Convergence Technologies

Convergence Technologies

Lesson 1:Convergent Network

Traffic Protocols


Objectives

• Compare and contrast circuit-switched and packet-switched technologies, including ways that packets traverse multiple WAN links, and call and call flow descriptions

• Define the Realtime Transport Protocol (RTP) and the Realtime Transport Control Protocol (RTCP)

• Identify the components of Session Initiation Protocol (SIP) and describe the format of an SIP Uniform Resource Identifier (URI)

• Identify the functions of signaling protocols for converged networks (e.g., Session Initiation Protocol [SIP], H.323, H.225, H.320, H.450, Media Gateway Control Protocol [MGCP], Media Gateway Control [Megaco])

• Compare and contrast the functions of gatekeepers, gateways and proxies in relation to SIP and H.323 devices

• Compare and contrast SIP, H.323 and Megaco/MGCP


Defining Convergence

• Convergence – The integration of telephony and data technologies

• Integration includes:– Placing the voice network (telephony), the

video network (television, satellite) and the Internet (rich media) onto common platforms


Smart Network and Dumb Network


Circuit-Based vs.Convergence Calling

• Circuit-switched network – uses a dedicated physical path to send and receive information

• Circuit-based calls:– Provide very good voice quality– May fail if the destination is busy or the network fails

at any point in the connection• Packet-switched network – places addressing information

into data packets• Convergence-based calls:

– Dynamically reroute packets to other network nodes if a network node fails

– Result in increased latency because packetization and compression add processing time to the signal


Transport Through a Packet-Switched Network

• Packets are encapsulated in Ethernet frames– At Layer 4, source and destination port

numbers are added– At Layer 3, source and destination IP

addresses are added– At Layer 2, source and destination MAC

addresses are added


User Datagram Protocol

• UDP header is very simple, consisting of source and destination port numbers, a length field, and a checksum field


Realtime Transport Protocol (RTP)

• Used to transport voice and video payloads for real-time applications

• Provides end-to-end delivery services• Runs over both UDP and TCP• Uses even port numbers that are generally

assigned dynamically• Default port is 5004• RTP profiles define a set of codes for each type

of payload


RTP Packets

• RTP packets are encapsulated in UDP packets


Realtime Transport Control Protocol (RTCP)

• Does not transport any data itself• Partners with Realtime Transport Protocol (RTP)• Monitors the media stream• Provides feedback on the Quality of Service

(QoS) being provided by RTP• While RTP uses an even port number, RTCP

always uses the next odd port number• Default port is 5005


Session Initiation Protocol (SIP)

• Signaling protocol only — does not deliver media streams, nor does it control the delivery of media streams

• Initiates and manages sessions (or connections) between 2 or more participants

• Primary function is to set up, modify and tear down a connection

• Developed by the IETF, SIP is modeled after Hypertext Transfer Protocol (HTTP)


SIP Related Protocols

• Session Description Protocol (SDP)– Describes the characteristics of end points in

a session• Multiprotocol Label Switching (MPLS)

– Can provide QoS for SIP connections• Resource Reservation Protocol (RSVP)

– Can provide QoS for SIP connections• Differentiated Services (DiffServ)

– Can provide QoS for SIP connections


SIP ports and URIs

• SIP uses both UDP and TCP ports 5060 by default

• SIP URI takes the following format:sip:user@host

• SIP URI examples:sip:[email protected]:[email protected]:[email protected]


SIP Components

• User agents– User agent client (UAC): initiates an SIP request– User agent server (UAS): responds to SIP request

• Servers:– Proxy: perform routing, authentication and

accounting functions– Redirect: relays information to a user agent, such

as the IP address of the party to be called– Registrar: enables a client to let a proxy or

redirect server know how the client can be reached


SIP Messages

• Requests– INVITE– ACK– BYE– Cancel– Options– Register

• Each request (except for an ACK request) requires a response


SIP Messages (cont'd)

• Responses are composed of a 3-digit Status Code and an associated Reason Phrase


SIP Calls

Session Invitation• Consists of one

INVITE request, usually sent to an SIP proxy

• A 200 OK response is generated when the called party answers the phone

• Media streams are sent directly between end points


H.323

• Defines the following:– How an audiographic call is set up across a

network– How to negotiate capabilities– How to transmit data and control

conferencing– Which default audio and video codecs to use


H.323 Architecture

• Terminals – H.323 end points– Can be a stand-alone device (IP phone) or a

logical device within a PC– Includes audio and video codecs– Must support H.245 for capabilities

negotiation– Uses Q.931 for call signaling and setup– Uses H.225 RAS for communicating with

gatekeepers– Must support RTP and RTCP


H.323 Architecture (cont'd)

• Gateways– Connect and translate protocols between

dissimilar networks– Provide protocol translation, media format

conversion and data transfer between H.323 and non-H.323 networks

– Optional element; not required for connections within one LAN

– Required to establish connections between terminals in H.323 networks and terminals in networks with different protocols


H.323 Architecture (cont'd)

• Gatekeeper functionality:– Admission control– Address translation– Bandwidth control– Zone management– Call control for point-to-point conferences– Codec translation– Call authorization– Bandwidth and call management– Accounting and billing– Call routing

• Multipoint Control Unit (MCU) – required whenever three or more H.323 terminals are connected


H.323 Protocol Stack


H.225 RAS

• RAS messages (requests and responses) are sent between end points and gatekeepers via UDP– Gatekeeper messages are sent for gatekeeper

discovery (GRQ, GCF, GRJ)– Registration messages are sent for negotiating a

registration with a gatekeeper (RRQ, RCF, RRJ)– Admission messages are requests and replies for

address translation (ARQ, ACF, ARJ)– Status messages are used to monitor end point status

during calls that are routed through a gatekeeper (IRQ, IRR)

– Disengage messages signal the end of a call (DRQ, DCF)


H.323 Calls

• In a typical call:– A client contacts a gatekeeper and requests

an address using H.225 RAS admission request (ARQ)

– Gatekeeper forwards address to the client– Client establishes session using H.225– Session is negotiated using H.245


H.323 Calls (cont'd)

H.225 call signaling is used between terminals to set up and tear down a connection


H.323 Calls (cont'd)

H.245 call control signaling is used for negotiating capabilities and master/slave determination


Media Gateway Control Protocol (MGCP)

• Media Gateway Control Protocol (MGCP) – a signaling protocol used in IP telephony systems– MGCP controls media gateways by sending

signals from a media gateway controller– MGCP is a master/slave protocol– MGCP assumes that call logic and call state

are maintained by intelligent end points


Network Call Signaling (NCS)

• Network Call Signaling (NCS) – a protocol that creates embedded agents to use MGCP in a network


Megaco/H.248

• Enhanced version of MGCP• Result of a joint effort between IETF and ITU• Megaco enables the separation of call control

from media conversion• Megaco instructs an MG to connect streams

coming from outside a packet or cell data network onto a packet or cell stream such as Realtime Transport Protocol (RTP) streams


SIP vs. H.323 vs. Megaco


Summary

Compare and contrast circuit-switched and packet-switched technologies, including ways that packets traverse multiple WAN links, and call and call flow descriptions

Define the Realtime Transport Protocol (RTP) and the Realtime Transport Control Protocol (RTCP)

Identify the components of Session Initiation Protocol (SIP) and describe the format of an SIP Uniform Resource Identifier (URI)

Identify the functions of signaling protocols for converged networks (e.g., Session Initiation Protocol [SIP], H.323, H.225, H.320, H.450, Media Gateway Control Protocol [MGCP], Media Gateway Control [Megaco])

Compare and contrast the functions of gatekeepers, gateways and proxies in relation to SIP and H.323 devices

Lesson 2:Implementing VoIP


Objectives

• List essential steps for qualifying a network's ability to support convergence (e.g., cable inspection, existing and maximum device capacity, replacing hubs with switches, Power over Ethernet [PoE] requirements, VLAN creation, conducting network reconnaissance)

• Describe the features of Telephony Application Programming Interface (TAPI) and Messaging Application Programming Interface (MAPI) in a converged solution

• Implement Telephone Number Mapping (ENUM), elements of global and private numbering plans, Local Number Portability (LNP)/Wireless LNP, end-point addressing, path selection, calling classes, digit manipulation, overlapping number ranges

• Identify common G.7xx codecs and their bandwidth requirements in a converged environment (e.g., G.711, G.729, G.729a, G.726 and others)


Objectives (cont'd)

• Describe the impact of compression on voice quality, and identify issues involved when converting voice to analog and digital formats

• Identify benefits and drawbacks of various codecs in relation to bandwidth and voice quality

• Calculate and estimate bandwidth usage for various codecs, including considerations of overhead, connection quality, and other factors that affect theoretical calculations (e.g., capacity planning, choosing connection speeds)

• Recommend codecs for use with local/in-network/within-LAN calls, and for across WAN connections

• Explain wireless convergence technologies, including Digital Enhanced Cordless Telecommunications (DECT) and DECT layers, Personal Wireless Telephone (PWT), Generic Access Profile (GAP), expected ranges for interference-free communication, and the MHz ranges for each standard


Objectives (cont'd)

• Identify the elements of the IP Multimedia Subsystem (IMS)

• Explain real-time faxing, according to standards such as ITU T.38

• Explain store-and-forward faxing, according to standards such as ITU T.37

• Identify the features, benefits, problems and management of presencing, including single sign-on, features available in various devices

• List unified message methods and benefits (e.g., fax, voice, text, video)

• Identify common and essential videoconferencing codecs, standards and practices (e.g., Moving Picture Experts Group [MPEG], Quarter Common Intermediate Format [QCIF], etc.), and choose the appropriate codecs for various bandwidths


Objectives (cont'd)

• Summarize television/video-calling standards and practices

• Identify multimedia conferencing standards, including all subsets of T.120 (e.g., T.123, T.124, T.135)

• Explain fundamentals of Internet Protocol television (IPTV), including set-top box, Video on Demand (VoD), accepted codecs (e.g., Video Codec [VC-1])

• Identify the purpose and function of voice and videoconferencing hardware (e.g., Multipoint Control Unit [MCU], set-top box, Session Border Controller [SBC])

• Compare and contrast traditional and IP-based private branch exchange (PBX) systems

• Identify convergent terminal equipment and software, including analog telephone adapter (ATA), single line adapter, soft phones (WiFi, PDA, PC-based), analog phones, time division multiplexer (TDM), protocol-specific handsets (e.g., SIP, Megaco)


Objectives (cont'd)

• Explain power issues, including redundancy planning, Power over Ethernet (PoE)/802.3af, PoE classes, expected voltage, wattage, power sourcing equipment (PSE), powered devices (PDs)


Planning aConvergent Network

• Major phases of an implementation plan include these steps:– Identifying expectations– Determining bandwidth requirements– Performing a network health check– Creating a phased deployment plan


Identifying Expectations

• Identify how network(s) will be used• Identify specific protocols that will be used• Identify and explain potential challenges


Determining Bandwidth Requirements

• Identify current digital connection• Determine bandwidth required by existing

network• Monitor current network performance• Evaluate current network performance• Calculate additional requirements for VoIP• Take wide area network (WAN) links into

account• Take growth into account


Performing a Network Health Check

• Check network cabling• Replace hubs with Layer 2 switches• Implement VLANs• Prioritize VLAN traffic• Check routers• Identify the entity that manages Internet router• Examine current IP addressing scheme• Examine Domain Name System (DNS)• Examine firewall• Identify whether NAT will be implemented• Identify whether VPNs must be supported• Identify whether any part of the LAN will be wireless


Creating a Phased Deployment Plan

• Create a detailed, approved implementation plan

• Use a test network• Deploy incrementally• Do not begin with the sales department


TAPI and MAPI

• Telephony Application Programming Interface (TAPI) is an API used for connecting a Windows PC to telephone services

• Messaging Application Programming Interface (MAPI) is a Windows API that allows different e-mail applications to work together to distribute mail


Numbering Plans

• Private numbering plans allow a company to create its own numbering system

• Extensions can be created based on an organization’s needs

• Number plan defines the format of telephone numbers

• Implementing VoIP involves designing a numbering plan and a dial plan.

• Dial plan must include rules for dealing with:– End point addressing– Path selection– Calling classes– Digit manipulation– Overlapping number

ranges


Telephone Number Mapping (ENUM)

• Maps E.164 telephone numbers into the Domain Name System (DNS)

• Creates a dynamic mapping of E.164 addresses to IP addresses

• ENUM domain names are hosted in the e164.arpa domain– A telephone number such as +1 (602) 555-

1212 is converted into the ENUM domain name 2.1.2.1.5.5.5.2.0.6.1.e164.arpa

– ENUM domain name resolves to one or more DNS NAPTR records


G.7xx Codecs

• Various codecs provide different amounts of compression

• Compression allows more voice traffic, but can also:– Introduce delay– Adversely affect voice quality– Put a significant strain on CPU resources,

depending on the complexity of the algorithm and the amount of compression


Comparison of G.7xx Codecs


Calculating VoIP Bandwidth Requirements

• Calculations for bandwidth requirements must factor in:– Codec, sample period and frame size– Frames per packet– IP overhead– Ethernet overhead– Number of simultaneous calls– Silence suppression– Compressed headers


Wireless Convergence Technologies

• Components– Radio exchange– Base stations (transceivers)– Portable phones

• Digital Enhanced Cordless Telecommunications (DECT) is an ETSI standard for digital portable phones– Generic Access Profile (GAP) guarantees

interoperability between any handset and any base station, regardless of make or model

– Operates in the 1880 MHz to 1900 MHz band in Europe, Africa, Australia and Asia (except China)

– Operates in the following bands in North America: 902 MHz to 928 MHz, 2400 MHz to 2483.5 MHz, 5725 MHz to 5850 MHz


IP Multimedia Subsystem (IMS)

• IP Multimedia Subsystem (IMS) is a network architecture designed to enable convergence of voice and data applications and various mobile network technologies

• IMS architecture includes 3 layers:– Connectivity layer (also called the transport layer)

composed of routers, media gateways and switches– Control layer composed of network control servers that

manage call setup, modification and release– Application layer (also called the service layer)

composed of application and content servers that deliver services within the network


Facsimile

• Fax transmissions impose special demands on VoIP because fax standards were designed for circuit-switched connections

• T.30 standardizes the way in which faxes are sent across standard circuit-switched telephone lines

• T.38 designed for real-time fax transmissions over an IP network

• T.37 designed for store-and-forward fax transmission over an IP network


Presencing

• Presence information is a status indicator that conveys a person’s willingness and ability to engage in communications

• Presencing can span different communication channels

• Multiple Points of Presence (MPOP) describes how multiple communications devices can combine state to provide a multidimensional view of a user’s availability status

• Presencing requires collaboration among a number of devices and the presence services with which each of them is connected

• Presencing raises privacy concerns


Unified Messaging

• In unified messaging (UM), all messaging media can come together in the form of a unified mailbox and/or alert service

• Unified messaging offers– Single delivery– Single repository– Single access– Single notification


Video Services

• Video codecs and standards include:– H.261– Common Intermediate Format (CIF)

• Quarter CIF (QCIF)• Sub Quarter CIF (SQCIF)• 4CIF• 16CIF

– H.263– Moving Picture Experts Group (MPEG)

• MPEG-1• MPEG-2/H.262• MPEG-4

– H.264/MPEG-4 Advanced Video Coding (AVC)– Realtime Streaming Protocol (RTSP)


T.120 Multimedia Conferencing Standards

• Key features:– Support for real-time communication between

two or more entities– Support for application sharing, electronic

whiteboarding, file exchange and chat– Support for interoperability between end

points from multiple vendors– Support for a broad range of transport

options– Co-existence with other standards


T.120 Architecture

Additional protocols:• T.128: Multipoint application sharing• T.134: Text chat application entity• T.135: User-to-reservation system transactions with T.120

conferences• T.136: Remote device control application protocol• T.137: Virtual meeting room management services and

protocol


Internet Protocol TV (IPTV)

• Internet Protocol TV (IPTV) can include – Live broadcast (uses Internet Group

Messaging Protocol [IGMP] version 2)– Video on Demand (uses Realtime Streaming

Protocol [RTSP])• Requires a set-top box or PC to receive content

from a media server• Common codecs include: H.264/MPEG-4 AVC,

MPEG-2, VC-1


Common Convergence Devices

• Videoconferencing hardware– Multipoint Control Unit (MCU)– Session Border Controller (SBC)

• IP PBX or traditional PBX• Terminal equipment

– VoIP phones– Digital phones– Analog telephones and adapters – Soft phones– Single line adapter– Time division multiplexer


Power Issues for Convergent Networks

• Redundant power – Uninterruptible power supply (UPS)

• Power over Ethernet (PoE)– Power Sourcing Equipment (PSE) provides power– Powered Devices (PD) use the power provided– Maximum power supplied is 15.4 watts at 48 volts– Five power classes (0-4)– Two power modes (Mode A and Mode B)

• PSE capable of determining the mode a PD uses– Injector supplies power into the appropriate wires of

the Ethernet cable– Deployment of PoE requires a power budget to ensure

that PSE can supply sufficient power to all PDs


Summary

List essential steps for qualifying a network's ability to support convergence (e.g., cable inspection, existing and maximum device capacity, replacing hubs with switches, Power over Ethernet [PoE] requirements, VLAN creation, conducting network reconnaissance

Describe the features of Telephony Application Programming Interface (TAPI) and Messaging Application Programming Interface (MAPI) in a converged solution

Implement Telephone Number Mapping (ENUM), elements of global and private numbering plans, Local Number Portability (LNP)/Wireless LNP, end-point addressing, path selection, calling classes, digit manipulation, overlapping number ranges

Identify common G.7xx codecs and their bandwidth requirements in a converged environment (e.g., G.711, G.729, G.729a, G.726 and others)


Summary (cont'd)

Describe the impact of compression on voice quality, and identify issues involved when converting voice to analog and digital formats

Identify benefits and drawbacks of various codecs in relation to bandwidth and voice quality

Calculate and estimate bandwidth usage for various codecs, including considerations of overhead, connection quality, and other factors that affect theoretical calculations (e.g., capacity planning, choosing connection speeds)

Recommend codecs for use with local/in-network/within-LAN calls, and for across WAN connections

Explain wireless convergence technologies, including Digital Enhanced Cordless Telecommunications (DECT) and DECT layers, Personal Wireless Telephone (PWT), Generic Access Profile (GAP), expected ranges for interference-free communication, and the MHz ranges for each standard


Summary (cont'd)

Identify the elements of the IP Multimedia Subsystem (IMS)

Explain real-time faxing, according to standards such as ITU T.38

Explain store-and-forward faxing, according to standards such as ITU T.37

Identify the features, benefits, problems and management of presencing, including single sign-on, features available in various devices

List unified message methods and benefits (e.g., fax, voice, text, video)

Identify common and essential videoconferencing codecs, standards and practices (e.g., Moving Picture Experts Group [MPEG], Quarter Common Intermediate Format [QCIF], etc.), and choose the appropriate codecs for various bandwidths


Summary (cont'd)

Summarize television/video-calling standards and practices

Identify multimedia conferencing standards, including all subsets of T.120 (e.g., T.123, T.124, T.135)

Explain fundamentals of Internet Protocol television (IPTV), including set-top box, Video on Demand (VoD), accepted codecs (e.g., Video Codec [VC-1])

Identify the purpose and function of voice and videoconferencing hardware (e.g., Multipoint Control Unit [MCU], set-top box, Session Border Controller [SBC])

Compare and contrast traditional and IP-based private branch exchange (PBX) systems

Identify convergent terminal equipment and software, including analog telephone adapter (ATA), single line adapter, soft phones (WiFi, PDA, PC-based), analog phones, time division multiplexer (TDM), protocol-specific handsets (e.g., SIP, Megaco)


Summary (cont'd)

Explain power issues, including redundancy planning, Power over Ethernet (PoE)/802.3af, PoE classes, expected voltage, wattage, power sourcing equipment (PSE), powered devices (PDs)

Lesson 3:Traffic, Troubleshooting

and Security


Objectives

• Define latency, jitter and wander • Implement methods for reducing or eliminating latency,

jitter and wander (e.g., implementing a jitter buffer, implementing QoS, traffic shaping, VLANs)

• Explain the impact of large frames on real-time communications

• Identify factors that affect the bandwidth of voice and video calls on convergent networks (e.g., latency, protocol incompatibility, MTU, codec choice, compression, QoS issues, packet reordering, loss of feature set)

• Use accepted industry standards such as the Mean Opinion Score (MOS) to determine voice and video quality, including MOS for popular codecs, standard MOS numbers, R-value and subjective video quality


Objectives (cont'd)

• Identify common network bottlenecks in convergent networks, including solutions (e.g., monitoring network devices and protocols, creating a baseline, changing configuration, upgrading hardware)

• Analyze traffic in a convergent network and resolve problems using a packet sniffer, monitoring software, and hardware solutions

• Troubleshoot convergent communications over wireless networks

• Identify problems in contacting emergency services through convergent networks

• Parse a Call Detail Record (CDR) and list relevant entries


Objectives (cont'd)

• Identify types and effects of attacks in convergent networks, including man-in-the-middle attacks (e.g., packet sniffing, TCP connection hijacking, registration hijacking), voice mail compromises, viruses, brute-force and dictionary attacks, zero-day attacks, illicit servers, toll fraud and unsolicited calls

• Define denial-of-service (DOS) and distributed DOS (DDOS) attacks, and identify ways to counteract them, including common traffic types used (e.g., SYN, UDP or ICMP flood), reconfiguring core upstream routers, using alternative sites, intentional and unintentional DOS

• Explain the practice and impact of VLAN hopping • Explain the significance and impact of MAC address

movements, additions and changes • Identify types of intrusion detection (e.g., host-based,

network-based, defining effective signatures, proactive detection)


Objectives (cont'd)

• Back up, upgrade and scan systems to thwart attacks, including backup types, system patches, service packs, firmware upgrades, optimal backup schedule


VoIP Variables

• VoIP variables – conditions that cause problems in voice communications

• VoIP variables include:– Delay – the amount of wait time between the

time a signal is sent and received– Latency – the amount of time required for

data to be transmitted across a network– Jitter – variability in the arrival rate of data

packets transmitted over a network– Wander – variability of more than one second

in the arrival rate of data packets transmitted over a network (long-term jitter)


Delay• Fixed delays

– Propagation delay – caused by the distance between the request and the server fulfilling the request

– Serialization delay – the time required to physically place voice call bits on a trunk line

– End point processing delay – caused by compressing/ decompressing and encoding/decoding data

– Packetization delay – the time required to place digital traffic into a particular medium

• Variable delays– Queuing delay – the time packets wait for other

packets to be placed onto a trunk line– Router processing delay – the time required for a

router to apply QoS settings, or to process packets that have arrived out of order


Latency

• Latency results when multiple delays occur• The most significant source of latency is the

digital signal processing that occurs in gateways and routers

• Round-trip latency is the total delay experienced by two users on a phone call

• Round-trip latency in the PSTN is typically less than 150 milliseconds, except on international calls

• ITU recommends that for good voice quality in VoIP calls, one-way latency must not exceed 150 milliseconds


Jitter

• Jitter occurs when packets in a voice transmission take different paths over a network, causing them to arrive out of sequence

• A jitter buffer can correct this variability by providing a space in memory that allows packet resequencing


Packet Handling in the Enterprise

• A chief cause of delay and jitter is queuing at the router

• By default, routers process queues on a first in first out (FIFO) basis

• Implementing QoS on routers improves voice quality

• Convergent QoS technologies include:– Creating VLANS– Assigning prioritization– Setting IP precedence values– Employing traffic-shaping algorithms


Wander

• Wander is due to synchronization problems in the network clocks used to control transmissions

• When wander is detected, the signal must be reclocked, or synchronized, at the next network element to avoid propagating the wander activity

• The Network Time Protocol (NTP) ensures that systems are accurate to within milliseconds

• NTP servers belong to two strata:– Stratum 1 – clocks that are the most accurate;

often GPS-enabled timekeeping systems– Stratum 3/3E – VoIP gatekeepers, gateways

and PBXs


Large Data Packets

• Voice packets can get “stuck” behind large data packets and incur significant delay

• Segmenting large data packets on the network can help control latency and jitter of voice services


Other Throughput Considerations

• Additional factors that affect the quality of voice transmissions include:– Choice of codec– Complexity of compression algorithm– Lack of QoS support on the network– Overutilization of routers– Packet reordering (caused by congestion and

queuing at routers)– Protocol and codec incompatibility– MTU setting– Loss of feature set


Connection QoS: Using Multiple Connections

• Connection QoS ensures that the gateway can protect calls from network problems in several ways, including:– Trunk busy-out– Alternative gateway selection– Fallback to the PSTN

• The gateway prevents a trunk from servicing a call if: – The IP network fails– The gateway detects an internal problem


Mean Opinion Score (MOS)

MOS is an industry standard numerical measurement of voice quality


R-value

• R-value is another industry standard for measuring voice quality

• R-values are derived from direct measurements of equipment and traffic parameters

• R-value score ranges from 1 (worst) to 100 (best)

• One MOS point is roughly equal to 20 R-value points, but the correlation is not linear


Mean Opinion Scores for Popular Audio Codecs


Maintaining and Troubleshooting Convergent Networks

• Monitoring is an important aspect of maintaining convergent networks

• The first step in monitoring is establishing a baseline– A baseline is a record of normal network

activity that serves as an example for comparing future network activity


Establishing a Baseline

• Baseline measurement statistics should include data on:– Traffic analysis/end-to-end performance

• Identifies latency, percentage of packet loss and link utilization

• Tools include: ping and traceroute, and hardware monitoring mode

– Device performance• Identifies factors such as CPU and memory

usage


Device Configuration

• Device configuration directly affects the performance of convergent networks

• Check configurations of – Switches

• Ensure VLANs are properly configured• Ensure proper communication mode (full-duplex,

half-duplex, auto-negotiation) settings on switch ports

– End points• Ensure each end point has a valid IP address• Ensure that communication mode on NIC is set

properly • Install firmware or software updates as they

become available


Troubleshooting Convergence in Wireless Networks

• In wireless networks, points for troubleshooting include:– Access points (APs):

• Should support enterprise-level QoS• Should provide overlapping coverage• Should be deployed in sufficient number• Should be able to reject calls when becoming

overloaded• Should support roaming

– Environment• Check for sources of interference

– Handsets• Ensure proper configurations (for example,

encryption keys)• Install available updates/upgrades


Call Detail Records

• Call Detail Records include information about the the following call details:– Time– Date– Call duration– Number dialed– Caller ID information– Extension– Line/trunk location– Cost– Call completion status


Security in Convergent Networks

• Security is a set of procedures designed to protect transmitted and stored information, as well as network resources

• In convergent networks, security includes preventing:– Call interception– Phone fraud– Network attacks


Protocol Review

• Inherent weaknesses in IPv4 include:– Transmission Control Protocol (TCP) handshake

– often manipulated by hackers– Internet Protocol (IP) – does not sign or encrypt

packets, and packets are easily manipulated– User Datagram Protocol (UDP) – often used to

conduct scans of systems, and UDP packets can be forged to wage distributed denial-of-service attacks

– Address Resolution Protocol (ARP) – does not authenticate the hosts it resolves and is subject to ARP cache poisoning


Overview of Network Attacks

• Network attacks include:– Spoofing attacks

• IP spoofing• ARP spoofing• DNS spoofing

– Man-in-the-middle (hijacking) attacks• Password sniffing• Connection termination• Connection hijacking• Packet insertion• Poisoning

– Password-guessing attacks• Brute-force attacks• Dictionary attacks


Malicious Code

• Types of malicious code include:– Viruses– Worms– Illicit servers– Trojan horses

• To avoid malicious code, use:– Virus and worm protection– Application management and testing– Configuration management– File signature checking software


Denial-of-Service (DOS) Attacks

• Purpose of a denial-of-service attack is to:– Crash a server and make it unusable to

everyone– Assume the identity of the system being

crashed– Install a Trojan

• Flooding is the process of sending an overwhelming number of packets to a system

• Flooding techniques include:– SYN flood– Ping flood– UDP flood


Distributed Denial-of-Service (DDOS) Attacks

• Involve the cooperation of several systems to wage a coordinated attack that generates an overwhelming amount of network traffic

• DDOS attacks involve: – A controlling application– An illicit service– A zombie– A target

• DOS and DDOS attacks can be diagnosed by:– Using a packet sniffer to view traffic– Using the netstat command to view connections– Using intrusion-detection systems


VLAN Hopping

• VLAN hopping is an attack in which a hacker intercepts packets as they are sent from one VLAN to another on a trunk

• To avoid VLAN hopping:– Disable autotrunking– Remove the native VLAN setting (VLAN 1)

from any trunk port


MAC Address Movements

• Sudden changes in MAC addresses, such as two systems suddenly exchanging IP addresses, can indicate that someone is attempting to poison the ARP cache

• Monitoring the ARP cache with a tool such as Arpwatch can guard against ARP spoofing


Intrusion Detection

• Intrusion detection strategies rely on:– Signature detection– Anomaly detection (less common)

• IDS applications require a current signature database• IDS application types are:

– Host-based• Captures traffic only on host, not on the network

wire– Network-based

• Does not capture traffic on switched networks• Port mirroring enables captures and monitoring on

switched networks


Maintaining Your Networks

• Essential tasks for maintaining a convergent network include:– Scan systems regularly to detect unusual behavior– Upgrade equipment as necessary– Install system patches and service packs– Keep antivirus files current– Install firmware upgrades– Perform regular backups

• Full• Differential• Incremental

– Verify that backups are successful– Ensure careful (off-site) storage of backup media– Choose an optimal backup schedule


Summary

Define latency, jitter and wander Implement methods for reducing or eliminating latency,

jitter and wander (e.g., implementing a jitter buffer, implementing QoS, traffic shaping, VLANs)

Explain the impact of large frames on real-time communications

Identify factors that affect the bandwidth of voice and video calls on convergent networks (e.g., latency, protocol incompatibility, MTU, codec choice, compression, QoS issues, packet reordering, loss of feature set)

Use accepted industry standards such as the Mean Opinion Score (MOS) to determine voice and video quality, including MOS for popular codecs, standard MOS numbers, R-value and subjective video quality


Summary (cont'd)

Identify common network bottlenecks in convergent networks, including solutions (e.g., monitoring network devices and protocols, creating a baseline, changing configuration, upgrading hardware)

Analyze traffic in a convergent network and resolve problems using a packet sniffer, monitoring software, and hardware solutions

Troubleshoot convergent communications over wireless networks

Identify problems in contacting emergency services through convergent networks

Parse a Call Detail Record (CDR) and list relevant entries


Summary (cont'd)

Identify types and effects of attacks in convergent networks, including man-in-the-middle attacks (e.g., packet sniffing, TCP connection hijacking, registration hijacking), voice mail compromises, viruses, brute-force and dictionary attacks, zero-day attacks, illicit servers, toll fraud and unsolicited calls

Define denial-of-service (DOS) and distributed DOS (DDOS) attacks, and identify ways to counteract them, including common traffic types used (e.g., SYN, UDP or ICMP flood), reconfiguring core upstream routers, using alternative sites, intentional and unintentional DOS

Explain the practice and impact of VLAN hopping Explain the significance and impact of MAC address

movements, additions and changes


Summary (cont'd)

Identify types of intrusion detection (e.g., host-based, network-based, defining effective signatures, proactive detection)

Explain the practice and impact of VLAN hopping Back up, upgrade and scan systems to thwart attacks,

including backup types, system patches, service packs, firmware upgrades, optimal backup schedule



Convergent Network Traffic Protocols Implementing VoIP Traffic, Troubleshooting and Security

Documents

Convergence Technologies