Introducción “El Imperio Romano”

Dialup is simply the application of the public switched telephone network (PSTN) that carries data on behalf of the end user. It involves a customer premises equipment (CPE) device sending the telephone switch a phone number to which to direct a connection. The carrier market has grown significantly, and the market now demands higher modem densities. The answer to this need is a higher degree of interoperation with the telephone company equipment and the development of the digital modem. This is a modem that is

capable of direct digital access to the PSTN. As a result, faster CPE modems have now been developed that take advantage of the clarity of signal that the digital modems enjoy. The fact that the digital modems connecting into the PSTN through a PRI or BRI can transmit data at over 53k using the V.90 communication standard, attests to the success of the idea.

A couple of outdated technologies bear mentioning in a historical discussion of dialer technology. Today dialup is still used as an economical alternative(depending on the connection requirements) to dedicated connectivity. It has important uses as backup connectivity, in case the primary lines go down. Dialup also offers the flexibility to create dynamic connections as needed.(Figure-3.24)

Figure –3.24 : Sample Dialup Connection infrastructure 3.6.1 Dialup Connectivity Technology

Plain Old Telephone Service

The Regular phone lines used in voice calls are referred to as Plain old telephone service (POTS). They are ubiquitous, familiar, and easy to obtain; local calls are normally free of charge. This is the kind of service that the telephone network was built on. Sounds carrier over this service are sampled at rate of 8000 times per second(using 8 bits per sample) in their conversion to digital signals so that sound can be carried on a 64 kbps channel at acceptable levels.(According to Nyquist Theorem max frequency of voice signals is 4000 Hz.)

The encoding and decoding of voice is done by a piece of telco gear called a CODEC. The CODEC was needed to allow backward-compatibility with the old analog phones that were already in widespread use when the digital network was introduced. Thus, most phones found in the home are simple analog devices.

Dialup connectivity across POTS lines has historically been limited to about 33,600 bps via modem –often referred to as V.34 Speeds. Recent improvements have increased the speed at which data can be sent from a digital source to a modem on a POTS line, but using POTS lines on both ends of the connection still results in V.34 connectivity in both directions.

Basic Rate Interface

Intended for home use, this application of ISDN uses the same copper as a POTS line, but it offers direct digital connectivity to the telephone network. A special piece of equipment known as terminal adapter is required(although, depending on the country, it may be integrated into the router or DCE device). Always make sure to check-the plug used to connect to the wall socket looks the same whether it’s the S/T or U demarcation point. Normally, a Basic Rate Interface (BRI) interface has two B (bearer) channels to carry data, and one D(delta) channel to carry control and signalling information. Local telephone carriers may have different plans to suit local needs. Each B channel is a 64 K line. The

individual 64k channels of the telephone network are commonly referred to as digital service 0(DS0). This is a common denominator regardless of the types of services offered. The BRI interface is a dedicated to the switch and will remain up even if no calls are placed.

Modem

A modem is a device that interprets digital and analog signals, enabling data to be transmitted over voice-grade telephone lines. At the source, digital signals are converted to a form suitable for transmission over analog communication facilities. At the destination, these analog signals are returned to their digital form. Figure-3.25 illustrates a simple modem-to-modem connection through a WAN.

Figure-3.25: A modem connection through a WAN handles analog and digital signals. Advances in access server technology have allowed the development of new standards to take advantage of better connectivity at the server end. Theses standards are X2,56K flex and V.90. X2 and 56 Kflex is an older (pre-V.90) 56k modem standard that was proposed by Rockwell. The assumption made by all these standards in that the access server has digital connectivity to the telephone network. The model is shown in Figure-3.26.

Figure-3.26: Modem connection to the PSTN 3.6.2 T Carrier Technical Summary

What is a T Carrier?

T Carrier is a generic name for any of several digitally multiplexed carrier systems. T Carrier systems were originally designed to transmit digitized voice signals. Current applications also include digital data transmission. T1 and T3 are the most popular T Carrier links used in United States and Canada. E1 and E3 are similar links used in Europe, and J1 and J3 are similar links used in Japan. The North American, European, and Japanese versions differ somewhat in the transmission rates and signalling protocols used. The "T" in T Carrier stands for "Trunk". If an "F" precedes the "T", it refers to an optical fiber cable system, but at the same speeds.

T Carrier Rates

North American Hierarchy Designator Transmission Rate Number of Voice Channels DS-0 64 Kbps 1 T-1 (DS-1) 1.544 Mbps 24 T-1C (DS-1C) 3.152 Mbps 48 T-2 (DS-2) 6.312 Mbps 96 T-3 (DS-3) 44.736 Mbps 672 T-4 (DS-4) 274.176 Mbps 4032 European Hierarchy Designator Transmission Rate Number of Voice Channels E-0 64 Kbps 1 E-1 2.048 Mbps 30 E-2 8.448 Mbps 120 E-3 34.368 Mbps 480 E-4 139.268 Mbps 1920 E-5 565.148 Mbps 7680 Japanese Hierarchy Designator Transmission Rate Number of Voice Channels J-0 64 Kbps 1 J-1 1.544 Mbps 24 J-1C 3.152 Mbps 48 J-2 6.312 Mbps 96 J-3 32.064 Mbps 480 J-3C 97.728 Mbps 1440 J-4 397.200 Mbps 5760 T-1

T-1, also spelled T-1, stands for Trunk Level 1. It is a digital transmission link with a total signalling speed of 1.544 Mbps (1,544,000 bits per second). T-1 is a standard for digital transmission in North America - the United States and Canada. It is part of a hierarchy of digital transmission pipes known generically as the DS (Digital Signal Level) hierarchy. T- 1 equates to DS-1 in the DS hierarchy.

A T-1 link has the following characteristics:

• Four wire circuit. T-1 originated from phone circuits using two pairs of unshielded twisted copper wires - one pair for transmit and one pair for receive. Today, T-1 is often delivered on fiber optic media. It can also run over other media types such as coaxial cable, microwave, or satellite systems.

• Full duplex. Transmission and reception of data can take place simultaneously.

• Digital. T-1 is an all-digital service. Data, analog voice, and analog fax are all converted into digital pulses (1s and 0s) for transmission on the line.

• Time-division multiplexing. T-1 consists of digital streams carrying 64-kbps in each channel. 24 channels are multiplexed together to create an aggregate of 1.536 Mbps. Time-division allows a channel to use a slot one-twenty-fourth of the time. These can be fixed time slots made available to each channel.

• Pulse code modulation. Analog voice or any other service is sampled 8,000 times a second. An 8-bit word represents each sample, thus yielding a 64-kbps channel capacity.

• Framed format. As the pulse code modulation scheme is used, the 24 channels are time-division multiplexed into a frame to be carried along the line. Each frame represents an 8-bit sample for each of the 24 channels. Added to this is the framing bit. The net result is a 193-bit frame. There are 8,000 frames per second, therefore a frame is 125 microseconds long. Framing accounts for 8-kbps overhead (1-bit x 8,000 frames). Adding this 8-kpbs to the 1.536 Mbps described above yields an aggregate of 1.544 Mbps.

• Bipolar format. T-1 uses an electrical voltage across the line to represent the pulses (1s). The bipolar format serves two purposes: it reduces the required bandwidth from 1.5 MHz to 770 kHz (which increases repeater spacing), and it averages out the signal voltage to zero to allow dc power to be simplexed on the line to power intermediate regenerators. Every other pulse will be represented by the negative equivalent of the pulse. For example, the first pulse will be represented by a positive 3 volts (+3V), the next pulse will be represented by negative 3 volts (- 3V) and so on. This effectively yields a 0 voltage on the line, since the positives and negatives equalize the current. This bipolar format is also called alternate mark inversion (AMI). The mark is a digital one (1). Alternate ones are inverted in polarity (+, -).

• Byte synchronous transmission. Each sample is made up of 8 bits from each channel. Timing for the channels is derived for the pulses that appear within the samples. If a long series of zeros are transmitted, the device on the receiving end of the line may lose synchronization with the transmitter due to the lack of pulses on the line. This can cause bits that are part of one channel to be mistakenly interpreted as part of another channel. To avoid this problem, techniques such as bit stuffing and bipolar 8 zero substitution (B8ZS) are used to ensure that 1s appear on the line frequently enough to maintain synchronization.

• Channelized or unchannelized. Generically, T-1 is 24 channels of 64 kbps each plus 8 kbps of overhead. This is considered channelized service. However, the multiplexing equipment can be configured in a number of ways. For example, the T-1 can be used as a single channel of 1.536 Mbps. Or it can be configured as two high-speed data channels at 384 kbps each, and a video channel at 768 kbps. Or as a high-speed data channel of 512 kbps, plus 16 channels of lower speed data and voice at 64 kbps each. These examples can be mixed into a variety of offerings. The point is that the service does not have to be "channelized" into 24 separate streams, but can be "unchannelized" into any usable data stream needed (equipment allowing of course).

Fractional T-1

Leasing a T-1 line means paying for the entire 1.544 Mbps bandwidth 24 hours a day, whether it is used or not. Fractional T-1 (FT-1) lets you lease any 64 kbps submultiple of a T-1 line. You might, for example, lease only six of the 64 kbps channels for an aggregate bandwidth of 384 kbps. Fractional T-1 is useful whenever the cost of a dedicated T-1 would be prohibitive.

T-3

T-3 is the North American standard for DS-3 (Digital Signal Level 3). T-3 operates at a signaling rate of 44.736 Mbps, equivalent to 28 T-1s. It is commonly referred to as 45

Megabits per second, rounded up. Capable of handling 672 voice conversations, T-3 runs on fiber optic or microwave transmission media, as twisted pair is not capable of supporting such a high signaling rate over long distances

For dialup purposes, there are two types of T1/E1: Primary Rate ınterface(PRI) and Channel Associated Signalling(CAS). PRI As CAS T1/E1s are normally seen in central locations that receive calls from remote sites or customers.

Primary Rate Interface

T1 Primary rate interface(PRI) service offers 23B channels at 64 kbps at the cost of one D- channel(the 24th channel) for call signalling. Using NFAS to allow multiple PRIs to use a single D channel can minimize this disadvantage. E1 PRI service Allows 30 channels, but it uses the 16th channel for ISDN signalling. The PRI service is an ISDN connection. It allows either voice-grade(modem) or true ISDN calls to be made and received through the T1/E1. This is the type of service most often seen in access servers because it fosters higher connection speeds.

Channel Associated Signalling

T1 Channel associated signalling(CAS) lines have 24 56K channels-part of each channel is borrowed for call signalling. This type of service is also called robbed-bit signalling. The E1 CAS still uses only the 16th channel for call signalling, but is uses the R2 international standard for analog call signals.

CAS is not an ISDN interface;it allows only analog calls to come into the access server. This is often done to allow an access server to work with a channel bank, and this scenario is seen more commonly in south Americe, Europe, and Asia.

Phases of PPP Negotiation

PPP negotiation consists of three phases: Link Control Protocol (LCP), Authentication, and Network Control Protocol (NCP). Each proceeds in order, following the establishment of the async or ISDN connection.

LCP

PPP does not follow a client/server model. All connections are peer-to-peer. Therefore, when there is a caller and a receiver, both ends of the point-to-point connection must agree on the negotiated protocols and parameters.

When negotiation begins, each of the peers wanting to establish a PPP connection must send a Configure Request (seen in debug ppp negotiation and referred to hereafter as CONFREQ). Included in the CONFREQ are any options that are not the link default. These often include Maximum Receive Unit (MRU), Async Control Character Map (ACCM), Authentication Protocol (AuthProto), and the Magic Number. Also seen are the Maximum Receive Reconstructed Unit (MRRU) and Endpoint Discriminator (EndpointDisc), used for Multilink PPP.

There are three possible responses to any CONFREQ:

• A Configure-Acknowledge (CONFACK) must be issued if the peer recognizes the options and agrees to the values seen in the CONFREQ.

• A Configure-Reject (CONFREJ) must be sent if any of the options in the CONFREQ are not recognized (for instance, some vendor-specific options) or if the values for any of the options have been explicitly disallowed in the configuration of the peer.

• A Configure-Negative-Acknowledge (CONFNAK) must be sent if all the options in the CONFREQ are recognized, but the values are not acceptable to the peer. The two peers continue to exchange CONFREQs, CONFREJs and CONFNAKs until each sends a CONFACK, until the dial connection is broken, or until one or both of the peers indicates that the negotiation can not be completed.

Authentication

After the successful completion of LCP negotiation and reaching an agreement on AuthProto, the next step is authentication. Authentication, while not mandatory per RFC1661, is highly recommended on all dial connections. In some instances, it is a requirement for proper operation; Dialer Profiles being a case in point.

The two principal types of authentication in PPP are the Password Authentication Protocol (PAP) and the Challenge Handshake Authentication Protocol (CHAP), defined by RFC1334 and updated by RFC1994.

PAP is the simpler of the two, but is less secure because the plain-text password is sent across the dial connection. CHAP is more secure because the plain-text password is not ever sent across the dial connection.

PAP may be necessary in one of the following environments:

• A large installed base of client applications that do not support CHAP

• Incompatibilities between different vendor implementations of CHAP

When discussing authentication, it is helpful to use the terms "requester" and "authenticator" to distinguish the roles played by the devices at either end of the connection, though either peer can act in either role. "Requester" describes the device that requests network access and supplies authentication information; the "authenticator" verifies the validity of the authentication information and either allows or disallows the connection. It is common for both peers to act in both roles when a DDR connection is being made between routers.

PAP

PAP is fairly simple. After successful completion of the LCP negotiation, the requester repeatedly sends its username/password combination across the link until the authenticator responds with an acknowledgment or until the link is broken. The authenticator may disconnect the link if it determines that the username/password combination is not valid. CHAP

CHAP is somewhat more complicated. The authenticator sends a challenge to the requester, which then responds with a value. This value is calculated by using a "one-way hash" function to hash the challenge and the CHAP password together. The resulting value is sent to the authenticator along with the requester’s CHAP hostname (which may be different from its actual hostname) in a response message.

The authenticator reads the hostname in the response message, looks up the expected password for that hostname, and then calculates the value it expects the requester sent in its response by performing the same hash function the requester performed. If the resulting values match, the authentication is successful. Failure should lead to a disconnect.

NCP

After successful authentication, the NCP phase begins. As in LCP, the peers exchange CONFREQs, CONFREJs, CONFNAKs and CONFACKs. However, in this phase of negotiation, the elements being negotiated have to do with higher layer protocols–IP, IPX, Bridging, CDP, and so on. One or more of these protocols may be negotiated. As it is the most commonly used, and because other protocols operate in much the same fashion,

Internet Protocol Control Protocol (IPCP), defined in RFC1332, is the focus of this discussion. Other pertinent RFCs include, but are not limited to:

• RFC1552 (IPX Control Protocol)

• RFC1378 (AppleTalk Control Protocol)

• RFC1638 (Bridging Control Protocol)

• RFC1762 (DECnet Control Protocol)

• RFC1763 (Vines Control Protocol) Alternate PPP Methodologies

Alternate PPP methodologies include multilink PPP, multichassis PPP, and virtual profiles. Multilink PPP

The Multilink Point-to-Point Protocol (MLP) feature provides load-balancing functionality over multiple WAN links. At the same time it provides multi-vendor interoperability, packet fragmentation and proper sequencing, and load calculation on both inbound and outbound traffic. The Cisco implementation of Multilink PPP supports the fragmentation and packet sequencing specifications in RFC1717.

Multilink PPP allows packets to be fragmented. These fragments can be sent at the same time over multiple point-to-point links to the same remote address. The multiple links come up in response to a dialer load threshold that you define. The load can be calculated on inbound traffic, outbound traffic, or on either, as needed for the traffic between the specific sites. MLP provides bandwidth on demand and reduces transmission latency across WAN links.

Multilink PPP works over the following interface types (single or multiple) which are configured to support both dial-on-demand rotary groups and PPP encapsulation:

• asynchronous serial interfaces

• BRIs

• PRIs

Multichassis Multilink PPP

Multilink PPP provides the capability of splitting and recombining packets to a single end- system across a logical pipe (also called a bundle) formed by multiple links. Multilink PPP provides bandwidth on demand and reduces transmission latency across WAN links. Multichassis Multilink PPP (MMP), on the other hand, provides the additional capability for links to terminate at multiple routers with different remote addresses. MMP can also handle both analog and digital traffic.

This functionality is intended for situations in which there are large pools of dial-in users, in which a single access server cannot provide enough dial-in ports. MMP allows companies to provide a single dialup number to its users and to apply the same solution to