in one network and one for the other network. A mapping between terminals and the internetworking unit must be available so that a terminal can make use of the internetworking unit when trying to connect to a terminal in the other networks. A long-haul connection may traverse many heterogeneous networks from source to destination.
Typical types of address include: Internet address, local area network address (such as Ethernet) and telephone network address (such as telephony number).
Routing is another important issue because the two networks can have significantly different transmission speeds, routing mechanisms, protocol functions and QoS requirements. Therefore, it is important to keep the routing independent within each network. The differences that the internetworking unit has to deal with include the protocol to access the networks, packet and frame format and size, and maintaining the QoS for the end-to-end connection requirement. In addition to the user information to be transported across different types of networks, control signalling and management also have to be considered.
Typical examples of heterogeneous routing can be found in the Internet and IP telephony services where end-to-end connections can traverse across LAN, MAN, telephony, mobile and satellite networks.
Evolution is an important issue for all actors in the area of telecommunications because it predicts the long-term future development of both the network and services. It is different from planning, which concentrates on precise tasks and gives information and detailed figures on actions to be taken in the future.
The driving force of evolution is technical and technological progress, which influence two main areas characterised by mutual interdependences: change and growth. In addition, economic considerations and conditions strongly influence evolution. One issue for future development is the transition from individual sub-networks of different capacities into a single network combining all components to offer even more facilities and capabilities.
Due to flexibility and adaptability, satellite systems have been used in a wide range of network topologies from simple point-to-point connections to complex multipoint-to-multipoint networks and transmission speeds from a few kbit/s to hundreds of Mbit/s. In many cases, satellite networks can present a alternative means of communication with respect to terrestrial networks, and bring advantages from both technical and economic viewpoints.
The functions of satellite networks can be the same as a traditional transmission medium, or provide advanced switching capabilities to work with all types of terrestrial networks.
In an early phase of broadband networks, satellite can be used as a flexible transport mechanism to provide an effective means to link users who cannot be reached by the broadband networks. For some users, satellites will provide an initial access to the broadband networks.
Once the broadband network is well established, satellite can be used to complement the terrestrial networks in areas where installation of other network technologies is difficult or expensive, and to provide services such as broadcast and mobile services to cover the globe.
To meet the challenge, satellite technologies are evolving, which exploit technological advantage by using higher frequency band on-board switching technologies to provide mobile and broadcasting services.
1 Explain these terms: interworking and internetworking.
2 Use a sketch to explain the concept of reference configuration.
3 Explain the different network traffic in user plane, control plane and management plane.
4 Explain the network hypothetical reference connection and related performance objectives for satellite.
5 Explain the basic models and parameters of traffic engineering in telephony networks.
6 Explain the principles of digital networks.
7 Explain different types of signalling schemes and their role in the network.
8 Explain how to calculate the performance objectives of satellite networks in end-to-end reference connections.
9 Discuss the issues of SDH over satellite.
10 Explain the concept of internetworking between the satellite and MANET.