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FDDIQ. What does FDDI stand for?
Fiber Distributed Data Interface
Q. What is FDDI?
It is a 100 Mbps Local Area Network, defined by ANSI and OSI standards. It was originally designed to operate over fiber optic cabling, but now also includes standard copper media for interconnection. FDDI uses a token ring media access control protocol.
Q. Who developed the FDDI standards?
Most of the standardization of FDDI was done in Acredited Standards Committee X3T9.5. In 1995, X3T9.5 became X3T12. FDDI standards are approved by both ANSI (American National Standards Institute) and ISO (International Standards Organization). ISO approval usually occurs after ANSI approval.
Q. What is the name of the standards?
FDDI STANDARDS (if year is listed as 199x it is still in approval)o Media Access Control (MAC), ANSI X3.139-1987, ISO 9314-2:1989o Physical Layer Protocol (PHY), ANSI X3.148-1988, ISO 9314-1:1989o Physical Layer, Medium Dependent (PMD), ANSI X3.166-1990, ISO 9314-
3:1990o Station Management (SMT), X3.229-1994, ISO 9314-6:199xo Single Mode Fiber PMD (SMF-PMD), ANSI X3.184-1993, ISO 9314-4:199xo Low Cost Fiber PMD (LCF-PMD), X3.237-1995, ISO 9314-9:199xo Twisted Pair PMD (TP-PMD), X3.263.1995, ISO 9314-10:199xo Physical Layer Repeater (PHY-REP), X3.278-199xo Conformance Test PICS Proforma for FDDI (CT-PICS), X3.262-199x, ISO 9314-
13:199xo Abstract Test Suite for MAC (MAC-ATS), X3.245-199x, ISO 9314-26:199xo Abstract Test Suite for PHY (PHY-ATS), X3.248-199x, ISO 9314-21:199xo Abstract Test Suite for PMD (PMD-ATS), X3.255-199x, ISO 9314-20o Abstract Test Suite for SMT (SMT-ATS), X3T9.5/92-102, Rev 1.4
FDDI-II STANDARDSo Hybrid Ring Control (HRC), ANSI X3.186-1993, ISO.IEC 9314-5:199xo Enhanced Media Access Control (MAC-2), X3.239-1994, ISO 9314-8:199xo Enhanced Physical Layer Protocol (PHY-2), X3.231-1994, ISO 9314-7:199xo Enhanced SMT Common Services (SMT-2-CS), X3.257-199xo Enhanced SMT Packet Services (SMT-2-PS), X3.259-199xo Enhanced SMT Isochronous Services (SMT-2-IS), X3.258-199x
This information may be obsoleted by either the FDDI document overview or the FDDI document summary pages.
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Q. Where can I get copies of the standards?
There are no on-line copies of the final standards documents. The standards are available in print, and now also on CD-ROM.
ANSI standards (published FDDI standards)o American National Standards Instituteo 1430 Broadway, New York, NY 10018, USAo Attention: Sales Dept.
Standards and X3T9.5 & X3T12 Documentso Global Engineering Documentso 800-854-7179 (USA)o 303-792-2181o 303-792-2192 (Fax)
FDDI also references 802.1 and 802.2 standardso IEEE Standardso IEEE Service Centero 445 Hoes Lane, Piscataway, NJ 08855, USA
Q. What are other good sources of printed information?
FDDI Handbook By Raj Jain Its ISBN # is 0-201-56376-2. FDDI Technology and Applications: Edited Mirchandani and Khanna Handbook of Computer Communications Standards Vol 2: By Stallings DEC's FDDI primer is now available through Digital Press. It is called "FDDI: An
introduction to Fiber Distributed Data Interfce" Its ISBN # is 1-55558-093-9.
Q. I've heard that FDDI uses a token passing scheme for access arbitration, how does this work?
A token is a special three octet FDDI frame. The station waits until a token comes by, grabs the token, transmits one or more frames and release the token. The amount of frames that can be transmitted is determined by timers in the MAC protocol chips.
Q. How is FDDI's token protocol different from 802.5 Token Ring?
FDDI uses a timed token protocol, while 802.5 uses a priority/reservation token access method. Therefore, there are some differences in frame formats, and significant differences in how a station's traffic is handled (queueing, etc.) Management of the rings is also very different.
Q. I've heard that FDDI is a counter-rotating ring, what does this mean?
FDDI is a dual ring technology. And each ring is running in the opposite direction to improve fault recovery.
A B C
Q. What is a dual ring of trees?
A concentrator provides for connection of either another concentrator or station into one of the dual rings. This allows a tree to extend from each of the master ports of the concentrator.
A B C
D E F
H I
Q. What is dual homing?
When a DAS is connected to two concentrator ports, it is called dual-homing. One port is the active link, where data is transmitted and the other port is a hot standby. The hot standby link is constantly tested and will kick in if the active link fails or is disconnected. The B-port in a DAS is usually the active port and the A-port the hot-standby.
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Q. What is a DAS?
DAS (Dual Attachment Station) is a station with two peer ports (A-Port and B-Port). The A-port connects to the B-Port of another DAS, and the B-port is going to connect to the A-Port the yet another DAS. ie:
A B A B A B
Q. What is a SAS?
SAS (Single Attach Station) is a station with one slave port (S-Port). It is usually connected to a master port (M-Port) of a concentrator.
Q. What is a wrapped ring?
When a link in the dual-ring is broken or not connected, the two adjacent ports connecting to the
broken link will be removed from the ring and the both stations enter the wrap state. This joins the two counter-rotating rings into one ring.
Wrap A Wrap B
A B A B A B
Q. Do I need a concentrator port for each workstation, or can workstations be chained together?
Usually you will need a concentrator port (M-Port) to connect each SAS. A DAS can be connected in the dual rings or to concentrator port(s). FDDI allows two S ports to be connected (a two station ring), or an S port to be connected to an A or B port of a DAS causing a wrapped dual ring. If more than one DAS is used, ring redundancy is lost. (Not all equipment vendors allow S to A, B, or S connections without special configuration.)
Q. If I use a concentrator, what are the advantages/disadvantages?
Advantages: Fault tolerance. When a link breaks, the ring can be segmented. A concentrator can just bypass the problem port and avoid most segmentations. It also gives you better physical planning. Usually people prefer tree physical topology. Generally star configuration of a concentrator system is easier to troubleshoot. Stations can be powered off without serious ring effects. Disadvatages: A concentrator represents a single point of failure. A concentrator configuration may also be more costly.
Q. Can I cascade concentrators? Are there limitations as to how many?
Yes. And you can build a tree as deep as you want. Many users have multiple levels of concentrators. The only limit is FDDI's limit of 500 stations.
Q. What is a bypass and what are the issues in having or not having one?
A bypass relay is a device that is used to skip a station on the dual ring if it is turned off, without causing the ring to wrap. One problem with them is that they attenuate the light in the fiber, so you can't have too many of them. (The maximum number in a row depends on the bypass loss, and how the cable plant is constructed. When the bypass joins two fiber links, the number of connectors between the optical transmitter and reciver will usually increase.)
Q. What are the minimum/maximum distances between stations?
No min, 2 km max for stations with the original multimode interface (PMD). 40-60 km max is possible for single mode fiber (SMF-PMD). An optical attenuator will
be necessary if high power (Category II lasers are used on a short link). No min, 500 m for the new Low Cost Fiber on multimode (LCF-PMD).
Q. What are the types of fiber that are supported?
Multimode (62.5/125 micron graded index multimode fiber) and other fiber like 50/125. 85/125. 100/140 allowed Single mode (8-10 micron)
Q. Can different PMD types be connected together?
All the FDDI optical PMDs use the same wavelength (1300nm), so they can be connected together. For example, PMD can be connected to LCF-PMD as long as you stick to LCF-PMD configuration rules. If you don't understand this optical stuff, don't attempt to mix SMF-PMD with PMD or LCF-PMD devices without advice.
Q. Are there any hazards with the fiber optics?
Too much optical power can permanently damage someone's eyes. This is not generally a problem with PMD and LCF PMD but can be with SMF-PMD especially Category II SMF-PMD. Inspecting the end of any fiber (e.g., with magnification) without knowing what is at the other end is not a smart thing to do.
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Q. I've hear of FDDI over Copper, what type of cable does this scheme use?
Type 1 STP - distance between connections must be less than 100 m Category 5 UTP - distance between connections must be less than 100 m. There is no minimum distance for either of these TP-PMD links.
Q. Is there any advantage to separating the fiber pairs (will the ring work better if only one strand is broken on a DAS connection?)
Not in most applications. Some military applications anticipate physical damage to the cable and therefore separate it to improve survivability.
Q. I have ethernet, can I bridge/route between the 2 protocols?
Yes.
Any problems that arise are generally with the transport protocol. Frame fragmentation is standardized for TCP/IP. It should also be noted that frame fragmentation will not work for DECNET, IPX, LAT, Appletalk, NETBEUI etc. IP is the only protocol that has a standard method of fragmenting. Other protocols destined for Ethernet Lans must stay below the 1500 MTU.
Q. I've heard that there is a frame length difference, what are the issues and problems
here?
FDDI frames have a max size of 4500 bytes and Enet only 1500 bytes. Therefore your bridge or router needs to be smart enough to fragment the packets (eg into smaller IP fragments). Or you need to reduce your frame size to 1500 bytes (of data).
Q. What does an FDDI frame look like?
PA SD FC DA SA INFO ED FS
Symbol Meaning DefinitionPA Preamble (II) (8 or more Idle symbol pairs at initial transmission)
SDStarting Delimiter (JK)
(J followed by K control symbol)
FCFrame Control (nn)
(Tell you if it is a token, MAC frame, LLC frame, SMT frame, frame priority, sync or async)
DADestination Address (nn)
(6 bytes of MAC Address in MSb first format)
SASource Address (nn)
(6 bytes of MAC Address of this station)
INFOInformation field (nn)
(Varibale Length. Usually starts with LLC header, then SNAP field, then the payload eg IP packet)
EDEnding Delimiter (T)
(one T control symbol)
FSFrame Status (EAC)
(Three symbols of status: Error, Address_match, and Copied. Each symbol is either SET or RESET. eg If EAC == RSS, then the frame has no error, some station on the ring matched the DA, and some station on the ring copied the frame into its buffer.
Q. So FDDI operates at 100 Megbits per second, what is the practical maximum bps?
Depends. You can get aggregate usage greater than 95Mbit/s in most installations. As with any LAN, high utilization corresponds to longer queuing times. Many systems run fine at 75Mbps. The maximum for a station is implementation dependent. Some can't do more than 20 Mpbs, while others can sustain more than 90Mbps.
Q. What happens when I bridge between a 100 Mbps FDDI and a 10Mbps ethernet if the FDDI traffic destined for the ethernet gets above 8 Mbps? 10 Mbps?
After the buffer fills Frames start dropping. This is not a problem unique to FDDI however. Consider ethernet to T1, or multiple ethernets to a single ethernet, or a lightly loaded ethernet to a heavly loaded ethernet.
Q. What is the latency across a bridge/router? (Yes I know that different vendors are
different, but what is the window?)
Ask the vendors.
Q. Are there FDDI repeaters?
Yes. The FDDI Physical Layer Repeater Protocol (PHY-REP) standard is in the final stages of approval. Repeaters are sometimes used to convert between single mode and multimode.
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Q. What type of test and trouble shooting equipment is available for FDDI?
FDDI protocol analyzers of varying capability are available from multiple manufacturers. Many FDDI equipment vendors also have management software that implements ring mapping, inspection of other stations operational parameters, general ring state monitoring, etc.
An optical time domain reflectometer (OTDR) and optical power meters are used for testing optical fiber. There are also FDDI link testers that measure power and low level FDDI protocol response.
Q. What about network station management? Does FDDI support SNMP?
Yes. The IETF has an FDDI SNMP MIB. The new MIB is called RFC1512. The previous FDDI SNMP MIB (RFC1285) was based on X3T9.5 SMT 6.2.
Q. Is there RMON for FDDI?
The IETF has not standardized an RMON MIB for FDDI, but many RMON vendors have done their own.
Q. What is a beaconing ring? Does FDDI beacon?
Beacon is a special frame that FDDI MAC sends when something is wrong. When Beaconing persists, SMT will kick in to detect and try to solve the problem. A few FDDI implementations will beacon on initial entry into the ring, but this is a short term condition.
Q. How about interoperability, does one manufacture's equipment work with others?
Just like any networking products, Ethernet, Token, FDDI, ATM, there is a possibility that one vendor does not work with another. But most of the equipment shipping today is tested for interoperability. There are test labs like UNH and ANTC. Ask the vendor what type of testing they did.
Q. Can I interface FDDI to a PC (ISA Bus), PC (EISA Bus), PC (Micro channel Bus),
Macintosh, Sun workstation, DECstation 5000, NEXT computer, Silicon Graphics, Cisco router, WellFleet router, SNA gateway (McData), other?
Yes.
I am not sure if NeXT has any FDDI adaptor software, but there are ~5 different NuBus FDDI cards in the market. But FDDI adaptors are available for all other buses or vendors.
Q. What is the maximum time a station has to wait for media access. What type of applications care?
MaxTime = ~(#of stations * T_neg) (T_neg is the negotiated target token rotation time)
Even at maximum load, this is unlikely. To get this worst case the ring basically must go from zero offered load to maximum offered load (all stations enqueueing ~T-neg of frames) virtually instantenously. If this doesn't happen, the ~T_neg of available bandwidth is split between multiple stations with each fraction of the bandwidth being passed along the ring with each token rotation. This results in a given station having multiple transmit opportunities within MaxTime.
To reduce MaxTime change the T_req of a station to some lower value (eg 8 msec). This is done through the MIB parameter fddiPATHMaxT-Req.
Q. Can I bridge/route TCP/IP, SNA, Novell's IPX, Sun protocols, DecNet, Banyan Vines, Appletalk, X windows, LAT?
Yes, but some equipment may not support bridging of all of these protocols. Check with the bridge vendor for your required protocols.
Q. What are the applications that would use FDDI's bandwidth?
Basically anything will be at least a bit faster. From NFS to images transmission. Even if a single station cannot take advantage of the 100M bit/sec, the aggregate bandwidth will help a lot if your Ethernet is saturated. However, note that though FDDI has higher bandwidth than ethernet, the signals travel at the same speed. The propogation of a signal on the transmission line is the same for ethernet, token ring, and FDDI.
Q. What are the effects of powering off a workstation on a DAS or SAS connection?
Depends. Let's do SAS first, it is easier. If a SAS is connected to a concentrator, then the concentrator will bypass the SAS connection using an internal data path. If a DAS is connected to a concentrator, then the concentrator will also bypass the DAS. If a DAS is connected to the trunk rings without using an optical bypass switch, then the trunk ring will wrap. If multiple stations power off on the trunk rings, then the ring will be segmented. Now if a DAS is using an optical bypass switch, the switch will kick in and prevent the ring from wrapping.
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Q. What are the effects of disconnecting a link on a DAS or SAS connection?
If either conductor (fiber or wire) in a link is broken or disconnected or loses a transmitter or receiver the link is removed from the ring.
SAS connecting to concentrator:o Same as above.
DAS dual-homed to concentrator(s):o If A-port link breaks, usually no effect since A port is generally the backup port.
(And SMT will NOT send out alert msg.)o If B-port fiber breaks, A-port will be joined into the ring after a short disruption.
DAS on trunk rings, with no bypass:o If one link breaks, then the ring will wrap.o If both links break, ring will wrap, station won't be able to communicate.
DAS on trunk rings, with bypass:o When a bypass is switched it breaks its connections before making the new ones.
This causes a temporary wrap when either inserting or removing a station from the ring. The A and B ports of the inserting station will usually be added serially, though it may appear to be simultaneous.
o If any one of the four link entering the bypass breaks, the ring will wrap, living the bypass station in the ring.
o If both links between the bypass and its neighboring stations break, or both links between the bypass and its station break, the ring will wrap, and the station connected to the bypass won't be able to communicate.
Q. What is a common topology?
Connect concentrators and other equipment which is powered on all the time (e.g., bridges, routers, ring monitors, etc.) into the trunk ring. In a large enterprise, workgroup concentrators and users stations are then connected to the backbone concentrators. Some connect bridges and routers and critical servers to backbone concentrators using dual- homing.
Q. What is Graceful Insertion? Should I demand it from my vendors?
Graceful Insertion is a method to insert a station (or a tree) into a ring minimizing disruption. Graceful Insertion is not standard, and therefore different by vendor. Some implementations are both "frame friendly" (don't corrupt frames) and "token friendly" (don't destroy the token). The theory goes that Graceful Insertion can minimize ring non_op and lost frames, therefore saving transmission timeout in upper layer protocols (eg TCP).
The following is the counter argument: Graceful Insertion can use more ring bandwidth (holding the token) than is consumed by a ring recovery. And upper layer protocols are designed to perform frame recovery and retransmission anyway. Also, no vendor can gaurantee 100%
Graceful Insertion anyway.
Q. Is there a Graceful De-insertion?
Not really, but on some concentrators, if you know a station is to be removed from the ring, it could be removed gracefully by sending a command to the concentrator.
Q. Can I run FDDI on coax cable?
There is no standard for FDDI on coax, but check with DEC.
Q. What does SMT stand for? What does it do? Do I need it?
Station ManagemenT (SMT). It is part of the ANSI FDDI Standards that provides link-level management for FDDI. SMT is a low-level protocol that addresses the management of FDDI functions provided by the MAC, PHY, and PMD. It performs functions like ring recovery, frame level management, link control, etc. Every stations on FDDI needs to have SMT.
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Q. What is SMT 6.2 or SMT 7.3?
lot of FDDI equipment was shipped before the SMT standard was completed. Most of this equipment was shipped with SMT software based on SMT Revision 6.2, and earlier shipments were even made with SMT 5.1. SMT 7.3 was the final working document in the standards committee. It is functionally identical to SMT 7.2 and the approved SMT standard. All of these versions of SMT work together on a ring, but they will look different to a management station. (The SMT 6.2 and SMT 7.2 MIBs are very different, as are the frame protocols related to the MIB.)
Q. Can I connect two Single attach stations together and form a two station ring without a concentrator?
Yes. You can do that if both stations support the S-S port connection. Most vendors support the S-S connections.
Q. What are ports? What are the different type of ports?
A port is the connector and supporting logic for one end of an FDDI link. Each port has a transmitter and a receiver. Ports are given names descriptive of their position in FDDI topoligies. SMT defines four types of ports (A, B, M, S). A dual-attachment station has two ports, one A-port and one B-port. A single attach station has only one port (S-port). A concentrator will have many M-port for connecting to other stations' A, B or S-ports.
Q. What are the port connection rules?
When connecting DASs, one should connect the A-port of one station to the B-port of another. S-port on the SAS is to connect to the M-port on a concentrator. A and B-port on DASs can also connect to the M-port of a concentrator. M-ports of concentrators will not connect to each other. In more detail, SMT suggests the following rules:
Other Port
A B S M
This Port
A V,U V V,U V,W
B V V,U V,U V,W
S V,U V,U V V
M V V V I,Uwhere:
V indicates a valid connection I indicates an illegal connection U indicates an undesirable connection with notification to SMT required W indicates, if Active, prevent THRU in CFM and Port B takes precedence
Q. What is the difference between FDDI and FDDI-II?
Both FDDI and FDDI-II run at 100 M bits/sec on the fiber. FDDI can transport both asynchronous and synchronous types of frames.
FDDI-II has a new mode of operation called Hybrid Mode. Hybrid mode uses a 125usec cycle structure to transport isochronus traffic, in addition to sync/async frames.
FDDI and FDDI-II stations can be operated in the same ring only in Basic (FDDI) mode.
There is very little FDDI-II product available on the market.
Function and Frame Format of FDDI
Fiber Distributed-Data Interface (FDDI) :
FDDI (Fiber-Distributed Data Interface) is a standard for data transmission on fiber optic lines in that
can extend in range up to 200 km (124 miles). The FDDI protocol is based on the token ring protocol.
In addition to being large geographically, an FDDI local area network can support thousands of users.
An FDDI network contains two token rings, one for possible backup in case the primary ring fails. The
primary ring offers up to 100 Mbps capacity. If the secondary ring is not needed for backup, it can also
carry data, extending capacity to 200 Mbps. The single ring can extend the maximum distance; a dual
ring can extend 100 km (62 miles).
FDDI is a product of American National Standards Committee X3-T9 and conforms to the open system
interconnect (OSI) model of functional layering. It can be used to interconnect LANs using other
protocols. FDDI-II is a version of FDDI that adds the capability to add circuit-switched service to the
network so that voice signals can also be handled. Work is underway to connect FDDI networks to the
developing Synchronous Optical Network.
Function of FDDI
Background
The Fiber Distributed Data Interface (FDDI) specifies a 100-Mbps token-passing, dual-ring LAN using
fiber-optic cable. FDDI is frequently used as high-speed backbone technology because of its support
for high bandwidth and greater distances than copper. It should be noted that relatively recently, a
related copper specification, called Copper Distributed Data Interface (CDDI) has emerged to provide
100-Mbps service over copper. CDDI is the implementation of FDDI protocols over twisted-pair copper
wire. This chapter focuses mainly on FDDI specifications and operations, but it also provides a high-
level overview of CDDI.
FDDI uses a dual-ring architecture with traffic on each ring flowing in opposite directions (called
counter-rotating). The dual-rings consist of a primary and a secondary ring. During normal operation,
the primary ring is used for data transmission, and the secondary ring remains idle. The primary
purpose of the dual rings, as will be discussed in detail later in this chapter, is to provide superior
reliability and robustness. Figure 1 shows the counter-rotating primary and secondary FDDI rings.
Figure 1: FDDI uses counter-rotating primary and secondary rings.
FDDI Specifications
FDDI specifies the physical and media-access portions of the OSI reference model. FDDI is not actually
a single specification, but it is a collection of four separate specifications each with a specific function.
Combined, these specifications have the capability to provide high-speed connectivity between upper-
layer protocols such as TCP/IP and IPX, and media such as fiber-optic cabling.
FDDI's four specifications are the Media Access Control (MAC), Physical Layer Protocol (PHY), Physical-
Medium Dependent (PMD), and Station Management (SMT). The MAC specification defines how the
medium is accessed, including frame format, token handling, addressing, algorithms for calculating
cyclic redundancy check (CRC) value, and error-recovery mechanisms. The PHY specification defines
data encoding/decoding procedures, clocking requirements, and framing, among other functions. The
PMD specification defines the characteristics of the transmission medium, including fiber-optic links,
power levels, bit-error rates, optical components, and connectors. The SMT specification defines FDDI
station configuration, ring configuration, and ring control features, including station insertion and
removal, initialization, fault isolation and recovery, scheduling, and statistics collection.
FDDI is similar to IEEE 802.3 Ethernet and IEEE 802.5 Token Ring in its relationship with the OSI model.
Its primary purpose is to provide connectivity between upper OSI layers of common protocols and the
media used to connect network devices. Figure 3 illustrates the four FDDI specifications and their
relationship to each other and to the IEEE-defined Logical-Link Control (LLC) sublayer. The LLC
sublayer is a component of Layer 2, the MAC layer, of the OSI reference model.
Figure 2: FDDI specifications map to the OSI hierarchical model.
FDDI Station-Attachment Types
One of the unique characteristics of FDDI is that multiple ways actually exist by which to connect FDDI
devices. FDDI defines three types of devices: single-attachment station (SAS), dual-attachment station
(DAS), and a concentrator.
An SAS attaches to only one ring (the primary) through a concentrator. One of the primary advantages
of connecting devices with SAS attachments is that the devices will not have any effect on the FDDI
ring if they are disconnected or powered off. Concentrators will be discussed in more detail in the
following discussion.
Each FDDI DAS has two ports, designated A and B. These ports connect the DAS to the dual FDDI ring.
Therefore, each port provides a connection for both the primary and the secondary ring. As you will
see in the next section, devices using DAS connections will affect the ring if they are disconnected or
powered off. Figure 3 shows FDDI DAS A and B ports with attachments to the primary and secondary
rings.
Figure 3: FDDI DAS ports attach to the primary and secondary rings.
An FDDI concentrator (also called a dual-attachment concentrator [DAC]) is the building block of an
FDDI network. It attaches directly to both the primary and secondary rings and ensures that the failure
or power-down of any SAS does not bring down the ring. This is particularly useful when PCs, or similar
devices that are frequently powered on and off, connect to the ring. Figure 4 shows the ring
attachments of an FDDI SAS, DAS, and concentrator.
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Figure 4: A concentrator attaches to both the primary and secondary rings.
FDDI Fault Tolerance
FDDI provides a number of fault-tolerant features. In particular, FDDI's dual-ring environment, the
implementation of the optical bypass switch, and dual-homing support make FDDI a resilient media
technology.
Dual Ring
FDDI's primary fault-tolerant feature is the dual ring. If a station on the dual ring fails or is powered
down, or if the cable is damaged, the dual ring is automatically wrapped (doubled back onto itself) into
a single ring. When the ring is wrapped, the dual-ring topology becomes a single-ring topology. Data
continues to be transmitted on the FDDI ring without performance impact during the wrap condition.
Figure 5 and Figure 6 illustrate the effect of a ring wrapping in FDDI.
Figure 5: A ring recovers from a station failure by wrapping.
Figure 6: A ring also wraps to withstand a cable failure.
When a single station fails, as shown in Figure 5, devices on either side of the failed (or powered down)
station wrap, forming a single ring. Network operation continues for the remaining stations on the ring.
When a cable failure occurs, as shown in Figure 6, devices on either side of the cable fault wrap.
Network operation continues for all stations.
It should be noted that FDDI truly provides fault-tolerance against a single failure only. When two or
more failures occur, the FDDI ring segments into two or more independent rings that are unable to
communicate with each other.
Optical Bypass Switch
An optical bypass switch provides continuous dual-ring operation if a device on the dual ring fails. This
is used both to prevent ring segmentation and to eliminate failed stations from the ring. The optical
bypass switch performs this function through the use of optical mirrors that pass light from the ring
directly to the DAS device during normal operation. In the event of a failure of the DAS device, such as
a power-off, the optical bypass switch will pass the light through itself by using internal mirrors and
thereby maintain the ring's integrity. The benefit of this capability is that the ring will not enter a
wrapped condition in the event of a device failure. Figure 7 shows the functionality of an optical bypass
switch in an FDDI network.
Figure 7: The optical bypass switch uses internal mirrors to maintain a network.
Dual Homing
Critical devices, such as routers or mainframe hosts, can use a fault-tolerant technique called dual
homing to provide additional redundancy and to help guarantee operation. In dual-homing situations,
the critical device is attached to two concentrators. Figure 8 shows a dual-homed configuration for
devices such as file servers and routers.
Figure 8: A dual-homed configuration guarantees operation.
One pair of concentrator links is declared the active link; the other pair is declared passive. The
passive link stays in back-up mode until the primary link (or the concentrator to which it is attached) is
determined to have failed. When this occurs, the passive link automatically activates.
FDDI Frame Format
The FDDI frame format is similar to the format of a Token Ring frame. This is one of the areas where
FDDI borrows heavily from earlier LAN technologies, such as Token Ring. FDDI frames can be as large
as 4,500 bytes. Figure 9 shows the frame format of an FDDI data frame and token.
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Figure 9: The FDDI frame is similar to that of a Token Ring frame.
FDDI Frame Fields
The following descriptions summarize the FDDI data frame and token fields illustrated in Figure 9.
Preamble---A unique sequence that prepares each station for an upcoming frame.
Start Delimiter---Indicates the beginning of a frame by employing a signaling pattern that differentiates
it from the rest of the frame.
Frame Control---Indicates the size of the address fields and whether the frame contains asynchronous
or synchronous data, among other control information.
Destination Address---Contains a unicast (singular), multicast (group), or broadcast (every station)
address. As with Ethernet and Token Ring addresses, FDDI destination addresses are 6 bytes long.
Source Address---Identifies the single station that sent the frame. As with Ethernet and Token Ring
addresses, FDDI source addresses are 6 bytes long.
Data---Contains either information destined for an upper-layer protocol or control information.
Frame Check Sequence (FCS)---Filed by the source station with a calculated cyclic redundancy check
value dependent on frame contents (as with Token Ring and Ethernet). The destination address
recalculates the value to determine whether the frame was damaged in transit. If so, the frame is
discarded.
End Delimiter---Contains unique symbols, which cannot be data symbols, that indicate the end of the
frame.
Frame Status---Allows the source station to determine whether an error occurred and whether the
frame was recognized and copied by a receiving station.
FDDI Frame Format
FDDI Frame
Frame Control (FC): 8 bits
has bit format CLFFZZZZ
C indicates synchronous or asynchronous frame
L indicates use of 16 or 48 bit addresses
FF indicates whether it is a LLC, MAC control or reserved frame
in a control frame ZZZZ indicates the type of control
Destination Address (DA): 16 or 48 bits
specifies station for which the frame is intended
Source Address (SA): 16 or 48 bits
specifies station that sent the frame
Here is what the FDDI frame format looks like:
FDDI Frame Format
PA - Preamble 16 symbols
SD - Start Delimiter 2 symbols
FC - Frame Control 2 symbols
DA - Destination Address 4 or 12 symbols
SA - Source Address 4 or 12 symbols
FCS - Frame Check Sequence 8 symbols, covers the FC, DA, SA and Information
ED - End Delimiter 1 or 2 symbols
FS - Frame Status 3 symbols
Token is just the PA, SD, FC and ED
Preamble
The Token owner as a minimum of transmits the preamble 16 symbols of Idle. Physical Layers of the
subsequent repeating stations can change the length of the Idle pattern according to the Physical
Layer requirements. Therefore, each repeating station may see a variable length preamble from the
original preamble. Tokens will be recognized as long as its preamble length is greater than zero. If a
valid token is received and cannot be processed (repeated), due to expiration of ring timing or latency
constraints the station will issue a new token to be put on the ring.
A given MAC implementation is not required to be capable of copying frames received with less than
12 symbols of preamble; Nevertheless, with such frames, it cannot be correctly repeated.
Since the preamble cannot be repeated, the rest of the frame will not be repeated as well.
Starting Delimiter
This field of the frame denodes the start of the frame. It can only have symbols 'J' and 'K'. These
symbols will not be used anywhere else but in the starting delimiter of a token or a frame.
Frame Control
Frame Control field descibes what type of data it is carrying inthe INFO field. Here are the most
common values that are allowed in the FC field:
40: Void Frame.
41,4F: Station Management (SMT) Frame.
C2,C3: MAC Frame.
50,51: LLC Frame.
60: Implementor Frame.
70: Reserved Frame.
Please note that the list here are only the most common values that can be formed by a 48 bit
addressedynchronous data frames.
TOP
Destination Address
Destination Address field contains 12 symbols that identifies the station that is receiving this particular
frame. When FDDI is first setup, each station is given a unique address that identifies themselves from
the others. When a frame passed by the station, the station will compare its address against the DA
field of the frame. If it is a match, station then copies the frame into its buffer area waiting to be
processed. There is not restriction on the number of stations that a frame can reach at a time. If the
first bit of the DA field is set to '1', then the address is called a group or global address. If the first bit
is '0', then the address is called individual address. As the name suggests, a frame with a global
address setting can be sent to multiple stations on the network. If the frame is intended for everyone
on the network, the address bits will be set to all 1's. Therefore, a global address contains all 'F'
symbols. There are also two different ways of administer these addresses. One's called local and the
other's called universal. The second bit of the address field determine whether or not the address is
locally or universally administered. If the second bit is '1' then it is locally administered address. If the
second bit is a '0', then it is universally administered adress.A locally administer address are addresses
that havebeen assigned by the network administrator and a universally administered addresses are
pre-assigned by the manufacturer's OUI.
Source Address<>
A Source Address identifies the station that created the frame. This field is used for remove frames
from the ring. Each time a frame is sent, it travels around the ring, visiting each station,and eventually
(hopefully) comes back to the station that originally sent that frame. If the address of a station
matches the SA field inthe frame, the station will strip the frame off the ring. Each stationis
responsible for removing its own frame from the ring.
Information Field
INFO field is the heart and soul of the frame. Every components of the frame is designed around this
field; Who to send it to, where's this coming from, how it is received and so on.The type of information
in the INFO field can be found by looking in the FC field of the frame. For example:'50'(hex) denodes a
LLC frame. So, the INFO field will have a LLC header followed by other upper layer headers. For
example SNAP, ARP, IP, TCP, SNMP, etc.'41'(hex or '4F'(hex) denode s SMT (Station Management)
frame. Therefore, a SMT header will appear in the INFO field.
Frame Check Sequence
Frame Check Sequence field is used to check or verify the traversingframe for any bit errors. FCS
information is generated by the stationthat sends the frame, using the bits in FC, DA, SA, INFO, and
FCSfields. To verify if there are any bit errors in the frame, FDDI uses 8 symbols (32 bits) CRC (Cyclic
Redundancy Check) to ensure the transmission of a frame on the ring.
End Delimiter
As the name suggests, the end delimiter denodes the end of the frame. The ending delimiter consist of
a 'T' symbol. This 'T' symbols indicates that the frame is complete or ended. Any data sequence that
does not end with this 'T' symbol is not considered to be a frame.
Frame Status
Frame Status (FS) contains 3 indicators that dictates the condition of the frame. Each indicator can
have two values: Set ('S') or Reset ('R'). The indicators could possibly be corrupted. In this case, the
indicators is neither 'S' nor 'R'. All frame are initially set to 'R'.Three types of indicators are as
follows:Error (E):This indicator is set if a station determines an error for that frame. Might be a CRC
failiure or other causes. If a frame has its E indicator set, then, that frame is discarded by the first
station that encounters the frame.Acknowledge(A): Sometime this indicator is called 'address
recognized'. This indicator is set whenever a frame in properly received; meaning the frame has
reached its destination address. Copy (C): This indicator is set whenever a station is able to copy the
received frame into its buffer section. Thus, Copy and Acknowledge indicators are usually set at the
same time. But, sometimes when a station is receiving too many frames and cannot copy all the
incoming frames. If this happens, it would re-transmit the frame with indicator 'A' set indicator 'C' left
on reset.
ALOHA PROTOCOL
Aloha Protocol
Slotted Aloha
Pure Aloha
Aloha and Network Stability
Aloha Simulation & Reservation Aloha Protocol
Slotted ALOHA Simulation Parameters
ALOHA PROTOCOL IN C - LANGUAGE
Aloha, also called the Aloha method, refers to a simple communications scheme in which each source
(transmitter) in a network sends data whenever there is a frame to send. If the frame successfully
reaches the destination (receiver), the next frame is sent. If the frame fails to be received at the
destination, it is sent again. This protocol was originally developed at the University of Hawaii for use
with satellite communication systems in the Pacific.
In a wireless broadcast system or a half-duplex two-way link, Aloha works perfectly. But as networks
become more complex, for example in an Ethernet system involving multiple sources and destinations
in which data travels many paths at once, trouble occurs because data frames collide (conflict). The
heavier the communications volume, the worse the collision problems become. The result is
degradation of system efficiency, because when two frames collide, the data contained in both frames
is lost.
To minimize the number of collisions, thereby optimizing network efficiency and increasing the number
of subscribers that can use a given network, a scheme called slotted Aloha was developed. This
system employs signals called beacons that are sent at precise intervals and tell each source when the
channel is clear to send a frame. Further improvement can be realized by a more sophisticated
protocol called Carrier Sense Multiple Access with Collision Detection (CSMA).
In 1070s, Norman Abram son and his colleagues at the University of Hawaii devised a new and elegant
method to solve the channel allocation problem. Many researchers have extended their work since
then. Although Abranson's work, called the Aloha System, used ground-based radio broadcasting, the
basic idea is applicable to any system in which uncoordinated users are completing for the use of a
single shared channel.
The two several of ALOHA are:
PURE ALOHA
SLOTTED ALOHA
The Aloha Protocol
simple: if you have Packet to send, "just do it"
if Packet suffers collision, will try resending later
Analyzing the Aloha Protocol
Goal: quantitative understanding of performance of Aloha protocol
fixed length Packets
Packet transmission time is unit of time
throughput: S: number Packets successfully (without collision) transmitted per unit time
o in previous example, S = 0.2 Packet/unit time
offered load: G: number Packet transmissions attempted per unit time
o note: S<G, but S depends on G
o Poisson model: probability of k Packet transmission attempts in t time units:
Prob[k trans in t] = ((Gt)**k )(e**(-Gt))/(k!)
capacity of multiple access protocol: maximum value of S over all values of G
Analyzing Aloha (cont)
focus on a given attempted packet transmission
S = rate attempted Packet trans * prob[trans successful]
= G*prob[no other Packet's overlap with attempted trans]
= G*prob[0 other attempted trans in 2 time units]
= Ge*(-2G)
Aloha throughput
Note: maximum thorughput is 18% of physical channel capacity
you buy 1 Mb link, thoughput will never be more than 180Kb!
Comparative analysis - TCP - UDP
TCP
Abbreviation of Transmission Control Protocol, and pronounced as separate letters. TCP is one of the
main protocols in TCP/IP networks. Whereas the IP protocol deals only with packets, TCP enables two
hosts to establish a connection and exchange streams of data. TCP guarantees delivery of data and
also guarantees that packets will be delivered in the same order in which they were sent.
TCP stands for Transmission Control Protocol. It is described in STD-7/RFC-793. TCP is a connection-
oriented protocol that is responsible for reliable communication between two end processes. The unit
of data transferred is called a stream, which is simply a sequence of bytes.
Being connection-oriented means that before actually transmitting data, you must open the
connection between the two end points. The data can be transferred in full duplex (send and receive
on a single connection). When the transfer is done, you have to close the connection to free system
resources. Both ends know when the session is opened (begin) and is closed (end). The data transfer
cannot take place before both ends have agreed upon the connection. The connection can be closed
by either side; the other is notified. Provision is made to close gracefully or just abort the connection.
Being stream oriented means that the data is an anonymous sequence of bytes. There is nothing to
make data boundaries apparent. The receiver has no means of knowing how the data was actually
transmitted. The sender can send many small data chunks and the receiver receive only one big
chunk, or the sender can send a big chunk, the receiver receiving it in a number of smaller chunks. The
only thing that is guaranteed is that all data sent will be received without any error and in the correct
order. Should any error occur, it will automatically be corrected (retransmitted as needed) or the error
will be notified if it can't be corrected.
At the program level, the TCP stream look like a flat file. When you write data to a flat file, and read it
back later, you are absolutely unable to know if the data has been written in only one chunk or in
several chunks. Unless you write something special to identify record boundaries, there is nothing you
can do to learn it afterward. You can, for example, use CR or CR LF to delimit your records just like a
flat text file.
At the programming level, TWSocket is fairly simple to use. To send data, you just need to call the
Send method (or any variation such as SendStr) to give the data to be transmitted. TWSocket will put it
in a buffer until it can be actually transmitted. Eventually the data will be sent in the background (the
Send method returns immediately without waiting for the data to be transmitted) and the OnDataSent
event will be generated once the buffer is emptied.
To receive data, a program must wait until it receives the OnDataAvailable event. This event is
triggered each time a data packet comes from the lower level. The application must call the Receive
method to actually get the data from the low-level buffers. You have to Receive all the data available
or your program will go in an endless loop because TWSocket will trigger the OnDataAvailable again if
you didn't Receive all the data.
As the data is a stream of bytes, your application must be prepared to receive data as sent from the
sender, fragmented in several chunks or merged in bigger chunks. For example, if the sender sent
"Hello " and then "World!", it is possible to get only one OnDataAvailable event and receive "Hello
World!" in one chunk, or to get two events, one for "Hello " and the other for "World!". You can even
receive more smaller chunks like "Hel", "lo wo" and "rld!". What happens depends on traffic load,
router algorithms, random errors and many other parameters you can't control.
On the subject of client/server applications, most applications need to know command boundaries
before being able to process data. As data boundaries are not always preserved, you cannot suppose
your server will receive a single complete command in one OnDataAvailable event. You can receive
only part of a request or maybe two or more request merged in one chunk. To overcome this difficulty,
you must use delimiters.
Most TCP/IP protocols, like SMTP, POP3, FTP and others, use CR/LF pair as command delimiter. Each
client request is sent as is with a CR/LF pair appended. The server receives the data as it arrives,
assembles it in a receive buffer, scans for CR/LF pairs to extract commands from the received stream,
and removes them from the receive buffer.
UDP
Short for User Datagram Protocol, a connectionless protocol that, like TCP, runs on top of IP networks.
Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send
and receive datagrams over an IP network. It's used primarily for broadcasting messages over a
network.
UDP stands for User Datagram Protocol. It is described in STD-6/RFC-768 and provides a
connectionless host-to-host communication path. UDP has minimal overhead:; each packet on the
network is composed of a small header and user data. It is called a UDP datagram.
UDP preserves datagram boundaries between the sender and the receiver. It means that the receiver
socket will receive an OnDataAvailable event for each datagram sent and the Receive method will
return a complete datagram for each call. If the buffer is too small, the datagram will be truncated. If
the buffer is too large, only one datagram is returned, the remaining buffer space is not touched.
UDP is connectionless. It means that a datagram can be sent at any moment without prior advertising,
negotiation or preparation. Just send the datagram and hope the receiver is able to handle it.
UDP is an unreliable protocol. There is absolutely no guarantee that the datagram will be delivered to
the destination host. But to be honest, the failure rate is very low on the Internet and nearly null on a
LAN unless the bandwidth is full.
TOP
Not only the datagram can be undelivered, but it can be delivered in an incorrect order. It means you
can receive a packet before another one, even if the second has been sent before the first you just
received. You can also receive the same packet twice.
Your application must be prepared to handle all those situations: missing datagram, duplicate
datagram or datagram in the incorrect order. You must program error detection and correction. For
example, if you need to transfer some file, you'd better set up a kind of zmodem protocol.
The main advantages for UDP are that datagram boundaries are respected, you can broadcast, and it
is fast.
The main disadvantage is unreliability and therefore complicated to program at the application level.
ADDRESSING
TCP and UDP use the same addressing scheme. An IP address (32 bits number, always written as four
8-bit number expressed as unsigned 3-digit decimal numbers separated by dots such as
193.174.25.26) and a port number (a 16-bit number expressed as a unsigned decimal number).
The IP address is used by the low-level protocol (IP) to route the datagram to the correct host on the
specified network. Then the port number is used to route the datagram to the correct host process (a
program on the host).
For a given protocol (TCP or UDP), a single host process can exist at a time to receive data sent to the
given port. Usually one port is dedicated to one process.
tcp vs. udp for file transfer
i think tcp is an overused protocol; i think udp is an underused protocol. this is an argument i've been
having quite a bit with people lately, so i've decided i'll lay out my reasoning here so i don't have to
type or recite it at people over and over.
since i first wrote this (jan 1998 or so), tcp extensions that fix many of my complaints (selective
acknowledgment, large windows, tcp for transactions) have seen more widespread implementation.
while they are a step in the right direction (well, t/tcp was a terrible security blunder...), tcp will always
be a stream protocol, and thus will never be an optimal transport for some applications.
advantages of tcp
the operating system does all the work. you just sit back and watch the show. no need to have
the same bugs in your code that everyone else did on their first try; it's all been figured out for
you.
since it's in the os, handling incoming packets has fewer context switches from kernel to user
space and back; all the reassembly, acking, flow control, etc is done by the kernel.
tcp guarantees three things: that your data gets there, that it gets there in order, and that it
gets there without duplication. (the truth, the whole truth, and nothing but the truth...)
routers may notice tcp packets and treat them specially. they can buffer and retransmit them,
and in limited cases preack them.
tcp has good relative throughput on a modem or a lan.
disadvantages of tcp
the operating system may be buggy, and you can't escape it. it may be inefficient, and you
have to put up with it. it may be optimized for conditions other than the ones you are facing,
and you may not be able to retune it.
tcp makes it very difficult to try harder; you can set a few socket options, but beyond that you
have to tolerate the built in flow control.
tcp may have lots of features you don't need. it may waste bandwidth, time, or effort on
ensuring things that are irrelevant to the task at hand.
tcp has no block boundaries; you must create your own.
routers on the internet today are out of memory. they can't pay much attention to tcp flying
by, and try to help it. design assumptions of tcp break down in this environment.
tcp has relatively poor throughput on a lossy, high bandwidth, high latency link, such as a
satellite connection or an overfull t1.
tcp cannot be used for broadcast or multicast transmission.
tcp cannot conclude a transmission without all data in motion being explicitly acked.
disadvantages of udp
there are no guarantees with udp. a packet may not be delivered, or delivered twice, or
delivered out of order; you get no indication of this unless the listening program at the other
end decides to say something. tcp is really working in the same environment; you get roughly
the same services from ip and udp. however, tcp makes up for it fairly well, and in a
standardized manner.
udp has no flow control. implementation is the duty of user programs.
routers are quite careless with udp. they never retransmit it if it collides, and it seems to be
the first thing dropped when a router is short on memory. udp suffers from worse packet loss
than tcp.
advantages of udp
it doesn't restrict you to a connection based communication model, so startup latency in
distributed applications is much lower, as is operating system overhead.
all flow control, acking, transaction logging, etc is up to user programs; a broken os
implementation is not going to get in your way. additionally, you only need to implement and
use the features you need.
the recipient of udp packets gets them unmangled, including block boundaries.
broadcast and multicast transmission are available with udp.
TOP
disadvantages of tcp for file transfer
startup latency is significant. it takes at least twice rtt to start getting data back.
tcp allows a window of at most 64k, and the acking mechanism means that packet loss is
misdetected. tcp stalls easily under packet loss. tcp is more throttled by rtt than bandwidth.
tcp transfer servers have to maintain a separate socket (and often separate thread) for each
client.
load balancing is crude and approximate. especially on local networks that allow collisions, two
simultaneous tcp transfers have a tendency to fight with each other, even if the sender is the
same.
advantages of udp for file transfer
latency can be as low as rtt if the protocol is suitably designed.
flow control is up to user space; windows can be infinite, artificial stalls nonexistant, latency
well tolerated, and maximum speeds enforced only by real network bandwidth, yet actual
speeds chosen by agreement of sender and receiver.
receiving an image simultaneously from multiple hosts is much easier with udp, as is sending
one to multiple hosts, especially if they happen to be part of the same broadcast or multicast
group.
a single sending host with multiple transfers proceeding can balance them with excellent precision.
--------------------------------------------------------------------------------
The Internet runs on a hierarchical protocol stack. A simplified version of this is shown in figure 1 . The
layer common to all Internet applications is the IP (Internet Protocol) layer. This layer provides a
connectionless, unreliable packet based delivery service. It can be described as connectionless
because packets are treated independently of all others. The service is unreliable because there is no
guarantee of delivery. Packets may be silently dropped, duplicated or delayed and may arrive out of
order. The service is also called a best effort service, all attempts to deliver a packet will be made, with
unreliability only caused by hardware faults or exhausted resources.
As there is no sense of a connection at the IP level there are no simple methods to provide a quality of
service (QoS). QoS is a request from an application to the network to provide a guarantee on the
quality of a connection. This allows an application to request a fixed amount of bandwidth from the
network, and assume it will be provided, once the QoS request has been accepted. Also a fixed delay,
i.e. no jitter and in order delivery can be assumed. A network that supports QoS will be protected from
congestion problems, as the network will refuse connections that request larger resources than can be
supplied. An example of a network that supports QoS is the current telephone network, where every
call is guaranteed the bandwidth for the call. Most users at some point have heard the overloaded
signal where the network cannot provide the requested resource required to make a call.
The application decides which transport protocol is used. The two protocols shown here, TCP and UDP
are the most commonly used ones. TCP provides a reliable connection and is used by the majority of
current Internet applications. TCP, besides being responsible for error checking and correcting, is also
responsible for controlling the speed at which this data is sent. TCP is capable of detecting congestion
in the network and will back off transmission speed when congestion occurs. These features protect
the network from congestion collapse.
As discussed in the introduction, VoIP is a real-time service. For real-time properties to be guaranteed
to be met, a network with QoS must be used to provide fixed delay and bandwidth. It has already been
said that IP cannot provide this. This then presents a choice. If IP is a requirement, which transport
layer should be used to provide a system that is most likely to meet real-time constraints.
As TCP provides features such as congestion control, it would be the preferred protocol to use.
Unfortunately due to the fact that TCP is a reliable service, delays will be introduced whenever a bit
error or packet loss occurs. This delay is caused by retransmission of the broken packet, along with
any successive packets that may have already been sent. This can be a large source of jitter.
TCP uses a combination of four algorithms to provide congestion control, slow start, congestion
avoidance, fast retransmit and fast recovery. These algorithms all use packet loss as an indication of
congestion, and all alter the number of packets TCP will send before waiting for acknowledgments of
those packets. These alterations affect the bandwidth available and also change delays seen on a link,
providing another source of jitter.
Figure 1: Simplified IP protocol stack
Combined, TCP raises jitter to an unacceptable level rendering TCP unusable for real-time services.
Voice communication has the advantage of not requiring a completely reliable transport level. The loss
of a packet or bit error will often only introduce a click or a minor break into the output.
For these reasons most VoIP applications use UDP for the voice data transmission. UDP is a thin layer
on top of IP that provides a way to distinguish among multiple programs running on a single machine.
UDP also inherits all of the properties of IP that TCP attempts to hide. UDP is therefore also a packet
based, connectionless, best-effort service. It is up to the application to split data into packets, and
provide any necessary error checking that is required.
Because of this, UDP allows the fastest and most simple way of transmitting data to the receiver.
There is no interference in the stream of data that can be possibly avoided. This provides the way for
an application to get as close to meeting real-time constraints as possible.
UDP however provides no congestion control systems. A congested link that is only running TCP will be
approximately fair to all users. When UDP data is introduced into this link, there is no requirement for
the UDP data rates to back off, forcing the remaining TCP connections to back off even further. This
can be though of as UDP data not being a ``good citizen''. The aim of this project is to characterise the
quantity of this drop off in TCP performance.
TCP vs. UDP
TCP UDP
Connection-Oriented
Reliability in delivery of messages
Connectionless
No attempt to fragment messages
Splitting messages into datagrams
Keep track of order (or sequence)
Use checksums for detecting errors
No reassembly and synchronization
In case of error, message is retransmitted
No acknowledgment
Remote procedures are not idempotent
Reliability is a must
Messages exceed UDP packet size
Remote procedures are idempotent
Server and client messages fit completely
within a packet
The server handles multiple clients (UDP is
stateless)
Server Process
socket()
|
bind()
| TCP Server Process UDP
listen socket() |
| Client Process | Client Process
accept() socket() bind() |
| | | socket()
Get a blocked client <-1-> connect() recvfrom() |
| | | bind()
read() <-2-- write() Get a blocked client |
| | | <--- sendto()
process request | process request |
| | | |
write --3-> read() sendto() ---> recvfrom()
Addressing Formats and QoS parameters
<< Back to Networking
When an application process wishes to setup a connection to a remote application process, it must
specify which one to connect to. The method normally used is to define transport addresses to which
processes can listen for connection requests. In the internet, these end points are pairs. In ATM
networks, they are AAL-SAPs, we eill use the neutral term TSAP (Transport Service Access Point). The
analogous end porints in the network layer (ie., network layer addresses) are then called NSAPs. IP
addresses are examples of NSAPs.
The format of a transport protocal data unit (TPDU) is shown below. Each TPDU consists of four general
fields : length, fixed parameters, variable parameters and data.
Length Fixed parameters Variable parameters Data
Length : The length field occupies the first byte and indicates the total number of bytes (excluding
the length field itself) in the TPDU.
Fixed Parameters : The fixed parameters field contains parameters, or control fields that are
commonly present in all transport layer packets. It consists of five parts: Code, source reference,
destination reference, sequence number, and credit allocation.
Code : The code identifies the type of the data unit for example, CR for connection request or DT for
data.
CR : Connection request
CC : Connection confirm
DR : Disconnect request
DC : Disconnect Confirm
DT : Data
ED : Expedited data
AK: Data acknowledge
EA : Expedited data acknowledge
RJ : Reject
ER : Error
Source and destination reference : The source and destination reference fields
contain the addresses of the original sender and the ultimate destination of the packet.
Sequence Number : As a transmission is divided into smaller packets for transport, each
segment is given a number that identifies its place in the sequence. Sequence numbers are used for
acknowledgement, flow control, and reordering of packets at the destination.
Credit Allocation : Credit allocation enables a receiving station to tell the sender how many
more data units may be sent before the sender must wait for an acknowledgement.
Variable parameters : It contains parameters that occur in frequently. These control codes
are used mostly for management.
Data : It may contain regular data or expedited data coming from the upper layers. Expedited data
consist of a high priority message that must be handled out of sequence. An urgent request can
supersede the incoming queue of the receiver and be processed ahead of packets that have been
received before it.
QoS ( Quality of Service ) :
Another way of looking at the transport layer is to regard its primary function as enhancing the QoS
(Quality of Service) provided by the network layer. If the network service is impeccable, the transport
layer has an easy job. If however, the network service is poor, the transport layer has to bridge the gap
between what the transport users want and what the network layer provides.
The transport service may allow the user to specify preferred, acceptable, and minimum values for
various service parameters at the time a connection is setup.
Now we will discuss some of the QoS parameters
The connection establishment delay is the amount of time elapsing between a transport connection
being requested and the confirmation being received by the user of the transport service. It includes
the processing delay in the remote transport entity. As with all parameters measuring a delay, the
shorter the delay, the better the service.
The connection establishment failure probability is the chance of a connection not being established
with in the maximum establishment delay time, for example, due to network congestion, lack of table
space some where, or other internal problems.
The Throughput parameter measures the number of bytes of user data transferred per second,
measured over some time interval. The throughput is measured separately for each direction.
The Transit delay measures the time between a message being sent by the transport user on the
source machine and its being received by the transport user on the destination machine. As with
throughput, each direction is handled separately.
The Residual error ratio measures the number of lost or garbled messages as a fraction of the total
sent. In theory, the residual error rate should be zero, since it is the job of the transport layer to hide
all network layer errors. In practice, it may have some finite value.
The Protection parameter provides a way for the transport user to specify interest in having the
transport layer provide protection against unauthorized third parties (wire tapers) reading or modifying
the transmitted data.
The priority parameter provides a way for a transport user to indicate that some of its connections are
more important than other ones. And in the event of congestion, to make sure that the high-priority
connections get serviced before the low-priority ones.
Finally, the Resilience parameter gives the probability of the transport layer itself spontaneously
terminating a connection due to internal problems or congestion.
The QoS parameters are specifies by the transport user when a connection is requested. Both the
desired and minimum acceptable values can be given . In some cases, upon seeing the QoS
parameters, the transport layer may immediately realize that some of them are unachievable.
IPv6
<< Back to Networking
IPv6 or IPng (IP Next generation):
IPv6 is short for "Internet Protocol Version 6". IPv6 is the "next generation" protocol designed by the
IETF (The Internet Engineering Task Force) to replace the current version Internet Protocol, IP Version
4 ("IPv4"). The IP v 6 specifications are in rfc2460.
Most of today's internet uses IPv4, which is now nearly twenty years old. IPv4 has been remarkably
resilient in spite of its age, but it is beginning to have problems. Most importantly, there is a growing
shortage of IPv4 addresses, which are needed by all new machines added to the Internet.
IPv6 fixes a number of problems in IPv4, such as the limited number of available IPv4 addresses. It also
adds many improvements to IPv4 in areas such as routing and network autoconfiguration. IPv6 is
expected to gradually replace IPv4, with the two coexisting for a number of years during a transition
period.
Contents
1 Introduction
2 Key Issues
3 History of the IPng Effort
4 IPng Overview
5 IPng Header Format
6 IPng Extensions
7 IPng Addressing
8 IPng Routing
9 IPng Quality-of-Service Capabilities
10. IPng Security
11. IPng Transition Mechanisms
12. Why IPng?
1. Introduction
This paper presents an overview of the Next Generation Internet Protocol (IPng). IPng was
recommended by the IPng Area Directors of the Internet Engineering Task Force at the Toronto IETF
meeting on July 25, 1994, and documented in RFC 1752, "The Recommendation for the IP Next
Generation Protocol" [1]. The recommendation was approved by the Internet Engineering Steering
Group on November 17, 1994 and made a Proposed Standard.
The formal name of this protocol is IPv6 (where the "6" refers to it being assigned version number 6).
The current version of the Internet Protocol is version 4 (referred to as IPv4). This overview is intended
to give the reader an overview of the IPng protocol. For more detailed information the reader should
consult the documents listed in the reference section.
IPng is a new version of IP which is designed to be an evolutionary step from IPv4. It is a natural
increment to IPv4. It can be installed as a normal software upgrade in internet devices and is
interoperable with the current IPv4. Its deployment strategy was designed to not have any "flag" days.
IPng is designed to run well on high performance networks (e.g., ATM) and at the same time is still
efficient for low bandwidth networks (e.g., wireless). In addition, it provides a platform for new internet
functionality that will be required in the near future.
This paper describes the work of IETF IPng working group. Several individuals deserve specific
recognition. These include Paul Francis, Bob Gilligan, Dave Crocker, Ran Atkinson, Jim Bound, Ross
Callon, Bill Fink, Ramesh Govindan, Christian Huitema, Erik Nordmark, Tony Li, Dave Katz, Yakov
Rekhter, Bill Simpson, and Sue Thompson.
2.0 Key Issues
There are several key issues that should be considered when reviewing the design of the next
generation internet protocol. Some are very straightforward. For example the new protocol must be
able to support large global internetworks. Others are less obvious. There must be a clear way to
transition the current large installed base of IPv4 systems. It doesn't matter how good a new protocol
is if there isn't a practical way to transition the current operational systems running IPv4 to the new
protocol.
2.1 Growth Growth is the basic issue which caused there to be a need for a next generation IP. If
anything is to be learned from our experience with IPv4 it is that the addressing and routing must be
capable of handling reasonable scenarios of future growth. It is important that we have an
understanding of the past growth and where the future growth will come from.
Currently IPv4 serves what could be called the computer market. The computer market has been the
driver of the growth of the Internet. It comprises the current Internet and countless other smaller
internets which are not connected to the Internet. Its focus is to connect computers together in the
large business, government, and university education markets. This market has been growing at an
exponential rate. One measure of this is that the number of networks in current Internet (40,073 as of
10/4/94) is doubling approximately every 12 months. The computers which are used at the endpoints
of internet communications range from PC's to Supercomputers. Most are attached to Local Area
Networks (LANs) and the vast majority are not mobile.
The next phase of growth will probably not be driven by the computer market. While the computer
market will continue to grow at significant rates due to expansion into other areas such as schools
(elementary through high school) and small businesses, it is doubtful it will continue to grow at an
exponential rate. What is likely to happen is that other kinds of markets will develop. These markets
will fall into several areas. They all have the characteristic that they are extremely large. They also
bring with them a new set of requirements which were not as evident in the early stages of IPv4
deployment. The new markets are also likely to happen in parallel with one another. It may turn out
that we will look back on the last ten years of Internet growth as the time when the Internet was small
and only doubling every year. The challenge for an IPng is to provide a solution which solves todays
problems and is attractive in these emerging markets.
Nomadic personal computing devices seem certain to become ubiquitous as their prices drop and their
capabilities increase. A key capability is that they will be networked. Unlike the majority of todays
networked computers they will support a variety of types of network attachments. When disconnected
they will use RF wireless networks, when used in networked facilities they will use infrared attachment,
and when docked they will use physical wires. This makes them an ideal candidate for internetworking
technology as they will need a common protocol which can work over a variety of physical networks.
These types of devices will become consumer devices and will replace the current generation of
cellular phones, pagers, and personal digital assistants. In addition to the obvious requirement of an
internet protocol which can support large scale routing and addressing, they will require an internet
protocol which imposes a low overhead and supports auto configuration and mobility as a basic
element. The nature of nomadic computing requires an internet protocol to have built in authentication
and confidentiality. It also goes without saying that these devices will need to communicate with the
current generation of computers. The requirement for low overhead comes from the wireless media.
Unlike LAN's which will be very high speed, the wireless media will be several orders of magnitude
slower due to constraints on available frequencies, spectrum allocation, error rates, and power
consumption.
Another market is networked entertainment. The first signs of this emerging market are the proposals
being discussed for 500 channels of television, video on demand, etc. This is clearly a consumer
market. The possibility is that every television set will become an Internet host. As the world of digital
high definition television approaches, the differences between a computer and a television will
diminish. As in the previous market, this market will require an Internet protocol which supports large
scale routing and addressing, and auto configuration. This market also requires a protocol suite which
imposes the minimum overhead to get he job done. Cost will be the major factor in the selection of an
appropriate technology.
Another market which could use the next generation IP is device control. This consists of the control of
everyday devices such as lighting equipment, heating and cooling equipment, motors, and other types
of equipment which are currently controlled via analog switches and in aggregate consume
considerable amounts of electrical power. The size of this market is enormous and requires solutions
which are simple, robust, easy to use, and very low cost. The potential pay-back is that networked
control of devices will result in cost savings which are extremely large.
The challenge the IETF faced in the selection of an IPng is to pick a protocol which meets today's
requirements and also matches the requirements of these emerging markets. These markets will
happen with or without an IETF IPng. If the IETF IPng is a good match for these new markets it is likely
to be used. If not, these markets will develop something else. They will not wait for an IETF solution. If
this should happen it is probable that because of the size and scale of the new markets the IETF
protocol would be supplanted. If the IETF IPng is not appropriate for use in these markets, it is also
probable that they will each develop their own protocols, perhaps proprietary. These new protocols
would not interoperate with each other. The opportunity for the IETF is to select an IPng which has a
reasonable chance to be used in these emerging markets. This would have the very desirable outcome
of creating an immense, interoperable, world- wide information infrastructure created with open
protocols. The alternative is a world of disjoint networks with protocols controlled by individual
vendors.
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2.2 Transition At some point in the next three to seven years the Internet will require a
deployed new version of the Internet protocol. Two factors are driving this: routing and addressing.
Global internet routing based on the on 32-bit addresses of IPv4 is becoming increasingly strained.
IPv4 address do not provide enough flexibility to construct efficient hierarchies which can be
aggregated. The deployment of Classless Inter- Domain Routing [2] is extending the life time of IPv4
routing by a number of years, the effort to manage the routing will continue to increase. Even if the
IPv4 routing can be scaled to support a full IPv4 Internet, the Internet will eventually run out of network
numbers. There is no question that an IPng is needed, but only a question of when.</p>
The challenge for an IPng is for its transition to be complete before IPv4 routing and addressing break.
The transition will be much easier if IPv4 addresses are still globally unique. The two transition
requirements which are the most important are flexibility of deployment and the ability for IPv4 hosts
to communicate with IPng hosts. There will be IPng- only hosts, just as there will be IPv4-only hosts.
The capability must exist for IPng-only hosts to communicate with IPv4-only hosts globally while IPv4
addresses are globally unique.
The deployment strategy for an IPng must be as flexible as possible. The Internet is too large for any
kind of controlled roll out to be successful. The importance of flexibility in an IPng and the need for
interoperability between IPv4 and IPng was well stated in a message to the sipp mailing list by Bill
Fink, who is responsible for a portion of NASA's operational internet. In his message he said:
"Being a network manager and thereby representing the interests of a significant number of users,
from my perspective it's safe to say that the transition and interoperation aspects of any IPng is *the*
key first element, without which any other significant advantages won't be able to be integrated into
the user's network environment. I also don't think it wise to think of the transition as just a painful
phase we'll have to endure en route to a pure IPng environment, since the transition/coexistence
period undoubtedly will last at least a decade and may very well continue for the entire lifetime of
IPng, until it's replaced with IPngng and a new transition. I might wish it was otherwise but I fear they
are facts of life given the immense installed base.
"Given this situation, and the reality that it won't be feasible to coordinate all the infrastructure
changes even at the national and regional levels, it is imperative that the transition capabilities
support the ability to deploy the IPng in the piecemeal fashion... with no requirement to need to
coordinate local changes with other changes elsewhere in the Internet...
"I realize that support for the transition and coexistence capabilities may be a major part of the IPng
effort and may cause some headaches for the designers and developers, but I think it is a duty that
can't be shirked and the necessary price that must be paid to provide as seamless an environment as
possible to the end user and his basic network services such as e-mail, ftp, gopher, X-Window clients,
etc...
"The bottom line for me is that we must have interoperability during the extended transition period for
the base IPv4 functionality..."
Another way to think about the requirement for compatibility with IPv4 is to look at other product
areas. In the product world, backwards compatibility is very important. Vendors who do not provide
backward compatibility for their customers usually find they do not have many customers left. For
example, chip makers put considerable effort into making sure that new versions of their processor
always run all of the software that ran on the previous model. It is unlikely that Intel would develop a
new processor in the X86 family that did not run DOS and the tens of thousands of applications which
run on the current versions of X86's.
Operating system vendors go to great lengths to make sure new versions of their operating systems
are binary compatible with their old version. For example the labels on most PC or MAC software
usually indicate that they require OS version XX or greater. It would be foolish for Microsoft come out
with a new version of Windows which did not run the applications which ran on the previous version.
Microsoft even provides the ability for windows applications to run on their new OS NT. This is an
important feature. They understand that it was very important to make sure that the applications
which run on Windows also run on NT.
The same requirement is also true for IPng. The Internet has a large installed base. Features need to
be designed into an IPng to make the transition as easy as possible. As with processors and operating
systems, it must be backwards compatible with IPv4. Other protocols have tried to replace TCP/IP, for
example XTP and OSI. One element in their failure to reach widespread acceptance was that neither
had any transition strategy other than running in parallel (sometimes called dual stack). New features
alone are not adequate to motivate users to deploy new protocols. IPng must have a great transition
strategy and new features.
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3.0 History of the IPng Effort
The IPng protocol represents the evolution of many different IETF proposals and working groups
focused on developing an IPng. It represents over three years of effort focused on this topic. A brief
history follows:
By the Winter of 1992 the Internet community had developed four separate proposals for IPng. These
were "CNAT", "IP Encaps", "Nimrod", and "Simple CLNP". By December 1992 three more proposals
followed; "The P Internet Protocol" (PIP), "The Simple Internet Protocol" (SIP) and "TP/IX". In the Spring
of 1992 the "Simple CLNP" evolved into "TCP and UDP with Bigger Addresses" (TUBA) and "IP Encaps"
evolved into "IP Address Encapsulation" (IPAE).
By the fall of 1993, IPAE merged with SIP while still maintaining the name SIP. This group later merged
with PIP and the resulting working group called themselves "Simple Internet Protocol Plus" (SIPP). At
about the same time the TP/IX Working Group changed its name to "Common Architecture for the
Internet" (CATNIP).
The IPng area directors made a recommendation for an IPng in July of 1994. This recommendation,
from [1], includes the following elements:
Current address assignment policies are adequate.
There is no current need to reclaim underutilized assigned network numbers.
There is no current need to renumber major portions of the Internet.
CIDR-style assignments of parts of unassigned Class A address space should be considered.
"Simple Internet Protocol Plus (SIPP) Spec. (128 bit ver)" [3] be adopted as the basis for IPng.
The documents listed in Appendix C be the foundation of the IPng effort.
An IPng Working Group be formed, chaired by Steve Deering and Ross Callon.
Robert Hinden be the document editor for the IPng effort.
An IPng Reviewer be appointed and that Dave Clark be the reviewer.
An Address Autoconfiguration Working Group be formed, chaired by Dave Katz and Sue
Thomson.
An IPng Transition Working Group be formed, chaired by Bob Gilligan and TBA.
The Transition and Coexistence Including Testing Working Group be chartered.
Recommendations about the use of non-IPv6 addresses in IPv6 environments and IPv6
addresses in non-IPv6 environments be developed.
The IESG commission a review of all IETF standards documents for IPng implications.
The IESG task current IETF working groups to take IPng into account.
The IESG charter new working groups where needed to revise old standards documents.
Informational RFCs be solicited or developed describing a few specific IPng APIs.
The IPng Area and Area Directorate continue until main documents are offered as Proposed
Standards in late 1994.
Support for the Authentication Header be required.
Support for a specific authentication algorithm be required.
Support for the Privacy Header be required.
Support for a specific privacy algorithm be required.
An "IPng framework for firewalls" be developed.
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4.0 IPng Overview
IPng is a new version of the Internet Protocol, designed as a successor to IP version 4 [4]. IPng is
assigned IP version number 6 and is formally called IPv6 [5].
IPng was designed to take an evolutionary step from IPv4. It was not a design goal to take a radical
step away from IPv4. Functions which work in IPv4 were kept in IPng. Functions which didn't work were
removed. The changes from IPv4 to IPng fall primarily into the following categories:
Expanded Routing and Addressing Capabilities
IPng increases the IP address size from 32 bits to 128 bits, to support more levels of
addressing hierarchy and a much greater number of addressable nodes, and simpler auto-
configuration of addresses.
The scalability of multicast routing is improved by adding a "scope" field to multicast
addresses.
A new type of address called a "anycast address" is defined, to identify sets of nodes where a
packet sent to an anycast address is delivered to one of the nodes. The use of anycast
addresses in the IPng source route allows nodes to control the path which their traffic flows.
Header Format Simplification
Some IPv4 header fields have been dropped or made optional, to reduce the common-case
processing cost of packet handling and to keep the bandwidth cost of the IPng header as low
as possible despite the increased size of the addresses. Even though the IPng addresses are
four time longer than the IPv4 addresses, the IPng header is only twice the size of the IPv4
header.
Improved Support for Options
Changes in the way IP header options are encoded allows for more efficient forwarding, less
stringent limits on the length of options, and greater flexibility for introducing new options in
the future.
Quality-of-Service Capabilities
A new capability is added to enable the labeling of packets belonging to particular traffic
"flows" for which the sender requests special handling, such as non-default quality of service
or "real- time" service.
Authentication and Privacy Capabilities
IPng includes the definition of extensions which provide support for authentication, data
integrity, and confidentiality. This is included as a basic element of IPng and will be included in
all implementations.
The IPng protocol consists of two parts, the basic IPng header and IPng extension headers.
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5.0 IPng Header Format
Version Prior Flow Label
Payload Length Next Header Hop Limit
Source Address
Destination Address
Ver
4-bit Internet Protocol version number = 6.
Prio
4-bit Priority value. See IPng Priority section.
Flow Label
24-bit field. See IPng Quality of Service section.
Payload Length
16-bit unsigned integer. Length of payload, i.e., the rest of the packet following the IPng header, in
octets.
Next Hdr
8-bit selector. Identifies the type of header immediately following the IPng header. Uses the same
values as the IPv4 Protocol field [6].
Hop Limit
8-bit unsigned integer. Decremented by 1 by each node that forwards the packet. The packet is
discarded if Hop Limit is decremented to zero.
Source Address
128 bits. The address of the initial sender of the packet. See [7] for details.
Destination Address
128 bits. The address of the intended recipient of the packet (possibly not the ultimate recipient, if an
optional Routing Header is present).
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6.0 IPng Extensions
IPng includes an improved option mechanism over IPv4. IPng options are placed in separate extension
headers that are located between the IPng header and the transport-layer header in a packet. Most
IPng extension headers are not examined or processed by any router along a packet's delivery path
until it arrives at its final destination. This facilitates a major improvement in router performance for
packets containing options. In IPv4 the presence of any options requires the router to examine all
options.
The other improvement is that unlike IPv4 options, IPng extension headers can be of arbitrary length
and the total amount of options carried in a packet is not limited to 40 bytes. This feature plus the
manner in which they are processed, permits IPng options to be used for functions which were not
practical in IPv4. A good example of this is the IPng Authentication and Security Encapsulation options.
In order to improve the performance when handling subsequent option headers and the transport
protocol which follows, IPng options are always an integer multiple of 8 octets long, in order to retain
this alignment for subsequent headers.
The IPng extension headers which are currently defined are:
Routing Extended Routing (like IPv4 loose source route).
Fragmentation Fragmentation and Reassembly.
Authentication Integrity and Authentication. Security
Encapsulation Confidentiality.
Hop-by-Hop Option Special options which require hop by hop processing.
Destination Options Optional information to be examined by the destination node.
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7.0 IPng Addressing
IPng addresses are 128-bits long and are identifiers for individual interfaces and sets of interfaces.
IPng Addresses of all types are assigned to interfaces, not nodes. Since each interface belongs to a
single node, any of that node's interfaces' unicast addresses may be used as an identifier for the node.
A single interface may be assigned multiple IPv6 addresses of any type.
There are three types of IPng addresses. These are unicast, anycast, and multicast. Unicast addresses
identify a single interface. Anycast addresses identify a set of interfaces such that a packet sent to a
anycast address will be delivered to one member of the set. Multicast addresses identify a group of
interfaces, such that a packet sent to a multicast address is delivered to all of the interfaces in the
group. There are no broadcast addresses in IPv6, their function being superseded by multicast
addresses.
IPng supports addresses which are four times the number of bits as IPv4 addresses (128 vs. 32). This is
4 Billion times 4 Billion times 4 Billion (2^^96) times the size of the IPv4 address space (2^^32). This
works out to be:
340,282,366,920,938,463,463,374,607,431,768,211,456
This is an extremely large address space. In a theoretical sense this is approximately
665,570,793,348,866,943,898,599 addresses per square meter of the surface of the planet Earth
(assuming the earth surface is 511,263,971,197,990 square meters).
In more practical terms the assignment and routing of addresses requires the creation of hierarchies
which reduces the efficiency of the usage of the address space. Christian Huitema performed an
analysis in [8] which evaluated the efficiency of other addressing architecture's (including the French
telephone system, USA telephone systems, current internet using IPv4, and IEEE 802 nodes). He
concluded that 128bit IPng addresses could accommodate between 8x10^^17 to 2x10^^33 nodes
assuming efficiency in the same ranges as the other addressing architecture's. Even his most
pessimistic estimate this would provide 1,564 addresses for each square meter of the surface of the
planet Earth. The optimistic estimate would allow for 3,911,873,538,269,506,102 addresses for each
square meter of the surface of the planet Earth.
The specific type of IPng address is indicated by the leading bits in the address. The variable-length
field comprising these leading bits is called the Format Prefix (FP). The initial allocation of these
prefixes is as follows:
Allocation Prefix(binary) Fraction of Address Space
Reserved 0000 0000 1/256
Unassigned 0000 0001 1/256
Reserved for NSAP Allocation 0000 001 1/128
Reserved for IPX Allocation 0000 010 1/128
Unassigned 0000 011 1/128
Unassigned 0000 1 1/32
Unassigned 0001 1/16
Unassigned 001 1/8
Provider-Based Unicast Address 010 1/8
Unassigned 011 1/8
Reserved for
Neutral-Interconnect-Based
Unicast Addresses 100 1/8
Unassigned 101 1/8
Unassigned 110 1/8
Unassigned 1110 1/16
Unassigned 1111 0 1/32
Unassigned 1111 10 1/64
Unassigned 1111 110 1/128
Unassigned 1111 1110 0 1/512
Link Local Use Addresses 1111 1110 10 1/1024
Site Local Use Addresses 1111 1110 11 1/1024
Multicast Addresses 1111 1111 1/256
This allocation supports the direct allocation of provider addresses, local use addresses, and multicast
addresses. Space is reserved for NSAP addresses, IPX addresses, and neutral-interconnect addresses.
The remainder of the address space is unassigned for future use. This can be used for expansion of
existing use (e.g., additional provider addresses, etc.) or new uses (e.g., separate locators and
identifiers). Note that Anycast addresses are not shown here because they are allocated out of the
unicast address space.
Approximately fifteen percent of the address space is initially allocated. The remaining 85% is
reserved for future use.
7.1 Unicast Addresses There are several forms of unicast address assignment in IPv6. These
are the global provider based unicast address, the neutral-interconnect unicast address, the NSAP
address, the IPX hierarchical address, the site-local-use address, the link-local-use address, and the
IPv4-capable host address. Additional address types can be defined in the future.
7.2 Provider Based Unicast Addresses Provider based unicast addresses are used for
global communication. They are similar in function to IPv4 addresses under CIDR. The assignment plan
for unicast addresses is described in [9] and [10]. Their format is:
| 3 | n bits | m bits | o bits | p bits | o-p bits |
+---+-----------+-----------+-------------+---------+----------+
|010|REGISTRY ID|PROVIDER ID|SUBSCRIBER ID|SUBNET ID| INTF. ID |
+---+-----------+-----------+-------------+---------+----------+
The first 3 bits identify the address as a provider- oriented unicast address. The next field (REGISTRY
ID) identifies the internet address registry which assigns provider identifiers (PROVIDER ID) to internet
service providers, which then assign portions of the address space to subscribers. This usage is similar
to assignment of IP addresses under CIDR. The SUBSCRIBER ID distinguishes among multiple
subscribers attached to the internet service provider identified by the PROVIDER ID. The SUBNET ID
identifies a specific physical link. There can be multiple subnets on the same physical link. A specific
subnet can not span multiple physical links. The INTERFACE ID identifies a single interface among the
group of interfaces identified by the subnet prefix.
7.3 Local-Use Addresses A local-use address is a unicast address that has only local
routability scope (within the subnet or within a subscriber network), and may have local or global
uniqueness scope. They are intended for use inside of a site for "plug and play" local communication
and for bootstrapping up to the use of global addresses [11].
There are two types of local-use unicast addresses defined. These are Link-Local and Site-Local. The
Link-Local-Use is for use on a single link and the Site-Local-Use is for use in a single site. Link-Local-
Use addresses have the following format:
| 10 |
| bits | n bits | 118-n bits |
+----------+-------------------------+----------------------------+
|1111111010| 0 | INTERFACE ID |
+----------+-------------------------+----------------------------+
Link-Local-Use addresses are designed to be used for addressing on a single link for purposes such as
auto-address configuration.
Site-Local-Use addresses have the following format:
| 10 |
| bits | n bits | m bits | 118-n-m bits |
+----------+---------+---------------+----------------------------+
|1111111011| 0 | SUBNET ID | INTERFACE ID |
+----------+---------+---------------+----------------------------+
For both types of local use addresses the INTERFACE ID is an identifier which much be unique in the
domain in which it is being used. In most cases these will use a node's IEEE-802 48bit address. The
SUBNET ID identifies a specific subnet in a site. The combination of the SUBNET ID and the INTERFACE
ID to form a local use address allows a large private internet to be constructed without any other
address allocation.
Local-use addresses allow organizations that are not (yet) connected to the global Internet to operate
without the need to request an address prefix from the global Internet address space. Local-use
addresses can be used instead. If the organization later connects to the global Internet, it can use its
SUBNET ID and INTERFACE ID in combination with a global prefix (e.g., REGISTRY ID + PROVIDER ID +
SUBSCRIBER ID) to create a global address. This is a significant improvement over IPv4 which requires
sites which use private (non-global) IPv4 address to manually renumber when they connect to the
Internet. IPng does the renumbering automatically.
7.4 IPv6 Addresses with Embedded IPV4 Addresses The IPv6 transition mechanisms
include a technique for hosts and routers to dynamically tunnel IPv6 packets over IPv4 routing
infrastructure. IPv6 nodes that utilize this technique are assigned special IPv6 unicast addresses that
carry an IPv4 address in the low-order 32-bits. This type of address is termed an "IPv4-compatible IPv6
address" and has the format:
| 80 bits | 16 | 32 bits |
+--------------------------------------+--------------------------+
|0000..............................0000|0000| IPV4 ADDRESS |
+--------------------------------------+----+---------------------+
A second type of IPv6 address which holds an embedded IPv4 address is also defined. This address is
used to represent the addresses of IPv4- only nodes (those that *do not* support IPv6) as IPv6
addresses. This type of address is termed an "IPv4-mapped IPv6 address" and has the format:
| 80 bits | 16 | 32 bits |
+--------------------------------------+--------------------------+
|0000..............................0000|FFFF| IPV4 ADDRESS |
+--------------------------------------+----+---------------------+
7.5 Anycast Addresses
An IPv6 anycast address is an address that is assigned to more than one interfaces (typically belonging
to different nodes), with the property that a packet sent to an anycast address is routed to the
"nearest" interface having that address, according to the routing protocols' measure of distance.
Anycast addresses, when used as part of an route sequence, permits a node to select which of several
internet service providers it wants to carry its traffic. This capability is sometimes called "source
selected policies". This would be implemented by configuring anycast addresses to identify the set of
routers belonging to internet service providers (e.g., one anycast address per internet service
provider). These anycast addresses can be used as intermediate addresses in an IPv6 routing header,
to cause a packet to be delivered via a particular provider or sequence of providers. Other possible
uses of anycast addresses are to identify the set of routers attached to a particular subnet, or the set
of routers providing entry into a particular routing domain.
Anycast addresses are allocated from the unicast address space, using any of the defined unicast
address formats. Thus, anycast addresses are syntactically indistinguishable from unicast addresses.
When a unicast address is assigned to more than one interface, thus turning it into an anycast
address, the nodes to which the address is assigned must be explicitly configured to know that it is an
anycast address.
7.6 Multicast Addresses A IPng multicast address is an identifier for a group of interfaces. A
interface may belong to any number of multicast groups. Multicast addresses have the following
format:
| 8 | 4 | 4 | 112 bits |
+------ -+----+----+---------------------------------------------+
|11111111|FLGS|SCOP| GROUP ID |
+--------+----+----+---------------------------------------------+
11111111 at the start of the address identifies the address as being a multicast address.
+-+-+-+-+ FLGS is a set of 4 flags: |0|0|0|T| +-+-+-+-+
The high-order 3 flags are reserved, and must be initialized to 0.
T=0 indicates a permanently assigned ("well-known") multicast address, assigned by the global
internet numbering authority.
T=1 indicates a non-permanently assigned ("transient") multicast address.
SCOP is a 4-bit multicast scope value used to limit the scope of the multicast group. The values are:
0 Reserved 8 Organization-local scope 1 Node-local scope 9 (unassigned) 2 Link-local scope A
(unassigned) 3 (unassigned) B (unassigned) 4 (unassigned) C (unassigned) 5 Site-local scope D
(unassigned) 6 (unassigned) E Global scope 7 (unassigned) F Reserved
GROUP ID identifies the multicast group, either permanent or transient, within the given scope.
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8.0 IPng Routing
Routing in IPng is almost identical to IPv4 routing under CIDR except that the addresses are 128- bit
IPng addresses instead of 32-bit IPv4 addresses. With very straightforward extensions, all of IPv4's
routing algorithms (OSPF, RIP, IDRP, ISIS, etc.) can used to route IPng.
IPng also includes simple routing extensions which support powerful new routing functionality. These
capabilities include:
Provider Selection (based on policy, performance, cost, etc.)
Host Mobility (route to current location)
Auto-Readdressing (route to new address)
The new routing functionality is obtained by creating sequences of IPng addresses using the IPng
Routing option. The routing option is used by a IPng source to list one or more intermediate nodes (or
topological group) to be "visited" on the way to a packet's destination. This function is very similar in
function to IPv4's Loose Source and Record Route option.
In order to make address sequences a general function, IPng hosts are required in most cases to
reverse routes in a packet it receives (if the packet was successfully authenticated using the IPng
Authentication Header) containing address sequences in order to return the packet to its originator.
This approach is taken to make IPng host implementations from the start support the handling and
reversal of source routes. This is the key for allowing them to work with hosts which implement the
new features such as provider selection or extended addresses.
Three examples show how the address sequences can be used. In these examples, address sequences
are shown by a list of individual addresses separated by commas. For example:
SRC, I1, I2, I3, DST
Where the first address is the source address, the last address is the destination address, and the
middle addresses are intermediate addresses.
For these examples assume that two hosts, H1 and H2 wish to communicate. Assume that H1 and H2's
sites are both connected to providers P1 and P2. A third wireless provider, PR, is connected to both
providers P1 and P2.
----- P1 ------
/ | \
/ | \
H1 PR H2
\ | /
\ | /
----- P2 ------
The simplest case (no use of address sequences) is when H1 wants to send a packet to H2 containing
the addresses:H1, H2
When H2 replied it would reverse the addresses and construct a packet containing the addresses: H2,
H1
In this example either provider could be used, and H1 and H2 would not be able to select which
provider traffic would be sent to and received from.
If H1 decides that it wants to enforce a policy that all communication to/from H2 can only use provider
P1, it would construct a packet containing the address sequence: H1, P1, H2
This ensures that when H2 replies to H1, it will reverse the route and the reply it would also travel over
P1. The addresses in H2's reply would look like: H2, P1, H1
If H1 became mobile and moved to provider PR, it could maintain (not breaking any transport
connections) communication with H2, by sending packets that contain the address sequence: H1, PR,
P1, H2
This would ensure that when H2 replied it would enforce H1's policy of exclusive use of provider P1 and
send the packet to H1 new location on provider PR. The reversed address sequence would be: H2, P1,
PR, H1
The address sequence facility of IPng can be used for provider selection, mobility, and readdressing. It
is a simple but powerful capability.
TOP
9.0 IPng Quality-of-Service Capabilities
The Flow Label and the Priority fields in the IPng header may be used by a host to identify those
packets for which it requests special handling by IPng routers, such as non-default quality of service or
"real-time" service. This capability is important in order to support applications which require some
degree of consistent throughput, delay, and/or jitter. These type of applications are commonly
described as "multi- media" or "real-time" applications.
9.1 Flow Labels The 24-bit Flow Label field in the IPv6 header may be used by a source to label
those packets for which it requests special handling by the IPv6 routers, such as non-default quality of
service or "real-time" service.
This aspect of IPv6 is, at the time of writing, still experimental and subject to change as the
requirements for flow support in the Internet become clearer. Hosts or routers that do not support the
functions of the Flow Label field are required to set the field to zero when originating a packet, pass
the field on unchanged when forwarding a packet, and ignore the field when receiving a packet.
A flow is a sequence of packets sent from a particular source to a particular (unicast or multicast)
destination for which the source desires special handling by the intervening routers. The nature of that
special handling might be conveyed to the routers by a control protocol, such as a resource
reservation protocol, or by information within the flow's packets themselves, e.g., in a hop-by-hop
option.
There may be multiple active flows from a source to a destination, as well as traffic that is not
associated with any flow. A flow is uniquely identified by the combination of a source address and a
non- zero flow label. Packets that do not belong to a flow carry a flow label of zero.
A flow label is assigned to a flow by the flow's source node. New flow labels must be chosen
(pseudo-)randomly and uniformly from the range 1 to FFFFFF hex. The purpose of the random
allocation is to make any set of bits within the Flow Label field suitable for use as a hash key by
routers, for looking up the state associated with the flow.
All packets belonging to the same flow must be sent with the same source address, same destination
address, and same non-zero flow label. If any of those packets includes a Hop-by-Hop Options header,
then they all must be originated with the same Hop-by-Hop Options header contents (excluding the
Next Header field of the Hop-by-Hop Options header). If any of those packets includes a Routing
header, then they all must be originated with the same contents in all extension headers up to and
including the Routing header (excluding the Next Header field in the Routing header). The routers or
destinations are permitted, but not required, to verify that these conditions are satisfied. If a violation
is detected, it should be reported to the source by an ICMP Parameter Problem message, Code 0,
pointing to the high-order octet of the Flow Label field (i.e., offset 1 within the IPv6 packet) [12].
Routers are free to "opportunistically" set up flow- handling state for any flow, even when no explicit
flow establishment information has been provided to them via a control protocol, a hop-by-hop option,
or other means. For example, upon receiving a packet from a particular source with an unknown, non-
zero flow label, a router may process its IPv6 header and any necessary extension headers as if the
flow label were zero. That processing would include determining the next-hop interface, and possibly
other actions, such as updating a hop-by-hop option, advancing the pointer and addresses in a Routing
header, or deciding on how to queue the packet based on its Priority field. The router may then choose
to "remember" the results of those processing steps and cache that information, using the source
address plus the flow label as the cache key. Subsequent packets with the same source address and
flow label may then be handled by referring to the cached information rather than examining all those
fields that, according to the requirements of the previous paragraph, can be assumed unchanged from
the first packet seen in the flow.
TOP
9.2 Priority The 4-bit Priority field in the IPv6 header enables a source to identify the desired
delivery priority of its packets, relative to other packets from the same source. The Priority values are
divided into two ranges: Values 0 through 7 are used to specify the priority of traffic for which the
source is providing congestion control, i.e., traffic that "backs off" in response to congestion, such as
TCP traffic. Values 8 through 15 are used to specify the priority of traffic that does not back off in
response to congestion, e.g., "real-time" packets being sent at a constant rate.
For congestion-controlled traffic, the following Priority values are recommended for particular
application categories:
0 Uncharacterized traffic
1 "Filler" traffic (e.g., netnews)
2 Unattended data transfer (e.g., email)
3 (Reserved)
4 Attended bulk transfer (e.g., FTP, HTTP, NFS)
5 (Reserved)
6 Interactive traffic (e.g., telnet, X)
7 Internet control traffic (e.g., routing protocols, SNMP)
For non-congestion-controlled traffic, the lowest Priority value (8) should be used for those packets
that the sender is most willing to have discarded under conditions of congestion (e.g., high-fidelity
video traffic), and the highest value (15) should be used for those packets that the sender is least
willing to have discarded (e.g., low-fidelity audio traffic). There is no relative ordering implied between
the congestion-controlled priorities and the non-congestion-controlled priorities.
TOP
10. IPng Security
The current Internet has a number of security problems and lacks effective privacy and authentication
mechanisms below the application layer. IPng remedies these shortcomings by having two integrated
options that provide security services [13]. These two options may be used singly or together to
provide differing levels of security to different users. This is very important because different user
communities have different security needs.
The first mechanism, called the "IPng Authentication Header", is an extension header which provides
authentication and integrity (without confidentiality) to IPng datagrams [14]. While the extension is
algorithm- independent and will support many different authentication techniques, the use of keyed
MD5 is proposed to help ensure interoperability within the worldwide Internet. This can be used to
eliminate a significant class of network attacks, including host masquerading attacks. The use of the
IPng Authentication Header is particularly important when source routing is used with IPng because of
the known risks in IP source routing. Its placement at the internet layer can help provide host origin
authentication to those upper layer protocols and services that currently lack meaningful protections.
This mechanism should be exportable by vendors in the United States and other countries with similar
export restrictions because it only provides authentication and integrity, and specifically does not
provide confidentiality. The exportability of the IPng Authentication Header encourages its widespread
deployment and use.
The second security extension header provided with IPng is the "IPng Encapsulating Security Header"
[15]. This mechanism provides integrity and confidentiality to IPng datagrams. It is simpler than some
similar security protocols (e.g., SP3D, ISO NLSP) but remains flexible and algorithm-independent. To
achieve interoperability within the global Internet, the use of DES CBC is being used as the standard
algorithm for use with the IPng Encapsulating Security Header.
TOP
11. IPng Transition Mechanisms
The key transition objective is to allow IPv6 and IPv4 hosts to interoperate. A second objective is to
allow IPv6 hosts and routers to be deployed in the Internet in a highly diffuse and incremental fashion,
with few interdependencies. A third objective is that the transition should be as easy as possible for
end- users, system administrators, and network operators to understand and carry out.
The IPng transition mechanisms are a set of protocol mechanisms implemented in hosts and routers,
along with some operational guidelines for addressing and deployment, designed to make transition
the Internet to IPv6 work with as little disruption as possible [16].
The IPng transition mechanisms provides a number of features, including:
Incremental upgrade and deployment. Individual IPv4 hosts and routers may be upgraded to
IPv6 one at a time without requiring any other hosts or routers to be upgraded at the same
time. New IPv6 hosts and routers can be installed one by one.
Minimal upgrade dependencies. The only prerequisite to upgrading hosts to IPv6 is that the
DNS server must first be upgraded to handle IPv6 address records. There are no pre-requisites
to upgrading routers.
Easy Addressing. When existing installed IPv4 hosts or routers are upgraded to IPv6, they may
continue to use their existing address. They do not need to be assigned new addresses.
Administrators do not need to draft new addressing plans.
Low start-up costs. Little or no preparation work is needed in order to upgrade existing IPv4
systems to IPv6, or to deploy new IPv6 systems. The mechanisms employed by the IPng
transition mechanisms include:
An IPv6 addressing structure that embeds IPv4 addresses within IPv6 addresses, and encodes
other information used by the transition mechanisms.
A model of deployment where all hosts and routers upgraded to IPv6 in the early transition
phase are "dual" capable (i.e. implement complete IPv4 and IPv6 protocol stacks).
The technique of encapsulating IPv6 packets within IPv4 headers to carry them over segments
of the end-to-end path where the routers have not yet been upgraded to IPv6.
The header translation technique to allow the eventual introduction of routing topologies that
route only IPv6 traffic, and the deployment of hosts that support only IPv6. Use of this
technique is optional, and would be used in the later phase of transition if it is used at all.
The IPng transition mechanisms ensures that IPv6 hosts can interoperate with IPv4 hosts anywhere in
the Internet up until the time when IPv4 addresses run out, and allows IPv6 and IPv4 hosts within a
limited scope to interoperate indefinitely after that. This feature protects the huge investment users
have made in IPv4 and ensures that IPv6 does not render IPv4 obsolete. Hosts that need only a limited
connectivity range (e.g., printers) need never be upgraded to IPv6.
The incremental upgrade features of the IPng transition mechanisms allow the host and router vendors
to integrate IPv6 into their product lines at their own pace, and allows the end users and network
operators to deploy IPng on their own schedules.
TOP
12. Why IPng?
There are a number of reasons why IPng is appropriate for the next generation of the Internet Protocol.
It solves the Internet scaling problem, provides a flexible transition mechanism for the current Internet,
and was designed to meet the needs of new markets such as nomadic personal computing devices,
networked entertainment, and device control. It does this in a evolutionary way which reduces the risk
of architectural problems.
Ease of transition is a key point in the design of IPng. It is not something was added in at the end. IPng
is designed to interoperate with IPv4. Specific mechanisms (embedded IPv4 addresses, pseudo-
checksum rules etc.) were built into IPng to support transition and compatibility with IPv4. It was
designed to permit a gradual and piecemeal deployment with a minimum of dependencies.
IPng supports large hierarchical addresses which will allow the Internet to continue to grow and
provide new routing capabilities not built into IPv4. It has anycast addresses which can be used for
policy route selection and has scoped multicast addresses which provide improved scalability over IPv4
multicast. It also has local use address mechanisms which provide the ability for "plug and play"
installation.
The address structure of IPng was also designed to support carrying the addresses of other internet
protocol suites. Space was allocated in the addressing plan for IPX and NSAP addresses. This was done
to facilitate migration of these internet protocols to IPng.
IPng provides a platform for new Internet functionality. This includes support for real-time flows,
provider selection, host mobility, end-to- end security, auto-configuration, and auto-reconfiguration.
In summary, IPng is a new version of IP. It can be installed as a normal software upgrade in internet
devices. It is interoperable with the current IPv4. Its deployment strategy was designed to not have
any "flag" days. IPng is designed to run well on high performance networks (e.g., ATM) and at the
same time is still efficient for low bandwidth networks (e.g., wireless). In addition, it provides a
platform for new internet functionality that will be required in the near future.
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Telnet
The Telnet Protocol
The Telnet protocol is often thought of as simply providing a facility for remote logins to computer via
the Internet. This was its original purpose although it can be used for many other purposes. It is best
understood in the context of a user with a simple terminal using the local telnet program (known as
the client program) to run a login session on a remote computer where his communications needs are
handled by a telnet server program. It should be emphasised that the telnet server can pass on the
data it has received from the client to many other types of process including a remote login server. It
is described in RFC854 and was first published in 1983.
Commands
The telnet protocol also specifies various commands that control the method and various details of the
interaction between the client and server. These commands are incorporated within the data stream.
The commands are distinguished by the use of various characters with the most significant bit set.
Commands are always introduced by a character with the decimal code 255 known as an Interpret as
command (IAC) character. The complete set of special characters isNam
eDecimal
CodeMeaning
SE 240 End of subnegotiation parameters.
NOP 241 No operation
DM 242Data mark. Indicates the position of a Synch event within the data stream. This should always be accompanied by a TCP urgent notification.
BRK 243 Break. Indicates that the "break" or "attention" key was hit.
IP 244 Suspend, interrupt or abort the process to which the NVT is connected.
AO 245Abort output. Allows the current process to run to completion but do not send its output to the user.
AYT 246Are you there. Send back to the NVT some visible evidence that the AYT was received.
EC 247 Erase character. The receiver should delete the last preceding undeleted character
from the data stream.
EL 248Erase line. Delete characters from the data stream back to but not including the previous CRLF.
GA 249Go ahead. Used, under certain circumstances, to tell the other end that it can transmit.
SB 250 Subnegotiation of the indicated option follows.
WILL 251Indicates the desire to begin performing, or confirmation that you are now performing, the indicated option.
WONT 252 Indicates the refusal to perform, or continue performing, the indicated option.
DO 253Indicates the request that the other party perform, or confirmation that you are expecting the other party to perform, the indicated option.
DONT 254Indicates the demand that the other party stop performing, or confirmation that you are no longer expecting the other party to perform, the indicated option.
IAC 255 Interpret as command
There are a variety of options that can be negotiated between a telnet client and server using
commands at any stage during the connection. They are described in detail in separate RFCs. The
following are the most important.
Decimal code Name RFC
1 echo 857
3 suppress go ahead 858
5 status 859
6 timing mark 860
24 terminal type 1091
31 window size 1073
32 terminal speed 1079
33 remote flow control 1372
34 linemode 1184
36 environment variables 1408
Options are agreed by a process of negotiation which results in the client and server having a common
view of various extra capabilities that affect the interchange and the operation of applications.Either
end of a telnet dialogue can enable or disable an option either locally or remotely. The initiator sends a
3 byte command of the form IAC,<type of operation>,<option>The response is of the same
form.Operation is one of
Description
Decimal Code
Action
WILL 251 Sender wants to do something.
DO 252 Sender wants the other end to do something.
WONT 253 Sender doesn't want to do something.
DONT 254 Sender wants the other not to do something.
Associated with each of the these there are various possible responses
Sender Sent
Receiver Responds
Implication
WILL DOThe sender would like to use a certain facility if the receiver can handle it. Option is now in effect
WILL DONT Receiver says it cannot support the option. Option is not in effect.
DO WILLThe sender says it can handle traffic from the sender if the sender wishes to use a certain option. Option is now in effect.
DO WONT Receiver says it cannot support the option. Option is not in effect.
WONT DONT Option disabled. DONT is only valid response.
DONT WONT Option disabled. WONT is only valid response.
For example if the sender wants the other end to suppress go-ahead it would send the byte
sequence255(IAC),251(WILL),3The final byte of the three byte sequence identifies the required
action.For some of the negotiable options values need to be communicated once support of the option
has been agreed. This is done using sub-option negotiation.
Values are communicated via an exchange of value query commands and responses in the following
form. IAC,SB,<option code number>,1,IAC,SEandIAC,SB,<option code>,0,<value>,IAC,SE
For example if the client wishes to identify the terminal type to the server the following exchange
might take place
Client 255(IAC),251(WILL),24
Server 255(IAC),253(DO),24
Server 255(IAC),250(SB),24,1,255(IAC),240(SE)
Client 255(IAC),250(SB),24,0,'V','T','2','2','0',255(IAC),240(SE)
The first exchange establishes that terminal type (option number 24) will be handled, the server then
enquires of the client what value it wishes to associate with the terminal type. The sequence SB,24,1
implies sub-option negotiation for option type 24, value required (1). The IAC,SE sequence indicates
the end of this request. The repsonse IAC,SB,24,0,'V'... implies sub-option negotiation for option type
24, value supplied (0), the IAC,SE sequence indicates the end of the response (and the supplied value).
The encoding of the value is specific to the option but a sequence of characters, as shown above, is
common.SMTP ProtocolThe SMTP ProtocolSimple Mail Transfer Protocol
SMTP (Simple Mail Transfer Protocol) is a TCP/IP protocol used in sending and receiving e-mail. However, since it's limited in its ability to queue messages at the receiving end, it's usually used with one of two other protocols, POP3 or Internet Message Access Protocol, that let the user save messages in a server mailbox and download them periodically from the server. In other words, users typically use a program that uses SMTP for sending e-mail and either POP3 or IMAP for receiving messages that have been received for them at their local server. Most mail programs such as Eudora let you specify both an SMTP server and a POP server. On UNIX-based systems, sendmail is the most widely-used SMTP server for e-mail. A commercial package, Sendmail, includes a POP3 server and also comes in a version for Windows NT.
SMTP usually is implemented to operate over Transmission Control Protocol port 25.
Simple Mail Transfer Protocol (SMTP), documented in RFC 821, is Internet's standard host-to-host mail transport protocol and traditionally operates over TCP, port 25. In other words, a UNIX user can type telnet hostname 25 and connect with an SMTP server, if one is present.
SMTP uses a style of asymmetric request-response protocol popular in the early 1980s, and still seen occasionally, most often in mail protocols. The protocol is designed to be equally useful to either a computer or a human, though not too forgiving of the human. From the server's viewpoint, a clear set of commands is provided and well-documented in the RFC. For the human, all the commands are clearly terminated by newlines and a HELP command lists all of them. From the sender's viewpoint, the command replies always take the form of text lines, each starting with a three-digit code identifying the result of the operation, a continuation character to indicate another lines following, and then arbitrary text information designed to be informative to a human.
If mail delivery fails, sendmail (the most important SMTP implementation) will queue mail messages and retry delivery later. However, a backoff algorithm is used, and no mechanism exists to poll all Internet hosts for mail, nor does SMTP provide any mailbox facility, or any special features beyond mail transport. For these reasons, SMTP isn't a good choice for hosts situated behind highly unpredictable lines (like modems). A better-connected host can be designated as a DNS mail exchanger, then arrange for a relay scheme. Currently, there two main configurations that can be used. One is to configure POP mailboxes and a POP server on the exchange host, and let all users use POP-enabled mail clients. The other possibility is to arrange for a periodic SMTP mail transfer from the exchange
host to another, local SMTP exchange host which has been queuing all the outbound mail. Of course, since this solution does not allow full-time Internet access, it is not too preferred.
RFC 1869 defined the capability for SMTP service extensions, creating Extended SMTP, or ESMTP. ESMTP is by definition extensible, allowing new service extensions to be defined and registered with IANA. Probably the most important extension currently available is Delivery Status Notification (DSN), defined in RFC 1891.
TCP
Next >
Example of using BSD sockets: "Echo Server"
Design
This program implements a simple client-server program. The client connects to the server via TCP
and sends it any ASCII text message. The server replies to the client with the same message. The
client then outputs the message the server returned.
The message sent from the client to the server consists of two parts. The first byte sent contains the
length of the message, not including itself. The rest of the message is the message to be repeated
back from the server to the client. The reply message is in the same format. The length is passed in
order to allow the server to be sure it has read all available data. Compiling
To compile this program, go to any Solaris machine. Copy the following files to any subdirectory in
your Unix home directory:
Makefile
client.c
server.c
common.h
Type "make" in the directory where the source code is located. Two binaries should be produced:
"client" and "server". Telnet to another machine. Run "server" on one machine and "client" on the
other machine.
Usage
Server: The server is started with a single command line argument, the port number. The server
will wait on this port until contact is made by the client. Once the server replies to the message sent
by the client, it exits.
Example: #server 2000 (note: the pound sign indicates the Unix prompt)
Client: The client is started with three command line arguments. These arguments include the host,
port number, and message to send. The client will then send the given message to the given host and
port number and wait for a reply. Once the reply is recieved, the client exits.
Example:#client ece2.ece 2000 "This is a test of the emergency broadcast system"
Design Tradeoffs
The program is limited to transmitting a message containing 255 characters. This could be improved
by seperating the message into smaller chunks. The server will continue to receive the chunks until all
has been sent. This could be done with minimal modification of code. All the is needed is some sort of
"all done" message sent between the client and server.
NOTE: There is a quick script called "Test" that will run a simple test case.
Page : 1 2 3 4 5 6 7 8 9 10 11 12 13
OSI
OSI (Open Systems Interconnection)
The OSI Reference Model
In 1983, the International Standards Organization (ISO) created the OSI, or X.200, model. It is a
multilayered model for facilitating the transfer of information on a network. The OSI model is made up
of seven layers, with each layer providing a distinct network service. By segmenting the tasks that
each layer performs, it is possible to change one of the layers with little or no impact on the others. For
example, you can now change your network configuration without having to change your application
or your presentation layer.
The OSI model was specifically made for connecting open systems. These systems are designed to be
open for communication with almost any other system. The model was made to break down each
functional layer so that overall design complexity could be lessened. The model was constructed with
several precepts in mind:
1) Each layer performs a separate function;
2) The model and its levels should be internationally portable; and
3) The number of layers should be architecturally needed, but not unwieldy.
Each layer of the model has a distinct function and purpose:
Application layer--Provides a means for the user to access information on the network through
an application. This layer is the main interface for the user to interact with the application and
therefore the network. Examples include file transfer (FTP), DNS, the virtual terminal (Telnet), and
electronic mail (SMTP).
Presentation layer--Manages the presentation of the information in an ordered and
meaningful manner. This layer's primary function is the syntax and semantics of the data transmission.
It converts local host computer data representations into a standard network format for transmission
on the network. On the receiving side, it changes the network format into the appropriate host
computer's format so that data can be utilized independent of the host computer. ASCII and EBCDIC
conversions, cryptography, and the like are handled here.
Session layer--Coordinates dialogue/session/connection between devices over the network. This
layer manages communications between connected sessions. Examples of this layer are token
management (the session layer manages who has the token) and network time synchronization.
Transport layer--Responsible for the reliable transmission of data and service specification
between hosts. The major responsibility of this layer is data integrity--that data transmitted between
hosts is reliable and timely. Upper layer datagrams are broken down into network-sized datagrams if
needed and then implemented using the appropriate transmission control. The transport layer creates
one or more than one network connection, depending on conditions. This layer also handles what type
of connection will be created. Two major transport protocols are the TCP (Transmission Control
Protocol) and the UDP (User Datagram Protocol
Network layer--Responsible for the routing of data (packets) to a system on the network;
handles the addressing and delivery of data. This layer provides for congestion control, accounting
information for the network, routing, addressing, and several other functions. ). IP (Internet Protocol) is
a good example of a network layer interface.
Data link layer--Provides for the reliable delivery of data across a physical network. This layer
guarantees that the information has been delivered, but not that it has been routed or accepted. This
layer deals with issues such as flow regulation, error detection and control, and frames. This layer has
the important task of creating and managing what frames are sent out on the network. The network
data frame, or packet, is made up of checksum, source address, destination address, and the data
itself. The largest packet size that can be sent defines the maximum transmission unit (MTU).
Physical layer--Handles the bit-level electrical/light communication across the network channel.
The major concern at this level is what physical access method to use. The physical layer deals with
four very important characteristics of the network: mechanical, electrical, functional, and procedural. It
also defines the hardware characteristics needed to transmit the data (voltage/current levels, signal
strength, connector, and media). Basically, this layer ensures that a bit sent on one side of the network
is received correctly on the other side.
Data travels from the application layer of the sender, down through the levels, across the nodes of the
network service, and up through the levels of the receiver. Not all of the levels for all types of data are
needed--certain transmissions might not be valid at a certain level of the model.
To keep track of the transmission, each layer "wraps" the preceding layer's data and header with its
own header. A small chunk of data will be transmitted with multiple layers attached to it. On the
receiving end, each layer strips off the header that corresponds to its respective level.
The OSI model should be used as a guide for how data is transmitted over the network. It is an
abstract representation of the data pathway and should be treated as such.
Physical Layer:
The physical later is concerned with transmitting raw bits over a communication channel. The design
issues have to do with making sure that when one side sends a 1 bit, it is received by the other side as
a 1 bit, not as a 0 bit. Typical questions here are how many volts should be used to represent a 1 and
how many for a 0, how many microseconds a bit lasts, whether transmission may proceed
simultaneously in both directions, how the initial connection is established and how it is torn down
when both sides are finished, and how many pins the network connector has and what each pin is used
for. The design issues here deal largely with mechanical, electrical, and procedural interfaces, and the
physical transmission medium, which lies below the physical layer. Physical layer design can properly
be considered to be within the domain of the electrical engineer.
Analysis of the behavior of the signal mathematically
1. Fourier Analysis
2. Bandwidth-Limited Signals
Hertz (cycles/sec.) = Amplitudes can transmit undiminished from 0 to some frequency, which
is measured in Hertz.
Baud = One signal change per second, a measure of data transmission speed. Named after the
French engineer and telegrapher Jean-Maurice-Emile Baudot and originally used to measure the
transmission speed of telegraph equipment, the term now most commonly refers to the data
transmission speed of a modem.
Baud Rate = The speed at which a modem can transmit data. The baud rate is the number of
events, or signal changes, that occur in one second--not the number of bits per second (bps)
transmitted. In high-speed digital communications, one event can actually encode more than one bit,
and modems are more accurately described in terms of bits per second than baud rate. For example, a
so-called 9,600-baud modem actually operates at 2,400 baud but transmits 9,600 bits per second by
encoding 4 bits per event (2,400 × 4 = 9,600) and thus is a 9,600-bps modem.
3. Maximum data rate of a channel
Nyquist’s Theorem
(For noiseless channels)
Maximum data rate = 2H log2 V bits / sec where, the signal consists of V discrete levels.
Amount of Noise is measured by Signal-to-noise ratio. If S is signal power and N is the noise power
than signal-to-noise ratio = 10 log 10 S/N (measured in [DB] decibels)
Claude Shanon’s
Maximum number of bits/sec = H log 2 (1+S/N)
Transmission Media
1. Magnetic Media
Tapes, Hard disks and Floppies
2. Twisted Pair
Oldest and the most common method of data transmission
Problems of crosstalk and low speed compared to other mediums
Coaxial Cable are available in 2 varieties 50 – ohm (Digital) and 75 – ohm (Analog).
3. Baseband Coaxial Cable (Digital)
Consists of a stiff wire as the core surrounded by an insulating material, encased by a closely woven
braided mesh, surrounded by a protective plastic.
Size = 50 – ohm (Digital)
Hider speed (1 km cable data rate of 1 to 2 Gbps) and less crosstalk
4. Broadband Coaxial Cable (Analog)
Used also for cable TV
Size = 75 – ohm (Analog)
Used for a large areas so they require Analog amplifiers to strengthen the signal periodically. These
amplifiers can transmit signals in only one direction, so dual cable systems are used. If single cable
systems are used than different frequencies for inbound and outbound communication is used.
Technically Broadband is inferior to Baseband but the advantage is that it is already in place as it has
been widely used by cable TV and Telephone Cos.
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5. Fiber Optics
There are 3 components : Light Source, Transmission medium and detector. (A pulse of light indicates
a 1 bit and absence of light indicates a 0 bit)
The transmission medium is of Fiber optic cable made of a center glass core , surrounded by a glass
cladding and protected by a plastic jacket (The glass cladding has a lower index of refraction than the
core so that the light from the core is not leaked)
Normally a transmission medium would leak light and transmission would not be possible but due to
the Physics of refraction the light bounces back into the silica.
A multimode fiber will have different rays of light bouncing around at different angles. In case of a
Single-mode fiber the fiber’s diameter is reduced to a few wave lengths of light and the light is
propagated in a straight line only. This kind of fiber is faster but more expensive. Attenuation
(weakening of transmitted power) in decibels = 10 log 10 transmitted power received power
Fibers can be connected in 3 ways :
1. Connectors are plugged into Fiber sockets (Easy but Connectors lose 10% to 20% light)
2. Spliced Mechanically (10% loss of light)
3. Two fibers can be fused(melted) to form a solid connection (Very small attenuation of light)
But in all three methods at the point of the join the refracted energy can interfere with the signal.
There are two types of light that can pass through a fiber :
LED light (date rate, distance and cost low --- life long --– Multimode --- Not temperature sensitive)
Laser light (date rate, distance and cost high --- Life short –-- Multimode and single mode ---
Temperature sensitive)
Advantages of Fiber Optic Cables :
1. Higher Bandwidth
2. Low attenuation (Repeaters needed every 30 kms compared to every 5 kms in case of copper
cables)
3. Not affected by power surges, electromagnetic interference or power failures
4. Not affected by corrosive chemicals in the air, ideal for harsh factory environment.
5. They don’t leak light hence quite difficult to tap, giving excellent security.
6. Electrons in case of a copper wire are effected by one another and also the stray electrons outside,
while in case of Fiber optic cables Protons are not affected by one another (as they have no electric
charge) and also by stray protons outside.
Disadvantages :
1. Requires skilled engineers.
2. Fiber optic cables are unidirectional, hence two cables or two frequency bands are required.
3. Very costly
Wireless Transmission
The Electromagnetic Spectrum (eg. Wireless)
Radio Transmission (eg. Radios)
Microwave Transmission (eg. TV)
Infrared and millimeter waves (eg. TV remote)
Lightwave transmission (uses lasers)
Telephone System
Problems with Transmission mediums
1. Attenuation (loss of energy)
2. Delay distortion
3. Noise
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Modem
A modem modulates outgoing digital signals from a computer or other digital device to analog signals
for a conventional copper twisted pair telephone line and demodulates the incoming analog signal and
converts it to a digital signal for the digital device.
In recent years, the 2400 bits per second modem that could carry e-mail has become obsolete. 14.4
Kbps and 28.8 Kbps modems were temporary landing places on the way to the much higher bandwidth
devices and carriers of tomorrow. From early 1998, most new personal computers came with 56 Kbps
modems. By comparison, using a digital Integrated Services Digital Network adapter instead of a
conventional modem, the same telephone wire can now carry up to 128 Kbps. With Digital Subscriber
Line (Digital Subscriber Line) systems, now being deployed in a number of communities, bandwidth on
twisted-pair can be in the megabit range.
RS-232-C connector
Multiple xing
1. Frequency Division Multiplexing & Wavelength Division Multiplexing (For Fiber optics) --- Analog
2. Time Division Multiplexing --- Digital Sonet
Switching
1. Circuit Switching, message switching
2. Packet Switching
Switch Hierarchy
1. Crossbar switches
2. Space Division switches
3. Time Division switches
Narrowband ISDN
Integrated Services Digital Network (ISDN) is a set of CCITT/ITU standards for digital transmission over
ordinary telephone copper wire as well as over other media. Home and business users who install an
ISDN adapter (in place of a modem) can see highly-graphic Web pages arriving very quickly (up to 128
Kbps). ISDN requires adapters at both ends of the transmission so your access provider also needs an
ISDN adapter. ISDN is generally available from your phone company in most urban areas in the United
States and Europe.
There are two levels of service: the Basic Rate Interface (BRI), intended for the home and small
enterprise, and the Primary Rate Interface (PRI), for larger users. Both rates include a number of B-
channels and a D-channels. Each B-channel carries data, voice, and other services. Each D-channel
carries control and signaling information.
The Basic Rate Interface consists of two 64 Kbps B-channels and one 16 Kbps D- channel. Thus, a Basic
Rate user can have up to 128 Kbps service. The Primary Rate consists of 23 B-channels and one 64
Kpbs D-channel in the United States or 30 B-channels and 1 D-channel in Europe.
Integrated Services Digital Network in concept is the integration of both analog or voice data together
with digital data over the same network. Although the ISDN you can install is integrating these on a
medium designed for analog transmission, broadband ISDN (BISDN) will extend the integration of both
services throughout the rest of the end-to-end path using fiber optic and radio media. Broadband ISDN
will encompass frame relay service for high-speed data that can be sent in large bursts, the Fiber
Distributed-Data Interface (FDDI), and the Synchronous Opical Network (SONET). BISDN will support
transmission from 2 Mbps up to much higher, but as yet unspecified, rates.
Broadband ISDN – Virtual Circuits
Broadband Integrated Services Digital Network
BISDN is both a concept and a set of services and developing standards for integrating digital
transmission services in a broadband network of fiber optic and radio media. BISDN will encompass
frame relay service for high-speed data that can be sent in large bursts, the Fiber Distributed-Data
Interface (Fiber Distributed-Data Interface), and the Synchronous Optical Network (Synchronous
Optical Network). BISDN will support transmission from 2 Mbps up to much higher, but as yet
unspecified, rates.
BISDN is the broadband counterpart to Integrated Services Digital Network, which provides digital
transmission over ordinary telephone company copper wires on the narrowband local loop.
ATM : asynchronous transfer mode
Asynchronous transfer mode (ATM) is a dedicated-connection switching technology that organizes
digital data into 53-byte cell units and transmits them over a physical medium using digital signal
technology. Individually, a cell is processed asynchronously relative to other related cells and is
queued before being multiplexed over the transmission path.
Because ATM is designed to be easily implemented by hardware (rather than software), faster
processing and switch speeds are possible. The prespecified bit rates are either 155.520 Mbps or
622.080 Mbps. Speeds on ATM networks can reach 10 Gbps. Along with Synchronous Optical Network
(SONET) and several other technologies, ATM is a key component of broadband ISDN (BISDN).
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ATM switches
Knockout switch
Batcher-Banyan switch
Cellular Radio
Paging
Cordless telephones
Analog Cellular telephones
Digital Cellular telephones
Personal communication systems
Communication satellites
1. Geo-synchronous
2. Low orbit
X.21
A digital signaling interface called X.21 was recommended by the CCITT in 1976. The recommendation
specifies how the customer's computer, the DTE, sets up and clears calls by exchanging signals with
the carrier's equipment, the DCE.
The names and functions of the eight wires defined by X.21 are given in the following figure. The
physical connector has 15 pins, but not all of them are used. the DTE uses the T and C lines to transmit
data and control information, respectively. The DCE uses the R and I lines for data and control. The S
line contains a signal stream emitted by the DCE to provide timing information, so the DTE knows
when each bit interval starts and stops. At the carrier's option, a B line may also be provided to group
the bits into 8-bit frames. If this option is provided, the DTE must begin each character on a frame
boundary. If the option is not provided, both DTE and DCE must begin every control sequence with at
least two SYN characters, to enable the other one to deduce the implied frame boundaries.
Although X.21 is a long and complicated document, the simple example of the next figure illustrates
the main features. In this example it is shown how the DTE places a call to a remote DTE, and how the
originating DTE clears the call when it is finished. To make the explanation clearer, the calling and
clearing procedures is described in terms of an analogy with the telephone system.
