Cisco®CCDA Exam Cram Notes : Considerations For Wireless Network Architectures


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4. Considerations for Expanding an Existing Network

4.1 Describe design considerations for wireless network architectures

Various methods are used in wireless controller redundancy design.

wireless network architectures

The above network diagram shows N+N+1 method

N+1 Wireless Controller Redundancy Design: A single WLC acts as the backup for multiple WLCs. The backup WLC is configured as the secondary on Aps.

N+N Wireless Controller Redundancy Design: An equal number of controllers back up each other.

N+N+1 Wireless Controller Redundancy Design: An equal number of controllers back up each other. The backup WLC is configured as the tertiary on Aps.

In a deterministic controller design, planned redundancy configurations such as N+N and N+N+1 provide for logical grouping of access points on controllers to minimize inter controller roaming.

The following are the recommended WLAN design practice

1. Cisco recommends 20 or less data devices per AP and 7 G.711 concurrent or 8 G.729 concurrent VoWLAN calls.

2. It is recommended to place APs in centralized locations such as conference rooms to accommodate peak traffic requirements.

3. It is recommended to use Power over Ethernet (PoE) to power APs and provide wired access.

4. Recommended practice is to place WLCs on secured wiring closets or in the data center. Deterministic redundancy is recommended, and inter controller roaming should be minimized.

Identify high-level considerations for collaboration (voice, streaming video, interactive video) applications

In voice communication, quantisation and coding process converts the analog value into a digital value. Assuming a standard word size is 8 bits, the following correctly represents the bandwidth requirement of 64kbps for voice transmission in digital form

The rate then becomes the sampling rate times the size of the code word (2 * 4 kHz * 8 bits = 64 kbps)." to "The rate then becomes the sampling rate times the size of the codeword (1 * 8 kHz * 8 bits = 64 kbps)

Dense Wavelength Division Multiplexing (DWDM) is an optical multiplexing technology used to increase bandwidth over existing fiber networks. DWDM works by combining and transmitting multiple signals simultaneously at different wavelengths on the same fiber. The technology creates multiple virtual fibers, thus multiplying the capacity of the physical medium. DWDM introduces a concept called fiber channel. Each fiber channel is the equivalent to several (Nx) Gigabit Ethernet links.

Note that DWDM uses signals at different wavelengths multiplexed on a single mode fiber. Multi mode fiber is not used because of severe attenuation. DWDM lets a variety of devices access the network, including IP routers, ATM switches, and SONET terminals.

Link Utilization: An organization using WAN links needs to monitor link utilization on a continual basis to avoid any future business interruptions due to slow WAN links. It takes long times (say, a month to 6 months) to plan and purchase additional bandwidths and network upgrades. Therefore, it is better to plan early. One should start monitoring a WAN link when the link utilization exceeds 50%, and plan for procurement of additional bandwidth when it exceeds 75%

Cisco Unified Communications Manager (CUCM) is an IP-based communications system integrating voice, video, data, and mobility products and applications. It provides services such as session management, call processing for voice and video, messaging, mobility, and web conferencing.

CUCM provides the following functions:

1. Call processing: Call processing refers to the complete process of originating, routing, and terminating calls, including any billing and statistical collection processes.

2. Signaling and device control: CUCM sets up all the signaling connections between call endpoints and directs devices such as phones, gateways, and conference bridges to establish and tear down streaming connections. Signaling is also referred to as call control and call setup/call teardown.

3. Dial plan administration: The dial plan is a set of configurable lists that CUCM uses to perform call routing. CUCM is responsible for digit analysis of all calls. CUCM enables users to create scalable dial plans.

4. Phone feature administration: CUCM extends services such as hold, transfer, forward, conference, speed dial, redial, call park, and many other features to IP phones and gateways.

5. Directory services: CUCM uses its own database to store user information. User authentication is performed locally or against an external directory. Directory synchronization allows for centralized user management. Directory synchronization allows CUCM to leverage users already configured in a corporate-wide directory. Microsoft Active Directory (2000 and 2003), Netscape 4.x, iPlanet 5.1, and Sun ONE 5.2 directory integrations are supported. The local CUCM database is a Lightweight Directory Access Protocol (LDAP)-compliant database (LDAPv3) component in the IBM Informix Database Server (IDS).

6. Programming interface to external applications: CUCM provides a programming interface to external applications such as Cisco IP SoftPhone, Cisco IP Communicator, Cisco Unified IP Interactive Voice Response (IP IVR), Cisco Personal Assistant, Cisco Unified Personal Communicator, and CUCM Attendant Console.

7. Backup and restore tools: CUCM provides a Disaster Recovery System (DRS) to back up and restore the CUCM configuration database. The DRS system also backs up call details records (CDR), call management records (CMR), and the CDR Analysis and Reporting (CAR) database.

1. CAC is configured for the site on the CUCM servers and ensures call quality by controlling the number of calls between two locations.

2. CAC causes excessive calls between two locations to be denied.

3. Cisco recommends that IPT voice packets should be marked with a DSCP of EF (IP precedence 5), and signalling packets should be marked with AF31 (IP precedence 3).

4. 802.1P to allow for prioritisation at Layer 2

Q.931 is a standard for call signalling used by H.323 within the context of H.225. H.245 specifies messages for opening and closing channels for media streams and other commands, requests, and indications. It is a control channel protocol.

H.323 is a standard published by the ITU that works as a framework document for multimedia protocols, including voice, video, and data conferencing, for use over packet switched networks.

MGCP is a client/server signalling protocol that is used to allow centralized call processing agents (such as CUCM) to control gateways in VoIP networks. MGCP is defined in RFC 3661.

In VoIP, RTP transports audio streams. RTP is a transport layer protocol that carries digitised voice in its payload. RTP was initially defined in RFC 1889 and the current RFC is 3550. RTP runs over UDP

SCCP is a Cisco proprietary client/server signalling protocol for call set-up and control. SCCP runs over TCP. SCCP is called a "skinny" protocol because it uses less overhead than the call set-up protocols used by H.323. IP phones typically use SCCP to register with CUCM and to establish calls.

RTP: Transports coded voice streams

H.323 standard defines the following devices:

Terminals : TCP/IP endpoint devices. Telephones, video phones, and voice-mail devices that provide real-time two-way transmission.

MCUs : Extends 2-way communications to 3 or more endpoints, comprised of MC and MP.

Gateways :Provides 2 way communications between terminal/MCU and other devices, such as H.320 terminals/MCU and telephones.

Gatekeepers : Provides address translations, access control, and optionally bandwidth management.

A device that provides transitional services from one network type to another such as connecting a VoIP network to a TDM network such as the PSTN (analog/T1 PRI).

H.323 terminals must support the following standards:

1. H.245 call capability control

2. Q.931 call set-up signalling

3. H.225 call signalling

4. RTP/RTCP voice streams

H.245 specifies messages for opening and closing channels for media streams and other commands, requests, and indications. It is a control channel protocol.

Q.931 is a standard for call signalling used by H.323 within the context of H.225. It is also used by PRI links.

H.225 performs registration, admission, and status (RAS) signalling for H.323 sessions.

RTP is the transport layer protocol used to transport VoIP packets. RTCP is also a transport layer protocol.

SIP uses a modular architecture that includes the following components:

1. SIP user agent (UA): These endpoints create and terminate sessions, SIP phones, SIP PC clients, or gateways. A UA client (UAC) initiates a SIP request.

2. SIP proxy server: Routes messages between SIP UAs. It acts as an intermediate that receives IP requests from a client and forwards the requests on behalf of the client. SIP proxy servers perform authentication, authorization, routing, reliable request retransmission, and security.

3. SIP redirect server: Call control device used to provide routing information to user agents. It provides information about the next hop or hops that a message should take.

4. SIP registrar server: Stores the location of all user agents in the domain. It processes requests from UACs for registration of their current locations. SIP proxy servers or redirect servers can contain registrar servers.

5. SIP location services: Provide logical location of UAs; used by the proxy, redirect, and registrar servers.

6. Back-to-back user agent: Call control device that divides a voice call into two call legs.

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