- a summary or tutorial about the basics and theory of the Erlang, the measure of telecommunications traffic, and the Erlang function or formula and the Erlang-B.
The Erlang is widely used in telecommunications technology. The Erlang is a statistical measure of the voice traffic density in a telecommunications system and it is widely used because, for any element in a telecommunications system, whether it is a landline, or uses cellular technology, it is necessary to be able to understand the traffic volume. As a result it is helps to have a definition of the telecommunications traffic so that the volume can be quantified in a standard way and calculations can be made. Telecommunications network designers make great use of the Erlang to understand traffic patterns within a voice network and they use the figures to determine the capacity that is required in any area of the network.
Who was Erlang?
The Erlang is named after a Danish telephone engineer named A.K Erlang (Agner Krarup Erlang). He was born on 1st January 1878 and although he trained as a mathematician, he was the first person to investigate traffic and queuing theory in telephone circuits.
After receiving his MA, Erlang worked in a number of schools. However, Erlang was a member of the Danish Mathematician's Association (TBMI) and it was through this organization that Erlang met the Chief Engineer of the Copenhagen Telephone Company (CTC) and as a result, he went to work for them from 1908 for almost 20 years.
While he was at CTC, Erlang studied the loading on telephone circuits, looking at how many lines were required to provide an acceptable service without installing too much over-capacity that would cost the company money. There was a trade-off between cost and service level.
Erlang developed his theories over a number of years, and published several papers. He expressed his findings in mathematical forms so that they could be used to calculate the required level of capacity, and today the same basic equations are in widespread use..
In view of his groundbreaking work, the International Consultative Committee on Telephones and Telegraphs (CCITT) honoured him in 1946 by adopting the name "Erlang" for the basic unit of telephone traffic.
Erlang died on 3rd February 1929 after an unsuccessful abdominal operation.
Erlang basics
The Erlang is the basic unit of telecommunications traffic intensity representing continuous use of one circuit and it is given the symbol "E". It is effectively call intensity in call minutes per sixty minutes. In general the period of an hour is used, but it actually a dimensionless unit because the dimensions cancel out (i.e. minutes per minute).
The number of Erlangs is easy to deduce in a simple case. If a resource carries one Erlang, then this is equivalent to one continuous call over the period of an hour. Alternatively if two calls were in progress for fifty percent of the time, then this would also equal one Erlang (1E). Alternatively if a radio channel is used for fifty percent of the time carries a traffic level of half an Erlang (0.5E)
From this it can be seen that an Erlang, E, may be thought of as a use multiplier where 100% use is 1E, 200% is 2E, 50% use is 0.5E and so forth.
Interestingly for many years, AT&T and Bell Canada measured traffic in another unit called CCS, 100 call seconds. If figures in CCS are encountered then it is a simple conversion to change CCS to Erlangs. Simply divide the figure in CCS by 36 to obtain the figure in Erlangs
Erlang function or Erlang formula and symbol
It is possible to express the way in which the number of Erlangs are required in the format of a simple function or formula.
A = λ x h
Where:
λ = the mean arrival rate of new calls
h = the mean call length or holding time
A = the traffic in Erlangs.
Using this simple Erlang function or Erlang formula, the traffic can easily be calculated.
Erlang-B and Erlang-C
Erlang calculations are further broken down as follows:
* Erlang B: The Erlang B is used to work out how many lines are required from a knowledge of the traffic figure during the busiest hour. The Erlang B figure assumes that any blocked calls are cleared immediately. This is the most commonly used figure to be used in any telecommunications capacity calculations.
* Extended Erlang B: The Extended Erlang B is similar to Erlang B, but it can be used to factor in the number of calls that are blocked and immediately tried again.
* Erlang C: The Erlang C model assumes that not all calls may be handled immediately and some calls are queued until they can be handled.
These different models are described in further detail below.
Erlang B
It is particularly important to understand the traffic volumes at peak times of the day. Telecommunications traffic, like many other commodities, varies over the course of the day, and also the week. It is therefore necessary to understand the telecommunications traffic at the peak times of the day and to be able to determine the acceptable level of service required. The Erlang B figure is designed to handle the peak or busy periods and to determine the level of service required in these periods.
Erlang C
The Erlang C model is used by call centres to determine how many staff or call stations are needed, based on the number of calls per hour, the average duration of call and the length of time calls are left in the queue. The Erlang C figure is somewhat more difficult to determine because there are more interdependent variables. The Erlang C figure, is nevertheless very important to determine if a call centre is to be set up, as callers do not like being kept waiting interminably, as so often happens.
Erlang summary
The Erlang formulas and the concepts put forward by Erlang are still an essential part of telecommunications network planning these days. As a result, telecommunications engineers should have a good understanding of the Erlang and the associated formulae.
despite the widespread use of the Erlang concepts and formulae, it is necessary to remember that there are limitations to their use. It is necessary to remember that the Erlang formulas make assumptions. Erlang B assumes that callers who receive a busy tone will not immediately try again. Also Erlang C assumes that callers will not hold on indefinitely. It is also worth remembering that the Erlang formulas are based on statistics, and that to make these come true an infinite number of sources is required. However for most cases a total of ten sources gives an adequate number of sources to give sufficiently accurate results.
The Erlang is a particularly important element of telecommunications theory, and it is a cornerstone of many areas of telecommunications technology today. However one must be aware of its limitations and apply the findings of any work using Erlangs, the Erlang B and Erlang C formulas or functions with a certain amount of practical knowledge.
Tuesday, March 16, 2010
E-Carrier, E1 tutorial
- summary or tutorial about the basics of E carrier and E1 system for providing telecommunications links.
The E carrier system has been created by the European Conference of Postal and Telecommunications Administrations (CEPT) as a digital telecommunications carrier scheme for carrying multiple links. The E-carrier system enables the transmission of several (multiplexed) voice/data channels simultaneously on the same transmission facility. Of the various levels of the E-carrier system, the E1 and E3 levels are the only ones that are used.
More specifically E1 has an overall bandwidth of 2048 kbps and provides 32 channels each supporting a data rate of 64 kbps. The lines are mainly used to connect between the PABX (Private Automatic Branch eXchange), and the CO (Central Office) or main exchange.
The E1 standard defines the physical characteristics of a transmission path, and as such it corresponds to the physical layer (layer 1) in the OSI model. Technologies such as ATM and others which form layer 2 are able to pass over E1 lines, making E1 one of the fundamental technologies used within telecommunications.
A similar standard to E1, known as T1 has similar characteristics, but it is widely used in North America. Often equipment used for these technologies, e.g. test equipment may be used for both, and the abbreviation E1/T1 may be seen.
E1 beginnings
The life of the standards started back in the early 1960s when Bell Laboratories, where the transistor was invented some years earlier, developed a voice multiplexing system to enable better use to be made of the lines that were required, and to provide improved performance of the analogue techniques that were used. The step of the process converted the signal into a digital format having a 64 kbps data stream. The next stage is to assemble twenty four of the data streams into a framed data stream with an overall data rate of 1.544 Mbps. This structured signal was called DS1, but it is almost universally referred to as T1.
In Europe, the basic scheme was taken by what was then the CCIT and developed to fit the European requirements better. This resulted in the development of the scheme known as E1. This has provision for 30 voice channels and runs at an overall data rate of 2.048 Mbps. In Europe E1 refers to both the formatted version and the raw data rate.
E1 Applications and standards
The E-carrier standards form part of the overall Synchronous Digital Hierarchy (SDH) scheme. This allows where groups of E1 circuits, each containing 30 circuits, to be combined to produce higher capacity. E1 to E5 are defined and they are carriers in increasing multiples of the E1 format. However in reality only E3 is widely used and this can carry 480 circuits and has an overall capacity of 34.368 Mbps.
Physically E1 is transmitted as 32 timeslots and E3 has 512 timeslots. Unlike Internet data services which are IP based, E-carrier systems are circuit switched and permanently allocate capacity for a voice call for its entire duration. This ensures high call quality because the transmission arrives with the same short delay (Latency) and capacity at all times. Nevertheless it does not allow the same flexibility and efficiency to be obtained as that of an IP based system.
In view of the different capacities of E1 and E3 links they are used for different applications. E1 circuits are widely used to connect to medium and large companies, to telephone exchanges. They may also be used to provide links between some exchanges. E3 lines are used where higher capacity is needed. They are often installed between exchanges, and to provide connectivity between countries.
E1 basics
An E1 link runs over two sets of wires that are normally coaxial cable and the signal itself comprises a nominal 2.4 volt signal. The signalling data rate is 2.048 Mbps full duplex and provides the full data rate in both directions.
For E1, the signal is split into 32 channels each of 8 bits. These channels have their own time division multiplexed slots. These are transmitted sequentially and the complete transmission of the 32 slots makes up a frame. These Time Slots are nominated TS0 to TS31 and they are allocated to different purposes:
* TS0 is used for synchronisation, alarms and messages
* TS1 - TS 15 used for user data
* TS16 is used for signalling, but it may also carry user data
* TS17 - TS31 are used for carrying user data
Time slot 0 is reserved for framing purposes, and alternately transmits a fixed pattern. This allows the receiver to lock onto the start of each frame and match up each channel in turn. The standards allow for a full Cyclic Redundancy Check to be performed across all bits transmitted in each frame.
E1 signalling data is carried on TS16 is reserved for signalling, including control, call setup and teardown. These are accomplished using standard protocols including Channel Associated Signalling (CAS) where a set of bits is used to replicate opening and closing the circuit. Tone signalling may also be used and this is passed through on the voice circuits themselves. More recent systems use Common Channel Signalling (CCS) such as ISDN or Signalling System 7 (SS7) which sends short encoded messages containing call information such as the caller ID.
Several options are specified in the original CEPT standard for the physical transmission of data. However an option or standard known as HDB3 (High-Density Bipolar-3 zeros) is used almost exclusively.
Future
E1 and also T1 are well established for telecommunications use. However with new technologies such as ADSL, DSL, and the other IP based systems that are now being widely deployed, these will spell the end of E1 and T1. Nevertheless they have given good service over many years, and they will remain in use as a result of this wide deployment for some years to come.
The E carrier system has been created by the European Conference of Postal and Telecommunications Administrations (CEPT) as a digital telecommunications carrier scheme for carrying multiple links. The E-carrier system enables the transmission of several (multiplexed) voice/data channels simultaneously on the same transmission facility. Of the various levels of the E-carrier system, the E1 and E3 levels are the only ones that are used.
More specifically E1 has an overall bandwidth of 2048 kbps and provides 32 channels each supporting a data rate of 64 kbps. The lines are mainly used to connect between the PABX (Private Automatic Branch eXchange), and the CO (Central Office) or main exchange.
The E1 standard defines the physical characteristics of a transmission path, and as such it corresponds to the physical layer (layer 1) in the OSI model. Technologies such as ATM and others which form layer 2 are able to pass over E1 lines, making E1 one of the fundamental technologies used within telecommunications.
A similar standard to E1, known as T1 has similar characteristics, but it is widely used in North America. Often equipment used for these technologies, e.g. test equipment may be used for both, and the abbreviation E1/T1 may be seen.
E1 beginnings
The life of the standards started back in the early 1960s when Bell Laboratories, where the transistor was invented some years earlier, developed a voice multiplexing system to enable better use to be made of the lines that were required, and to provide improved performance of the analogue techniques that were used. The step of the process converted the signal into a digital format having a 64 kbps data stream. The next stage is to assemble twenty four of the data streams into a framed data stream with an overall data rate of 1.544 Mbps. This structured signal was called DS1, but it is almost universally referred to as T1.
In Europe, the basic scheme was taken by what was then the CCIT and developed to fit the European requirements better. This resulted in the development of the scheme known as E1. This has provision for 30 voice channels and runs at an overall data rate of 2.048 Mbps. In Europe E1 refers to both the formatted version and the raw data rate.
E1 Applications and standards
The E-carrier standards form part of the overall Synchronous Digital Hierarchy (SDH) scheme. This allows where groups of E1 circuits, each containing 30 circuits, to be combined to produce higher capacity. E1 to E5 are defined and they are carriers in increasing multiples of the E1 format. However in reality only E3 is widely used and this can carry 480 circuits and has an overall capacity of 34.368 Mbps.
Physically E1 is transmitted as 32 timeslots and E3 has 512 timeslots. Unlike Internet data services which are IP based, E-carrier systems are circuit switched and permanently allocate capacity for a voice call for its entire duration. This ensures high call quality because the transmission arrives with the same short delay (Latency) and capacity at all times. Nevertheless it does not allow the same flexibility and efficiency to be obtained as that of an IP based system.
In view of the different capacities of E1 and E3 links they are used for different applications. E1 circuits are widely used to connect to medium and large companies, to telephone exchanges. They may also be used to provide links between some exchanges. E3 lines are used where higher capacity is needed. They are often installed between exchanges, and to provide connectivity between countries.
E1 basics
An E1 link runs over two sets of wires that are normally coaxial cable and the signal itself comprises a nominal 2.4 volt signal. The signalling data rate is 2.048 Mbps full duplex and provides the full data rate in both directions.
For E1, the signal is split into 32 channels each of 8 bits. These channels have their own time division multiplexed slots. These are transmitted sequentially and the complete transmission of the 32 slots makes up a frame. These Time Slots are nominated TS0 to TS31 and they are allocated to different purposes:
* TS0 is used for synchronisation, alarms and messages
* TS1 - TS 15 used for user data
* TS16 is used for signalling, but it may also carry user data
* TS17 - TS31 are used for carrying user data
Time slot 0 is reserved for framing purposes, and alternately transmits a fixed pattern. This allows the receiver to lock onto the start of each frame and match up each channel in turn. The standards allow for a full Cyclic Redundancy Check to be performed across all bits transmitted in each frame.
E1 signalling data is carried on TS16 is reserved for signalling, including control, call setup and teardown. These are accomplished using standard protocols including Channel Associated Signalling (CAS) where a set of bits is used to replicate opening and closing the circuit. Tone signalling may also be used and this is passed through on the voice circuits themselves. More recent systems use Common Channel Signalling (CCS) such as ISDN or Signalling System 7 (SS7) which sends short encoded messages containing call information such as the caller ID.
Several options are specified in the original CEPT standard for the physical transmission of data. However an option or standard known as HDB3 (High-Density Bipolar-3 zeros) is used almost exclusively.
Future
E1 and also T1 are well established for telecommunications use. However with new technologies such as ADSL, DSL, and the other IP based systems that are now being widely deployed, these will spell the end of E1 and T1. Nevertheless they have given good service over many years, and they will remain in use as a result of this wide deployment for some years to come.
Asynchronous Transfer Mode ATM Tutorial
- an overview or tutorial of the basics of asynchronous transfer mode, used in many data networking applications large and small.
The Asynchronous Transfer Mode (ATM) was developed to enable a single data networking standard to be used for both synchronous channel networking and packet-based networking. Asynchrnonous transfer mode also supports multiple levels of quality of service for packet traffic.
In this way, asynchronous transfer mode can be thought of as supporting both circuit-switched networks and packet-switched networks by mapping both bitstreams and packet-streams. It achieves this by sending data in a series or stream of fixed length cells, each of which has its own identifier. These data cells are typically sent on demand within a synchronous time-slot pattern in a synchronous bit-stream. Although this may not appear to be asynchronous, the asynchronous element of the "Asynchronous Transfer Mode", comes from the fact that the sending of the cells themselves is asynchronous and not from the synchronous low-level bitstream that carries them.
One of the original aims of Asynchronous Transfer Mode was that it should provide a basis for Broadband Integrated Services Digital Network (B-ISDN) to replace existing PSTN (Private � ). As a result of this the standards for Asynchronous Transfer Mode standards include not only the definitions for the Physical transmission techniques (Layer 1), but also layers 2 and 3.
In addition to this, the development of Asysnchronous Transfer Mode was focussed heavily on the requirements for telecommunications providers rather than local data networking requirements, and as a result it is more suited to large area telecommunications applications rather than smaller local area data network solutions, or general computer networking.
While Asynchronous Transfer Mode is widely used for many applications, it is generally only used for transport of IP traffic. It has not become the single standard for providing a single integrated technology for LANs, public networks, and user services.
Basic asynchronous transfer mode system
There are two basic elements to an ATM system. Any system can be made up a number of each of these elements:
* ATM switch This accepts the incoming cells or information "packets" from another ATM entity which may be either another switch or an end point. It reads and updates the cell header information and switches the information cell towards its destination
* ATM end point This element contains the ATM network interface adaptor to enable data entering or leaving the ATM network to interface to the external world. Examples of these end points include workstations, LAN switches, video codecs and many more items.
ATM networks can be configured in many ways. The overall network will comprise a set of ATM switches interconnected by point-to-point ATM links or interfaces. Within the network there are two types of interface and these are both supported by the switches. The first is UNI and this is used to connect ATM end systems (such as hosts and routers) to an ATM switch. The second type of interface is known as NNI. This connects two ATM switches.
ATM operation
In ATM the information is formatted into fixed length cells consisting of 48 bytes (each 8 bits long) of payload data. In addition to this there is a cell header that consists of 5 bytes, giving a total cell length of 53 bytes. This format has been chosen because time critical data such as voice packets is not affected by very long packets being sent. The data carried in the header comprises payload information as well as what are termed virtual-circuit identifiers and header error check data.
ATM is what is termed connection orientated. This has the advantage that the user can define the requirements that are needed to support the calls, and in turn this allows the network to allocated the required resources. By adopting this approach, several calls can be multiplexed efficiently and ensuring that the required resources can be allocated.
There are two types of connection that are specified for asynchronous transfer mode:
* Virtual Channel Connections - this is the basic connection unit or entity. It carries a single stream of data cells from the originator to the end user.
* Virtual Path Connections - this is formed from a collection of virtual channel connections. A virtual path is an end to end connection created across an ATM (asynchronous transfer mode) network. For a virtual path connection, the network routes all cells from the virtual path across the network in the same way without regard for the individual virtual circuit connection. This results in faster transfer.
The idea of virtual path connections are also used within the ATM network itself to route traffic between switches
ATM networks can be configured in many ways. The overall network will comprise a set of ATM switches interconnected by point-to-point ATM links or interfaces. Within the network there are two types of interface and these are both supported by the switches. The first is UNI and this is used to connect ATM end systems (such as hosts and routers) to an ATM switch. The second type of interface is known as NNI. This connects two ATM switches.
The Asynchronous Transfer Mode (ATM) was developed to enable a single data networking standard to be used for both synchronous channel networking and packet-based networking. Asynchrnonous transfer mode also supports multiple levels of quality of service for packet traffic.
In this way, asynchronous transfer mode can be thought of as supporting both circuit-switched networks and packet-switched networks by mapping both bitstreams and packet-streams. It achieves this by sending data in a series or stream of fixed length cells, each of which has its own identifier. These data cells are typically sent on demand within a synchronous time-slot pattern in a synchronous bit-stream. Although this may not appear to be asynchronous, the asynchronous element of the "Asynchronous Transfer Mode", comes from the fact that the sending of the cells themselves is asynchronous and not from the synchronous low-level bitstream that carries them.
One of the original aims of Asynchronous Transfer Mode was that it should provide a basis for Broadband Integrated Services Digital Network (B-ISDN) to replace existing PSTN (Private � ). As a result of this the standards for Asynchronous Transfer Mode standards include not only the definitions for the Physical transmission techniques (Layer 1), but also layers 2 and 3.
In addition to this, the development of Asysnchronous Transfer Mode was focussed heavily on the requirements for telecommunications providers rather than local data networking requirements, and as a result it is more suited to large area telecommunications applications rather than smaller local area data network solutions, or general computer networking.
While Asynchronous Transfer Mode is widely used for many applications, it is generally only used for transport of IP traffic. It has not become the single standard for providing a single integrated technology for LANs, public networks, and user services.
Basic asynchronous transfer mode system
There are two basic elements to an ATM system. Any system can be made up a number of each of these elements:
* ATM switch This accepts the incoming cells or information "packets" from another ATM entity which may be either another switch or an end point. It reads and updates the cell header information and switches the information cell towards its destination
* ATM end point This element contains the ATM network interface adaptor to enable data entering or leaving the ATM network to interface to the external world. Examples of these end points include workstations, LAN switches, video codecs and many more items.
ATM networks can be configured in many ways. The overall network will comprise a set of ATM switches interconnected by point-to-point ATM links or interfaces. Within the network there are two types of interface and these are both supported by the switches. The first is UNI and this is used to connect ATM end systems (such as hosts and routers) to an ATM switch. The second type of interface is known as NNI. This connects two ATM switches.
ATM operation
In ATM the information is formatted into fixed length cells consisting of 48 bytes (each 8 bits long) of payload data. In addition to this there is a cell header that consists of 5 bytes, giving a total cell length of 53 bytes. This format has been chosen because time critical data such as voice packets is not affected by very long packets being sent. The data carried in the header comprises payload information as well as what are termed virtual-circuit identifiers and header error check data.
ATM is what is termed connection orientated. This has the advantage that the user can define the requirements that are needed to support the calls, and in turn this allows the network to allocated the required resources. By adopting this approach, several calls can be multiplexed efficiently and ensuring that the required resources can be allocated.
There are two types of connection that are specified for asynchronous transfer mode:
* Virtual Channel Connections - this is the basic connection unit or entity. It carries a single stream of data cells from the originator to the end user.
* Virtual Path Connections - this is formed from a collection of virtual channel connections. A virtual path is an end to end connection created across an ATM (asynchronous transfer mode) network. For a virtual path connection, the network routes all cells from the virtual path across the network in the same way without regard for the individual virtual circuit connection. This results in faster transfer.
The idea of virtual path connections are also used within the ATM network itself to route traffic between switches
ATM networks can be configured in many ways. The overall network will comprise a set of ATM switches interconnected by point-to-point ATM links or interfaces. Within the network there are two types of interface and these are both supported by the switches. The first is UNI and this is used to connect ATM end systems (such as hosts and routers) to an ATM switch. The second type of interface is known as NNI. This connects two ATM switches.
Wireless technologies
- including Bluetooth, IEEE 802.11 (Wi-Fi), WiMax, Zigbee (IEEE 802.15.4), etc
Wireless technology in a variety of forms is an area of electronics that is developing and growing particularly fast. Wirless LAN (WLAN) technology including Wi-Fi (IEEE 802.11), Bluetooth, Ultra-Wide Band (UWB), Wimax, Zigbee, and more are all growing and finding their own market areas. As a result wireless technology is being more widely used and found in many new applications.
The development of wireless technologies
Wireless technologies and standards
Wireless technology is being widely used for many applications. Accordingly there is a growing number of different wireless technologies and standards that are being used. Multipage summaries of many of the more widely used wireless standards are given below.
* Bluetooth
* DECT
* HomeRF SWAP (now obsolete)
* IEEE 802.11 Wi-Fi standard
* IEEE 802.22 WRAN standard
* NFC Near Field Communications
* Short range device, SRD
* Ultra Wideband Technology
* Wibree
* WiMAX
* Wireless USB - a technlogy utilising UWB transmissions
* Zigbee standard
Wireless Analysis
Our contributions from industry experts on various aspects of wire;ess technology looking at technology trends, case studies and analysis of the current wireless technology situation.
* Real world NFC applications
Wireless technology in a variety of forms is an area of electronics that is developing and growing particularly fast. Wirless LAN (WLAN) technology including Wi-Fi (IEEE 802.11), Bluetooth, Ultra-Wide Band (UWB), Wimax, Zigbee, and more are all growing and finding their own market areas. As a result wireless technology is being more widely used and found in many new applications.
The development of wireless technologies
Wireless technologies and standards
Wireless technology is being widely used for many applications. Accordingly there is a growing number of different wireless technologies and standards that are being used. Multipage summaries of many of the more widely used wireless standards are given below.
* Bluetooth
* DECT
* HomeRF SWAP (now obsolete)
* IEEE 802.11 Wi-Fi standard
* IEEE 802.22 WRAN standard
* NFC Near Field Communications
* Short range device, SRD
* Ultra Wideband Technology
* Wibree
* WiMAX
* Wireless USB - a technlogy utilising UWB transmissions
* Zigbee standard
Wireless Analysis
Our contributions from industry experts on various aspects of wire;ess technology looking at technology trends, case studies and analysis of the current wireless technology situation.
* Real world NFC applications
Download Lagu - Wali Band [Yank]
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Lirik lagu:
Yang.. Coba kau jujur padaku
Yang.. Foto siapa di dompetmu
Yang… Kok kamu diam begitu
Sayang jawab atau aku pergi sayang
#
Aku tak mau bicara
Sebelum kau cerita semua
Apa maumu, siapa dirinya
Tak betah bila ada yang lain
Jangan hubungi ku lagi
Ini bisa jadi yang terakhir
Aku ngerti kamu
Kau tak ngerti aku
Sekarang atau tak selamanya
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Salahmu mau bicara
Dan ku akan cerita semua
Apa mauku, siapa dirinya
Karna memang tak ada yang lain
Terus hubungiku lagi
Jangan bilang ini yang terakhir
Aku ngerti kamu
Kamu ngerti aku
Aku sayang kamu selamanya. [Source...]
Monday, March 15, 2010
What is IPTV Technology
- information, or tutorial about the basics of what is IPTV, television using Internet Protocol, and distributed over an IP network such as the Internet.
There are now many ways in which television material can be delivered these days. One method that is set to make a huge impact is IPTV. As the name indicates, IPTV, or Internet Protocol Television, is based around the use of Internet Protocol, and this means that the service tends to be used in conjunction with telecommunications services of which broadband internet lines are the most widely used, although any service that can carry packet data can be used.
IPTV has many advantages. It provides a potentially lucrative market in view many telecommunications providers are looking keenly at the possibilities of providing new services and generating new revenue streams. This means that in the years to come IPTV seems certain to become a major player in the entertainment market.
IPTV Basics
There are two basic ways in which the material for IPTV can be handled. It can be sent out as broadcast or "multicast" material to many users simultaneously or it can be used to provide video-on-demand where the material is sent to just one subscriber who has requested that particular item.
It is also necessary to ensure that bandwidth is used efficiently. To achieve this, the IPTV video is compressed and two main formats are used for this. One is MPEG-2, and the other is the newer MPEG-4 standard. Of these the MPEG-4 standard is being used increasingly in view of its superior performance.
Basic IPTV system
For the user, an IPTV system is based around a set top box or an equivalent such as a computer. This receives the incoming data and re-assembles the data packets and decodes them to provide the required output for passing on to a television or other form of screen for viewing.
The system at the service provider end is considerably more complicated and requires a number of elements to ensure that the system operates correctly under both unicast and multicast conditions.
The core for any IPTV system is the operator's central distribution centre. Here the material is assembled and encoded.
Once encoded, the video stream is split into packets, so that it can enter the IP network system and routed to the relevant destination or destinations. The video streams consisting of these packets then travel from the centre of the network to outlying local exchanges and routing centres before being routed on to the individual subscribers. It is typically at these local centres that where authentication, channel change requests, billing and video-on-demand requests are handled.
Channel selection
One important aspect of multicasting is the ability of the system to be able to select the required channel and change it as required. In order that vast amounts of data are not set to the set top box, or any other form of IP receiver, one a single channel is sent to a given receiver.
Channel selection is accomplished by using a special protocol known as IP Group Membership Protocol (IPGMP). This is a communications protocol used to manage the membership of Internet Protocol multicast groups. It is used by IP hosts and adjacent multicast routers to establish multicast group memberships.
When a local routing centre receives a request to receive multicast data or to change from one channel to another, it checks to ensure that the user is authorised to connect to the channel and then it directs its routers to add the particular user to its distribution list. In this way only the channels that are being used are routed to the receiver, and this saves enormous amounts of data.
Unicast / video on demand
In order that users can connect to a video on demand service a different set-up is needed. The local server operates in a fashion that enables single streams of data to be drawn off as required and it is controlled using a different protocol to that used for multicasting is used. Known as Real Time Streaming Protocol, RTSP, it controls the data stream and allows DVD-style control over the media stream, enabling the user to play, pause and stop the programme being watched.
Quality of Service (QoS)
One of the major elements of any IP network is the control of the Quality of Service. To enable the video stream to be maintained and to prevent delays and fragmentation of the packets on the network, it is necessary to ensure that the Quality of Service is maintained. The Quality of Service tags assign the required priority to different packets on the network, and in this way enable time sensitive packets such as live video or voice traffic to have a higher priority than those such as Internet browsing where delays can be tolerated. By using QoS tags and tools, the required data streams can be transmitted to make maximum use of the available bandwidth, while still maintaining the required level of performance.
Error correction
Errors will always occur on any network. These need to be detected and corrected to enable the system to work satisfactorily. Error correction needs to be handled in different ways according to the type of broadcast. For Video on Demand, Unicast services it is relatively simple for the receiver to request a re-send of a packet if errors are detected. However this cannot be done for a multicast system as it is not possible to request missing or corrupted packets to be re-sent. It is essential that the integrity of any network carrying multicast data is high so that few errors occur. In addition to this a variety of error correction measures are incorporated into the system. These include forward error correction (FEC) techniques that enable the receiver to rebuild missing or corrupted data.
Summary
The use of IPTV is on the increase, and with more people using broadband connections to access an ever increasing number of services, it seems likely that IP-TV technology will grow in importance over the coming years. The system offers many advantages including the use of VOD which makes it particularly attractive to many. When bundled with other services, IPTV will offer a cost effective way of viewing for many users and in addition to this it will provide many telecommunications providers with new and lucrative services to offer. These advantages seem to provide IPTV with a compelling business case.
There are now many ways in which television material can be delivered these days. One method that is set to make a huge impact is IPTV. As the name indicates, IPTV, or Internet Protocol Television, is based around the use of Internet Protocol, and this means that the service tends to be used in conjunction with telecommunications services of which broadband internet lines are the most widely used, although any service that can carry packet data can be used.
IPTV has many advantages. It provides a potentially lucrative market in view many telecommunications providers are looking keenly at the possibilities of providing new services and generating new revenue streams. This means that in the years to come IPTV seems certain to become a major player in the entertainment market.
IPTV Basics
There are two basic ways in which the material for IPTV can be handled. It can be sent out as broadcast or "multicast" material to many users simultaneously or it can be used to provide video-on-demand where the material is sent to just one subscriber who has requested that particular item.
It is also necessary to ensure that bandwidth is used efficiently. To achieve this, the IPTV video is compressed and two main formats are used for this. One is MPEG-2, and the other is the newer MPEG-4 standard. Of these the MPEG-4 standard is being used increasingly in view of its superior performance.
Basic IPTV system
For the user, an IPTV system is based around a set top box or an equivalent such as a computer. This receives the incoming data and re-assembles the data packets and decodes them to provide the required output for passing on to a television or other form of screen for viewing.
The system at the service provider end is considerably more complicated and requires a number of elements to ensure that the system operates correctly under both unicast and multicast conditions.
The core for any IPTV system is the operator's central distribution centre. Here the material is assembled and encoded.
Once encoded, the video stream is split into packets, so that it can enter the IP network system and routed to the relevant destination or destinations. The video streams consisting of these packets then travel from the centre of the network to outlying local exchanges and routing centres before being routed on to the individual subscribers. It is typically at these local centres that where authentication, channel change requests, billing and video-on-demand requests are handled.
Channel selection
One important aspect of multicasting is the ability of the system to be able to select the required channel and change it as required. In order that vast amounts of data are not set to the set top box, or any other form of IP receiver, one a single channel is sent to a given receiver.
Channel selection is accomplished by using a special protocol known as IP Group Membership Protocol (IPGMP). This is a communications protocol used to manage the membership of Internet Protocol multicast groups. It is used by IP hosts and adjacent multicast routers to establish multicast group memberships.
When a local routing centre receives a request to receive multicast data or to change from one channel to another, it checks to ensure that the user is authorised to connect to the channel and then it directs its routers to add the particular user to its distribution list. In this way only the channels that are being used are routed to the receiver, and this saves enormous amounts of data.
Unicast / video on demand
In order that users can connect to a video on demand service a different set-up is needed. The local server operates in a fashion that enables single streams of data to be drawn off as required and it is controlled using a different protocol to that used for multicasting is used. Known as Real Time Streaming Protocol, RTSP, it controls the data stream and allows DVD-style control over the media stream, enabling the user to play, pause and stop the programme being watched.
Quality of Service (QoS)
One of the major elements of any IP network is the control of the Quality of Service. To enable the video stream to be maintained and to prevent delays and fragmentation of the packets on the network, it is necessary to ensure that the Quality of Service is maintained. The Quality of Service tags assign the required priority to different packets on the network, and in this way enable time sensitive packets such as live video or voice traffic to have a higher priority than those such as Internet browsing where delays can be tolerated. By using QoS tags and tools, the required data streams can be transmitted to make maximum use of the available bandwidth, while still maintaining the required level of performance.
Error correction
Errors will always occur on any network. These need to be detected and corrected to enable the system to work satisfactorily. Error correction needs to be handled in different ways according to the type of broadcast. For Video on Demand, Unicast services it is relatively simple for the receiver to request a re-send of a packet if errors are detected. However this cannot be done for a multicast system as it is not possible to request missing or corrupted packets to be re-sent. It is essential that the integrity of any network carrying multicast data is high so that few errors occur. In addition to this a variety of error correction measures are incorporated into the system. These include forward error correction (FEC) techniques that enable the receiver to rebuild missing or corrupted data.
Summary
The use of IPTV is on the increase, and with more people using broadband connections to access an ever increasing number of services, it seems likely that IP-TV technology will grow in importance over the coming years. The system offers many advantages including the use of VOD which makes it particularly attractive to many. When bundled with other services, IPTV will offer a cost effective way of viewing for many users and in addition to this it will provide many telecommunications providers with new and lucrative services to offer. These advantages seem to provide IPTV with a compelling business case.
IPTV ─ Technology Overview
The real-time nature of IPTV service prevents (in most cases) the network from performing retransmissions to correct errors; the end user’s perceived quality of experience (QoE) may therefore be affected to various degrees. Independent studies have shown that contrary to voice service customers, IPTV subscribers, are not expected to compromise on the quality of their service, thus the signal quality across the IPTV network must be routinely tested or monitored to minimize and quickly resolve potential threats to service revenue.
Figure 1. Eliminating video disruption and pixelization is the key to retaining subscribers.
IPTV Network Topology
IPTV technology is part of a new breed of services designed to facilitate access to video entertainment. It provides access to digital TV over the IP transport medium from a head-end device to the end user’s TV set-top box (STB). Most service providers use a dedicated transport network to support IPTV.
A typical IPTV network is comprised of the following functional blocks (see figure below):
* National head-end: Where most of the IPTV channels enter the network from national broadcasters
* Core network: Usually an IP/MPLS network transporting traffic to the access network
* Access network: Distributes the IPTV streams to the DSLAMs
* Regional head-end: Where local content is added to the network
* Customer premises: Where the IPTV stream is terminated and viewed
Figure 2: General IPTV Network Architecture
Broadcast information coming from an antenna or a satellite dish at the national head-end is mainly distributed using MPEG-2 multiprogram transport stream (or MPTS) to the video service node. Note that other more efficient, less bandwidth-hungry compression algorithms such as H.264 (MPEG-4 Part 10) or the Society of Motion Picture and Television Engineers (SMPTE) 421M (also known as VC-1) are making their way to the marketplace to complement this first offering.
The distribution of the actual SDTV or HDTV channel content is performed using various devices on the access network. Among these devices, digital subscriber line access multiplexers (DSLAM) as well as other technologies like fiber-to-the-home (FTTH) can be used to interface with the user’s STB. For IPTV, each channel is distributed using a multicast IP address.
Factors Affecting Service
Encoding and Compression The quality of the video being distributed across the network can be affected right at the source; i.e., at the video head-end. The encoding and compression process usually creates a trade-off between the quality of the video and the desired compression level.
Jitter Defined as a short-term variation in the packet arrival time, typically caused by network or server congestion. If the Ethernet frames arrive at the STB at a rate that is slower or faster, as determined by the network conditions, buffering is required to help smooth out the variations. Based on the size of the buffer, there are delivery conditions that can make the buffer overflow or underflow, which results in a degradation of the perceived video.
Limited Bandwidth As core IP infrastructure is usually based on optical networks with a low level of congestion, bandwidth limitations (and the total amount of video-stream data that can be sent) is limited mostly by the access network or the customer’s home network supported rate. When traffic levels hit the maximum bandwidth available, packets are discarded, leading to video quality degradation.
Packet Loss Loss of IP packets may occur for multiple reasons—bandwidth limitations, network congestion, failed links and transmission errors. Packet loss usually presents a bursty behavior, commonly related to periods of network congestion.
Quality of Experience (QoE)
Due to the structure of Ethernet and IP networks, the quality of the video/audio traffic is primarily influenced by network jitter and packet loss. With the type of video encoding that is used in MPEG or other similar compression algorithms, the actual impact to the user perception depends on the packet type that is lost in the network. In MPEG-2, the transported packets that are used to form an image are divided into I-frames, P-frames and B-frames. In simple terms, I-frames contain a complete image, while P-frames and B-frames contain predicted information from the other frames.
Figure 3. Typical group of picture (GOP) relationship in MPEG
Figure 3 provides a sample of the relationships between the various types of frames included in a group of picture (GOP). As shown, I-frames are independent and provide input to support the other frames; this means that an error in the I-frames will have more repercussions to the image being viewed than losing P-frames or B-frames.
Key QoE Parameters
Several metrics exist to quantify the impact of the network on the quality of the channel that is received by the end user. The most popular parameters are media delivery index (MDI) as well as PCR jitter for MPEG-2 TS. Other parameters are also used in the IPTV network, but they typically require further packet inspection to collect the information necessary for deeper analysis.
IPTV is an evolving technology and it is not completely driven by specific standards for testing and monitoring. However, the aforementioned parameters must be measured as a first alert to help qualify the user’s quality of experience (QoE) of the service delivered by the network over which IPTV services are being transported.
Media Delivery Index (MDI) as a Testing Metric
The nature of an IPTV service has inherent characteristics that are the primary drivers affecting the quality of the image being viewed; namely, bandwidth availability, packet loss and jitter. The use of MDI as a testing metric provides users the tools to measure and diagnose network-induced impairments for IPTV streaming media. MDI is the only standards-based (RCF-4445) video-quality metric available today and it is endorsed by the IP Video Quality Alliance.
MDI is comprised of two distinct measurements: delay factor (DF) and media loss rate (MLR), which together provide a QoS measure of the delivered media stream that can be directly correlated to the end users’ ultimate quality of experience.
Some of the key benefits of using MDI:
* MDI does not perform any type of stream decoding to achieve its metrics and therefore does not require significant real-time processing power.
* MDI can be used with encrypted media payloads.
* MDI is not dependent on any one type of video-encoding technique, so it can easily be scaled to monitor video quality on hundreds of simultaneous channels.
* MDI is typically sampled at multiple points throughout the stream path with the measurements serving as indicators of problems in the network that can be proactively addressed before they become service-affecting issues.
* Since MDI relies on transport-layer metrics (DF and MLR), it can be used to set network margins and it directly correlates to impending network problems with respect to video quality.
* Since MDI uses packet-level metrics, it helps validate the performance of network equipment such as switches and routers that play a key role in determining whether a packet is delayed or dropped.
Figure 4. Typical core-to-access IPTV testing application with media delivery index (MDI) measurement across the network.
Figure 1. Eliminating video disruption and pixelization is the key to retaining subscribers.
IPTV Network Topology
IPTV technology is part of a new breed of services designed to facilitate access to video entertainment. It provides access to digital TV over the IP transport medium from a head-end device to the end user’s TV set-top box (STB). Most service providers use a dedicated transport network to support IPTV.
A typical IPTV network is comprised of the following functional blocks (see figure below):
* National head-end: Where most of the IPTV channels enter the network from national broadcasters
* Core network: Usually an IP/MPLS network transporting traffic to the access network
* Access network: Distributes the IPTV streams to the DSLAMs
* Regional head-end: Where local content is added to the network
* Customer premises: Where the IPTV stream is terminated and viewed
Figure 2: General IPTV Network Architecture
Broadcast information coming from an antenna or a satellite dish at the national head-end is mainly distributed using MPEG-2 multiprogram transport stream (or MPTS) to the video service node. Note that other more efficient, less bandwidth-hungry compression algorithms such as H.264 (MPEG-4 Part 10) or the Society of Motion Picture and Television Engineers (SMPTE) 421M (also known as VC-1) are making their way to the marketplace to complement this first offering.
The distribution of the actual SDTV or HDTV channel content is performed using various devices on the access network. Among these devices, digital subscriber line access multiplexers (DSLAM) as well as other technologies like fiber-to-the-home (FTTH) can be used to interface with the user’s STB. For IPTV, each channel is distributed using a multicast IP address.
Factors Affecting Service
Encoding and Compression The quality of the video being distributed across the network can be affected right at the source; i.e., at the video head-end. The encoding and compression process usually creates a trade-off between the quality of the video and the desired compression level.
Jitter Defined as a short-term variation in the packet arrival time, typically caused by network or server congestion. If the Ethernet frames arrive at the STB at a rate that is slower or faster, as determined by the network conditions, buffering is required to help smooth out the variations. Based on the size of the buffer, there are delivery conditions that can make the buffer overflow or underflow, which results in a degradation of the perceived video.
Limited Bandwidth As core IP infrastructure is usually based on optical networks with a low level of congestion, bandwidth limitations (and the total amount of video-stream data that can be sent) is limited mostly by the access network or the customer’s home network supported rate. When traffic levels hit the maximum bandwidth available, packets are discarded, leading to video quality degradation.
Packet Loss Loss of IP packets may occur for multiple reasons—bandwidth limitations, network congestion, failed links and transmission errors. Packet loss usually presents a bursty behavior, commonly related to periods of network congestion.
Quality of Experience (QoE)
Due to the structure of Ethernet and IP networks, the quality of the video/audio traffic is primarily influenced by network jitter and packet loss. With the type of video encoding that is used in MPEG or other similar compression algorithms, the actual impact to the user perception depends on the packet type that is lost in the network. In MPEG-2, the transported packets that are used to form an image are divided into I-frames, P-frames and B-frames. In simple terms, I-frames contain a complete image, while P-frames and B-frames contain predicted information from the other frames.
Figure 3. Typical group of picture (GOP) relationship in MPEG
Figure 3 provides a sample of the relationships between the various types of frames included in a group of picture (GOP). As shown, I-frames are independent and provide input to support the other frames; this means that an error in the I-frames will have more repercussions to the image being viewed than losing P-frames or B-frames.
Key QoE Parameters
Several metrics exist to quantify the impact of the network on the quality of the channel that is received by the end user. The most popular parameters are media delivery index (MDI) as well as PCR jitter for MPEG-2 TS. Other parameters are also used in the IPTV network, but they typically require further packet inspection to collect the information necessary for deeper analysis.
IPTV is an evolving technology and it is not completely driven by specific standards for testing and monitoring. However, the aforementioned parameters must be measured as a first alert to help qualify the user’s quality of experience (QoE) of the service delivered by the network over which IPTV services are being transported.
Media Delivery Index (MDI) as a Testing Metric
The nature of an IPTV service has inherent characteristics that are the primary drivers affecting the quality of the image being viewed; namely, bandwidth availability, packet loss and jitter. The use of MDI as a testing metric provides users the tools to measure and diagnose network-induced impairments for IPTV streaming media. MDI is the only standards-based (RCF-4445) video-quality metric available today and it is endorsed by the IP Video Quality Alliance.
MDI is comprised of two distinct measurements: delay factor (DF) and media loss rate (MLR), which together provide a QoS measure of the delivered media stream that can be directly correlated to the end users’ ultimate quality of experience.
Some of the key benefits of using MDI:
* MDI does not perform any type of stream decoding to achieve its metrics and therefore does not require significant real-time processing power.
* MDI can be used with encrypted media payloads.
* MDI is not dependent on any one type of video-encoding technique, so it can easily be scaled to monitor video quality on hundreds of simultaneous channels.
* MDI is typically sampled at multiple points throughout the stream path with the measurements serving as indicators of problems in the network that can be proactively addressed before they become service-affecting issues.
* Since MDI relies on transport-layer metrics (DF and MLR), it can be used to set network margins and it directly correlates to impending network problems with respect to video quality.
* Since MDI uses packet-level metrics, it helps validate the performance of network equipment such as switches and routers that play a key role in determining whether a packet is delayed or dropped.
Figure 4. Typical core-to-access IPTV testing application with media delivery index (MDI) measurement across the network.
Mobile Broadband: to WiMAX or not to WiMAX?
When it comes to mobile broadband, there’s only one thing on operators’ minds: how.
Ask any operator in any market and they’ll tell you basically the same thing: the future for mobile broadband looks bright indeed.
From Asia, to Africa, Europe to the Americas, operators are not-so-secretly preparing for this massive opportunity - and challenge to their networks.
Unprecedented growth
According to GSMA figures published in April, worldwide mobile broadband connections jumped from three million in March 2007 to thirty-two million in March 2008, and by all accounts this is just the beginning.
What divides opinion is the protocol that people think will come to dominate as mobile broadband reaches maturity: should operators put their money on WiMAX or Long Term Evolution (LTE)?
According to a recent article published by Network World, "There are likely years of LTE-vs.-WiMAX technology arguments ahead [...] In the meantime, there are legitimate reasons to be interested in the progress of both."
Indeed, both protocols hold huge promise for operators, who will need to judge which protocol can be most effective given their particular business ecosystem, customer base, and legacy equipment they already have in operation and want to build from.
Can WiMAX and LTE just get along?
Nokia Siemens Networks has been at the forefront of both WiMAX and LTE, and was the first supplier in the world to demonstrate LTE with data speeds up to 160 Mb/s all the way back in 2006.
LTE offers an open and global specification for next generation networks that will bring scale and efficiencies for vendors, choice in network equipment and devices for operators, and ultimately lower costs and more benefits to consumers.
But WiMAX will continue to dominate, for the time being at least. Says Network World: "Globally, there are 305 service providers deploying WiMAX services in 118 countries," citing the WiMAX Forum, which last week certified 10 initial mobile WiMAX products in the 2.5GHz worldwide spectrum for interoperability.
"The forum expects to certify 100 mobile WiMAX products by year-end," according to Network World.
A short-term, long-term approach
To be sure, WiMAX has plenty to offer in terms of efficiency and practicality. But wise operators will keep their options open, taking advantage of WiMAX in the short term but keeping an eye on LTE as it comes into its own over the next few years.
Ask any operator in any market and they’ll tell you basically the same thing: the future for mobile broadband looks bright indeed.
From Asia, to Africa, Europe to the Americas, operators are not-so-secretly preparing for this massive opportunity - and challenge to their networks.
Unprecedented growth
According to GSMA figures published in April, worldwide mobile broadband connections jumped from three million in March 2007 to thirty-two million in March 2008, and by all accounts this is just the beginning.
What divides opinion is the protocol that people think will come to dominate as mobile broadband reaches maturity: should operators put their money on WiMAX or Long Term Evolution (LTE)?
According to a recent article published by Network World, "There are likely years of LTE-vs.-WiMAX technology arguments ahead [...] In the meantime, there are legitimate reasons to be interested in the progress of both."
Indeed, both protocols hold huge promise for operators, who will need to judge which protocol can be most effective given their particular business ecosystem, customer base, and legacy equipment they already have in operation and want to build from.
Can WiMAX and LTE just get along?
Nokia Siemens Networks has been at the forefront of both WiMAX and LTE, and was the first supplier in the world to demonstrate LTE with data speeds up to 160 Mb/s all the way back in 2006.
LTE offers an open and global specification for next generation networks that will bring scale and efficiencies for vendors, choice in network equipment and devices for operators, and ultimately lower costs and more benefits to consumers.
But WiMAX will continue to dominate, for the time being at least. Says Network World: "Globally, there are 305 service providers deploying WiMAX services in 118 countries," citing the WiMAX Forum, which last week certified 10 initial mobile WiMAX products in the 2.5GHz worldwide spectrum for interoperability.
"The forum expects to certify 100 mobile WiMAX products by year-end," according to Network World.
A short-term, long-term approach
To be sure, WiMAX has plenty to offer in terms of efficiency and practicality. But wise operators will keep their options open, taking advantage of WiMAX in the short term but keeping an eye on LTE as it comes into its own over the next few years.
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