Convergence Technologies SlideShowSIP

Convergence Technologies
Lesson 1: Convergent Network Traffic Protocols
Objectives
Compare and contrast circuit-switched and packetswitched 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:555-1110@ctpcertified.com sip:charles.chaplin@64.128.206.2 sip:charles.chaplin@sip.ctpcertified.com
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 packetswitched 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 analogue 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/withinLAN 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 practises (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 practises 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 analogue telephone adapter (ATA), single line adapter, soft phones (WiFi, PDA, PC-based), analogue 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 a Convergent 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 organisations 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 circuitswitched 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 persons 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 users 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 analogue 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 analogue 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/withinLAN 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 practises (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 practises 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 analogue telephone adapter (ATA), single line adapter, soft phones (WiFi, PDA, PC-based), analogue 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 practise 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, halfduplex, 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-ofservice 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 attach 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 behaviour 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 practise 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 practise 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

Convergence Technologies SlideShowSIP

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Convergence Technologies

Lesson 1: Convergent Network Traffic Protocols

Smart Network and Dumb Network

Circuit-Based vs. Convergence Calling

Transport Through a Packet-Switched Network

User Datagram Protocol

Realtime Transport Protocol (RTP)

Realtime Transport Control Protocol (RTCP)

Session Initiation Protocol (SIP)

SIP Related Protocols

SIP ports and URIs

SIP Messages (cont'd)

H.323 Architecture (cont'd)

H.323 Architecture (cont'd)

H.323 Protocol Stack

H.323 Calls (cont'd)

H.323 Calls (cont'd)

Media Gateway Control Protocol (MGCP)

Network Call Signaling (NCS)

SIP vs. H.323 vs. Megaco

Lesson 2: Implementing VoIP

Planning a Convergent Network

Determining Bandwidth Requirements

Performing a Network Health Check

Creating a Phased Deployment Plan

TAPI and MAPI

Telephone Number Mapping (ENUM)

Comparison of G.7xx Codecs

Calculating VoIP Bandwidth Requirements

Wireless Convergence Technologies

IP Multimedia Subsystem (IMS)

T.120 Multimedia Conferencing Standards

Internet Protocol TV (IPTV)

Common Convergence Devices

Power Issues for Convergent Networks

Lesson 3: Traffic, Troubleshooting and Security

Packet Handling in the Enterprise

Large Data Packets

Other Throughput Considerations

Connection QoS: Using Multiple Connections

Mean Opinion Score (MOS)

Mean Opinion Scores for Popular Audio Codecs

Maintaining and Troubleshooting Convergent Networks

Troubleshooting Convergence in Wireless Networks

Call Detail Records

Security in Convergent Networks

Overview of Network Attacks

Denial-of-Service (DOS) Attacks

Distributed Denial-of-Service (DDOS) Attacks

MAC Address Movements

Maintaining Your Networks

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