RouterStuff

Senin, 08 Juni 2009

Voice over ATM - signaling, addressing, routing, delay

Voice over ATM

The ATM Forum and the ITU have specified different classes of services to represent different possible traffic types for VoATM.

Designed primarily for voice communications, constant bit rate (CBR) and variable bit rate (VBR) classes have provisions for passing real-time traffic and are suitable for guaranteeing a certain level of service. CBR, in particular, allows the amount of bandwidth, end-to-end delay, and delay variation to be specified during the call setup.

Designed principally for bursty traffic, unspecified bit rate (UBR) and available bit rate (ABR) are more suitable for data applications. UBR, in particular, makes no guarantees about the delivery of the data traffic.

The method of transporting voice channels through an ATM network depends on the nature of the traffic. Different ATM adaptation types have been developed for different traffic types, each with its benefits and detriments. ATM adaptation layer 1 (AAL1) is the most common adaptation layer used with CBR services.

Unstructured AAL1 takes a continuous bit stream and places it within ATM cells. This is a common method of supporting a full E1 byte stream from end to end. The problem with this approach is that a full E1 may be sent, regardless of the actual number of voice channels in use. (An EI is a wide-area digital transmission scheme used predominantly in Europe that carries data at a rate of 2.048 Mbps.)

Structured AAL1 contains a pointer in the payload that allows the digital signal level 0 (DS0) structure to be maintained in subsequent cells. This allows network efficiencies to be gained by not using bandwidth for unused DS0s. (A DS0 is a framing specification used in transmitting digital signals over a single channel at 64 kbps on a T1 facility.)

The remapping option allows the ATM network to terminate structured AAL1 cells and remap DS0s to the proper destinations. This eliminates the need for permanent virtual circuits (PVCs) between every possible source/destination combination. The major difference from the previous approach is that a PVC is not built across the network from edge to edge.

VoATM Signaling

Figure 19-2 describes the transport method, in which voice signaling is carried through the network transparently. PVCs are created for both signaling and voice transport. First, a signaling message is carried transparently over the signaling PVC from end station to end station. Second, coordination between the end systems allows the selection of a PVC to carry the voice communication between end stations.

Figure 19-2 The VoATM Signaling Transport Model Describes the Transport Method, in Which Voice Signaling Is Carried Through the Network Transparently

At no time is the ATM network participating in the interpretation of the signaling that takes place between end stations. However, as a value-added feature, some products are capable of understanding channel associated signaling (CAS) and can prevent the sending of empty voice cells when the end stations are on-hook.

Figure 19-3 shows the translate model. In this model, the ATM network interprets the signaling from both non-ATM and ATM network devices. PVCs are created between the end stations and the ATM network. This contrasts with the previous model, in which the PVCs are carried transparently across the network.

Figure 19-3 In the VoATM Signaling Translate Model, the ATM Network Interprets the Signaling from Both Non-ATM and ATM Network Devices

A signaling request from an end station causes the ATM network to create an SVC with the appropriate QoS to the desired end station. The creation of an SVC versus the prior establishment of PVCs is clearly more advantageous for three reasons:

•SVCs are more efficient users of bandwidth than PVCs.

•QoS for connections do not need to be constant, as with PVCs.

•The capability to switch calls within the network can lead to the elimination of the tandem private branch exchange (PBX) and potentially the edge PBX. (A PBX is a digital or analog telephone switchboard located on the subscriber premises and used to connect private and public telephone networks.)

VoATM Addressing

ATM standards support both private and public addressing schemes. Both schemes involve addresses that are 20 bytes in length (shown in Figure 19-4).

Figure 19-4 ATM Supports a 20-Byte Addressing Format

The Authority and Format Identifier (AFI) identifies the particular addressing format employed. Three identifiers are currently specified: data country code (DCC), international code designator (ICD), and E.164. Each is administered by a standards body. The second part of the address is the initial domain identifier (IDI). This address uniquely identifies the customer's network. The E.164 scheme has a longer IDI that corresponds to the 15-digit ISDN network number. The final portion, the domain-specific part (DSP), identifies logical groupings and ATM end stations.

In a transport model, you don't need to be aware of the underlying addressing used by the voice network. However, in the translate model, the capability to communicate from a non-ATM network device to an ATM network device implies a level of address mapping. Fortunately, ATM supports the E.164 addressing scheme, which is employed by telephone networks throughout the world.

VoATM Routing

ATM uses a private network-to-network interface (PNNI), a hierarchical link-state routing protocol that is scalable for global usage. In addition to determining reachability and routing within an ATM network, it is also capable of call setup.

A virtual circuit (VC) call request causes a connection with certain QoS requirements to be requested through the ATM network. The route through the network is determined by the source ATM switch based on what it determines is the best path through the network, based on the PNNI protocol and the QoS request. Each switch along the path is checked to determine whether it has the appropriate resources for the connection.

When the connection is established, voice traffic flows between end stations as if a leased line existed between the two. This specification spells out routing in private networks. Within carrier networks, the switch-to-switch protocol is B-ICI. Current research and development of integrated non-ATM and ATM routing will yield new capabilities to build translate-level voice and ATM networks.

VoATM and Delay

ATM has several mechanisms for controlling delay and delay variation. The QoS capabilities of ATM allow the specific request of constant bit rate traffic with bandwidth and delay variation guarantees. The use of VC queues allows each traffic stream to be treated uniquely. Priority can be given for the transmission of voice traffic. The use of small, fixed-size cells reduces queuing delay and the delay variation associated with variable-sized packets.

Voice over ATM

Voice over ATM (VoATM) can be supported as standard pulse code modulated (PCM) voice via circuit emulation (AAL1, described later) or as variable bit rate voice in ATM cells as AAL2 (also described later). ATM offers many advantages for transport and switching of voice. First, quality of service (QoS) guarantees can be specified by service provisioning or on a per-call basis. In addition, call setup signaling for ATM switched virtual circuits (SVCs), Q.2931, is based on call setup signaling for voice ISDN, Q.931. Administration is similar to circuit-based voice networks.

However, VoATM suffers from the burden of additional complexity and incomplete support and interoperability among vendors. It also tends to be more expensive because it is oriented toward all optical networks. Most importantly, ATM is typically deployed
as a WAN Layer 2 protocol and therefore does not extend all the way to the desktop. Nevertheless, ATM is quite effective for providing trunking and tandem switching services between existing voice switches and PBXs.

Voice over Frame Relay (VoFR) has become widely deployed across many networks. Like VoATM, it is typically employed as a tie trunk or tandem-switching function between remote PBXs. It benefits from much simpler administration and relatively lower cost than VoATM, especially when deployed over a private WAN network. It also scales more economically than VoATM, supporting links from T1 down to 56 kbps. When deployed over a carefully engineered Frame Relay network, VoFR works very well and provides good quality. However, voice quality over Frame Relay can suffer depending on network latency and jitter. Although minimal bandwidth and burstiness are routinely contracted, latency and jitter are often not included in service level agreements (SLAs) with service providers. As a result, voice performance can vary. Even if quality is good at first, voice quality can degrade over time as a service provider's network becomes saturated with more traffic. For this reason, many large enterprise customers are beginning to specify latency and jitter, as well as overall packet throughput from carriers. In these situations, voice over Frame Relay can provide excellent service.

Voice over IP (VoIP) has begun to be deployed in recent years as well. Unlike voice over Frame Relay and Voice over ATM, Voice over IP is a Layer 3 solution, and it offers much more value and utility because IP goes all the way to the desktop. This means that in addition to providing basic tie trunk and tandem-switching functions to PBXs, VoIP can actually begin to replace those PBXs as an application. As a Layer 3 solution, VoIP is routable and can be carried transparently over any type of network infrastructure, including both Frame Relay and ATM. Of all the packet voice technologies, VoIP has perhaps the most difficult time supporting voice quality because QoS cannot be guaranteed. Normal applications such as TCP running on IP are insensitive to latency but must retransmit lost packets due to collisions or congestion. Voice is much more sensitive to packet delay than packet loss. In addition to normal traffic congestion, QoS for VoIP is often dependent on lower layers that are ignorant of the voice traffic mingled with the data traffic.

Voice Networking

Basic voice technology has been available for more than 100 years. During that time, the technology has matured to the point at which it has become ubiquitous and largely invisible to most users. This legacy of slow evolution continues to affect today's advanced voice networks in many ways, so it is important to understand the fundamentals of traditional voice technology before emulating it on data networks.

Traditional analog telephone instruments used for plain old telephone service (POTS) use a simple two-wire interface to the network. They rely on an internal two-wire/four-wire hybrid circuit to combine both transmit and receive signals. This economical approach has been effective but requires special engineering regarding echo.

Basic Telephony

Three types of signaling are required for traditional telephony: supervision, alerting, and addressing. Supervision monitors the state of the instrument—for example, allowing the central office or PBX to know when the receiver has been picked up to make a call, or when a call is terminated. Alerting concerns the notification of a user that a call is present (ringing) or simple call progress tones during a call (such as busy, ringback, and so on). Finally, addressing enables the user to dial a specific extension.

In addition to signaling, telephony services also provide secure media transport for the voice itself, analog-to-digital conversion, bonding and grounding for safety, power, and a variety of other functions when needed.

Analog voice interfaces have evolved over the years to provide for these basic functions while addressing specific applications. Because basic POTS two-wire analog interfaces operate in a master/slave model, two basic types of analog interfaces are necessary for data equipment to emulate: the user side and the network side. The user side (telephone) expects to receive power from the network as well as supervision.

A foreign exchange service (FXS) interface is used to connect an analog telephone, fax machine, modem, or any other device that would be connected to a phone line. It outputs 48 vdc power, ringing, and so on, and it accepts dialed digits. The opposite of an FXS interface is a foreign exchange office (FXO) interface. It is used to connect to a switching system providing services and supervision, and it expects the switch to provide supervision and other elements. (Why "foreign"? The terms FXS and FXO were originally used within telephone company networks to describe provision of telephone service from a central office other than normally assigned.)

Within FXS and FXO interfaces, it is also necessary to emulate variants in supervision. Typical telephones operate in a loop start mode. The telephone normally presents a high impedance between the two wires. When the receiver goes off-hook, a low-impedance closed circuit is created between the two wires. The switch, sensing current flow, then knows that the receiver is off-hook and applies a dial tone. The switch also checks to be sure that the receiver is on-hook before sending a ringing signal. This system works well for simple telephones, but it can cause problems on trunks between PBXs and COs with high activity. In that situation, the remote end and the CO switch can both try to seize the line at the same time. This situation, called glare, can freeze the trunk until one side releases it. The solution is to short tip or ring to ground as a signal for line seizure rather than looping it. This is called ground start.

After the line is seized, it is necessary to dial the number. Normal human fingers cannot outrun the dial receivers in a modern switch, but digits dialed by a PBX can. In that case, many analog trunks use a delay start or wink start method to notify the calling device when the switch is ready to accept digits.

Another analog interface often used for trunking is E&M. This is a four- or six-wire interface that includes separate wires for supervision in addition to the voice pair. E&M stands for "ear and mouth" or "Earth and magneto" and is derived from the early telephony days. The E&M leads are used to signal on-hook and off-hook states.

Analog voice works well for basic trunk connections between switches or PBXs, but it is uneconomical when the number of connections exceeds six to eight circuits. At that point, it is usually more efficient to use digital trunks. In North America, the T1 (1.544 Mbps) trunk speed is used, consisting of 24 digitized analog voice conversations. In other parts of the world, E1 (2.048 Mbps) is used to carry 30 voice channels. (Engineers refer to the adoption of E1 and T1 internationally as "the baseball rule"—there is a strong correlation of countries that play baseball to the use of T1. Therefore, the United States, Canada, and Japan have the largest T1 networks, while other countries use E1.)

The first step in conversion to digital is sampling. The Nyquist theorem states that the sampling frequency should be twice the rate of the highest desired frequency. Early telephony engineers decided that a range of 4000 hertz would be sufficient to capture human voices (which matches the performance of long analog loops). Therefore, voice channels are sampled at a rate of 8000 times per second, or once every 125 ms. Each of these samples consists of an 8-bit measurement, for a total of 64000 bits per second to be transmitted. As a final step, companding is used to provide greater accuracy of low-amplitude components. In North America, this is u-law (mu-law), while elsewhere it is typically A-law. For international interworking purposes, it is agreed that the North American side will make the conversion.

To construct a T1, 24 channels are assembled for a total of 1.536 Mbps, and an additional 8 bits are added every 125 ms for framing, resulting in a rate of 1.544 Mbps. Often, T1 frames are combined into larger structures called SuperFrames (12 frames) and Extended-SuperFrames (24 frames). Additional signaling can then be transmitted by "robbing bits" from the interior frames.

Basic T1 and E1 interfaces emulate a collection of analog voice trunks and use robbed bit signaling to transfer supervisory information similar to the E&M analog model. As such, each channel carries its own signaling, and the interface is called channel associated signaling (CAS). A more efficient method uses a common signaling channel for all the voice channels. Primary Rate Interface for ISDN is the most common example of this common channel signaling (CCS).

If voice/data integration is to be successful, all of these voice interfaces must be supported to provide the widest possible range of applications. Over the years, users have grown to expect a certain level of performance, reliability, and behavior of a telecommunications system, which must be supported going forward. All these issues have been solved by various packet voice systems today so that users can enjoy the same level of support to which they have become accustomed.

	PDU #1 is a frame.
	PDU #2 is an application layer PDU
	PDU #3 is a segment.
	PDU #4 is a transport layer PDU.
	The order in which these PDUs are created during the encapsulation process is 3, 4, 1, 2.
	The order in which these PDUs are created during the encapsulation process is 2, 1, 4, 3.

	SMTP
	FTP
	SNMP
	HTTP
	TFTP
	DHCP

	Telnet
	SNMP
	SMTP
	SQL
	TFTP

	establishment of logical connections between source and destination hosts
	end-to-end flow control
	determination of best paths through the network
	definition of the procedures used to interface with the network hardware
	packet switching
	Layer 5 and Layer 6 OSI model functions

	3 collision domains
	3 broadcast domains
	5 broadcast domains
	9 collision domains
	10 collision domains
	13 collision domains

RouterStuff

Kamis, 11 Juni 2009

CCNA Practice Certification Exam # 2- CCNA Exploration: Accessing the WAN (Version 4.0)

Senin, 08 Juni 2009

Voice over ATM - signaling, addressing, routing, delay

Voice over ATM

VoATM Signaling

VoATM Addressing

VoATM Routing

VoATM and Delay

Voice over ATM

Voice Networking

Basic Telephony

	Switch(config-if)# vlan all
	Switch(config-if)# switchport access vlan 30
	Switch(config-if)# switchport access vlan all
	Switch(config-if)# switchport mode trunk
	Switch(config-if)# switchport trunk encapsulation dot1q
	Switch(config-if)# switchport trunk encapsulation isl

	All packets will be dropped.
	The Telnet session will succeed.
	An error message that says “Session terminated” will appear.
	An error message that says “Password required, but none set” will appear.

	The Spanning Tree Protocol has been disabled on one switch.
	The Spanning Tree Protocol has been disabled on both switches.
	The IEEE 802.1Q trunking port has a speed mismatch on one of the switches.
	The SwA port is configured as a trunk port and the SwB port is configured as an access port.
	The SwA port has IEEE 802.1Q trunking enabled and the SwB port has ISL trunking enabled.

	The switch port is configured as an access mode port.
	The switch port does not support trunking on this port.
	The encapsulation type must be changed to a compatible protocol before the port can be placed in trunk mode.
	The no switchport trunk encapsulation auto command needs to be applied to the switch port before trunking can be enabled.

	RT_1(config)# interface fa0/1 RT_1(config-if)# no shutdown
	SW_1(config)# interface fa0/24 SW_1(config-if)# switchport mode client
	RT_1(config)# interface fa0/1 RT_1(config-if)# encapsulation trunk dot1q 24
	SW_1(config)# interface fa0/24 SW_1(config-if)# switchport mode trunk
	SW_1(config)# interface fa0/24 SW_1(config-if)# switchport access vlan 1

	SW_1(config)# username david password class
	SW_1(config)# line vty 0 15 SW_1(config-line)# login SW_1(config-line)# password cisco
	SW_1(config)# line vty 0 15 SW_1(config-line)# login local SW_1(config-line)# transport input ssh
	SW_1(config)# login ssh SW_1(config)# password class
	SW_1(config)# login key rsa SW_1(config)# password cisco
	SW_1(config)# password encrypted cisco

	improper placement of enterprise level servers
	addition of hosts to a physical segment
	replacement of hubs with workgroup switches
	increasing use of bandwidth intensive network applications
	creation of new collision domains without first adding network hosts
	migration to full-duplex Ethernet within the LAN

	designated switch
	edge switch
	root bridge
	enabled bridge
	local bridge

	1) fast forward 2) fragment free 3) store-and-forward
	1) store-and-forward 2) fragment free 3) fast forward
	1) fragment free 2) fast forward 3) store-and-forward
	1) fast forward 2) store-and-forward 3) fragment free

	hold-down timers
	poison reverse
	spanning tree protocol
	time to live
	VTP

	802.1Q
	CDP
	RIP
	RSTP

	Spanning Tree
	RIP v2
	IEEE 802.1Q
	ARP
	Rapid Spanning Tree

	listening
	learning
	blocking
	disabled
	forwarding

	192.168.16.63/27
	192.168.16.158/27
	192.168.16.192/27
	192.168.16.113/30
	192.168.16.145/30
	192.168.16.193/30

	ip dhcp server
	no ip domain-lookup
	no logging synchronous
	no show ip nat translations

	10.0.0.0/8
	64.100.0.0/14
	127.16.0.0/12
	172.16.0.0/12
	192.31.7.0/24
	192.168.0.0/16

	The default gateway configured on the network administrator's workstation is not a valid host address.
	The default gateway configured on the TFTP server in incorrect.
	The IP address of the TFTP server is a subnetwork address.
	The IP address of the S0/0/0 interface of the Heliopolis router is a broadcast address.
	The IP addresses for the Memphis S0/0/1 interface and the Heliopolis S0/0/0 interface are not in the same subnet.

	172.16.0.0
	172.16.134.0
	172.16.134.32
	172.16.134.48
	172.16.134.47
	172.16.134.63

	low processor overhead
	poison reverse
	routing loops
	split horizon
	shortest-path first calculations

	router ospf 1 network 192.168.10.0
	router ospf 1 network 192.168.10.64 0.0.0.63 area 0 network 192.168.10.192 0.0.0.3 area 0
	router ospf 1 network 192.168.10.64 255.255.255.192 network 192.168.10.192 255.255.255.252
	router ospf 1 network 192.168.10.0 area 0

	to activate the OSPF neighboring process
	to influence the DR/BDR election process
	to provide a backdoor for connectivity during the convergence process
	to streamline and speed up the convergence process

	The clock rate is not set on the DTE.
	An incorrect default gateway is set on the router.
	A network is missing from the routing protocol configuration.
	The encapsulation on the Serial0 interface is incorrect.
	Keepalives are not being sent by the remote device.

	Router# show ip interfaces brief
	Router# show ip route
	Router# show cdp neighbors
	Router# show running-config
	Router# show protocols

	The missing information for Blank 1 is the command show ip route.
	The missing information for Blank 1 is the command debug ip route.
	The missing information for Blank 2 is the number 100.
	The missing information for Blank 2 is the number 120.
	The missing information for Blank 3 is the letter R.
	The missing information for Blank 3 is the letter C.

	The router has been configured as a firewall.
	The Telnet server process must be activated on the router.
	The switch has not been configured to pass Telnet traffic.
	The enable secret command has not been entered on RtrA.
	The password command has not been entered in line configuration mode on RtrA.
	The password and login commands need to be configured on the console port of RtrA.

	B(config-router)# network 192.168.10.4 0.0.0.3 B(config-router)# network 192.168.10.8 0.0.0.3
	B(config-router)# network 192.168.10.4 0.0.0.3 B(config-router)# network 192.168.10.8 0.0.0.3 B(config-router)#network 192.168.10.128 0.0.0.63
	B(config-router)# network 192.168.10.4 0.0.0.3 area 0 B(config-router)# network 192.168.10.8 0.0.0.3 area 0 B(config-router)#network 192.168.10.128 0.0.0.63 area 0
	B(config-router)# network 192.168.10.4 0.0.0.3 as 100 B(config-router)# network 192.168.10.8 0.0.0.3 as 100 B(config-router)#network 192.168.10.128 0.0.0.63 as 100
	B(config-router)# network 192.168.10.4 0.0.0.3 B(config-router)# network 192.168.10.8 0.0.0.3 B(config-router)# network 192.168.10.64 0.0.0.63 B(config-router)# network 192.168.10.128 0.0.0.63 B(config-router)# network 192.168.10.192 0.0.0.63
	B(config-router)# network 192.168.10.4 0.0.0.3 as 100 B(config-router)# network 192.168.10.8 0.0.0.3 as 100 B(config-router)# network 192.168.10.64 0.0.0.63 as 100 B(config-router)# network 192.168.10.128 0.0.0.63 as 100 B(config-router)# network 192.168.10.192 0.0.0.63 as 100
	B(config-router)# network 192.168.10.4 0.0.0.3 area 100 B(config-router)# network 192.168.10.8 0.0.0.3 area 100 B(config-router)# network 192.168.10.64 0.0.0.63 area 100 B(config-router)# network 192.168.10.128 0.0.0.63 area 100 B(config-router)# network 192.168.10.192 0.0.0.63 area 100

	EIGRP does not support VLSM.
	The routers were not configured to monitor neighbor adjacency changes.
	The default bandwidth was used on the routers.
	An incorrect IP address was configured on a router interface.

	Routers send periodic updates only to neighboring routers.
	Routers send triggered updates in response to a change.
	Routers create a topology of the network by using information from other routers.
	The database information for each router is obtained from the same source.
	Paths are chosen based on the lowest number of hops to the designated router.

	ip summary-address eigrp 1 192.168.10.64 255.255.255.192
	ip area-range eigrp 1 192.168.10.80 255.255.255.224
	summary-address 192.168.10.80 0.0.0.31
	ip summary-address eigrp 1 192.168.10.64 0.0.0.63
	serial interface on router A
	serial interface on router B

	D 192.168.168.0/24 [90/2195456] via 192.168.200.1, 00:00:09, FastEthernet0/0
	O 192.168.168.0/24 [110/1012] via 192.168.200.1, 00:00:22, FastEthernet0/0
	R 192.168.168.0/24 [120/1] via 192.168.200.1, 00:00:17, FastEthernet0/0
	S 192.168.168.0/24 [1/0] via 192.168.200.1

	Router1(config)# interface loopback 1 Router1(config-if)# ip address 192.168.1.1
	Router1(config)# interface serial 0/0 Router1(config-if)# loopback 1 Router1(config-if)# ip address 192.168.1.1
	Router1(config)# interface serial 0/0 Router1(config-if)# loopback 1 Router1(config-if)# ip address 192.168.1.1 255.255.255.0
	Router1(config)# interface loopback 1 Router1(config-if)# ip address 192.168.1.1 255.255.255.255

	ACDP, which has more bits that are used for encryption than CDP has
	WPA2, which can use the more advanced AES for encryption
	VTP, which is compatible with all 802.11-type wireless networks
	RSA, which has a different algorithm than any of the older standards
	802.1X EAP, which is the most powerful type of encryption used for both wired and wireless networks

	show ip route
	show processes
	show running-config
	show cdp neighbor
	show access-lists

	Because the ISP has knowledge of the directly connected 200.0.0.1 network, no route is needed.
	A static route from the ISP is needed that uses the network number 172.16.0.0 and the mask 255.255.0.0.
	A default route from the ISP is needed that uses either the exit interface of S0/0/0 or the 200.0.0.1 IP address.
	A default route from the ISP is needed that uses either the exit interface of S0/0/1 or the 200.0.0.2 IP address.
	A static route from the ISP is needed that uses the network number 209.165.201.8 and the mask 255.255.255.252.