Exam Code: 920-803

Course Name: Technology Standards and Protocols for IP Telephony Solutions Certification Exam

Vendor: Nortel

Passing Score: 66% of Total Marks

Technology Standards and Protocols for IP Telephony Solutions Certification Exam, also known as 920-803 exam, is a Nortel certification. IT Professionals who wish to work as public network carriers, wireless operators, and multi-service operators can do that by taking 920-803 exam. Studying and taking the Nortel 920-803 exam can be for anyone in the IT field who wants to excel in IT field.

The 920-803 exam provides you with the technical definition of how VoIP traffic flows between gateways, gatekeepers, clients, and call servers for both call setup and media path. It enables you to identify the key infrastructure requirements needed to support the addition of VoIP traffic in a LAN/WAN environment. In order to be able to compare VoIP versus VoFR and VoATM as voice over packet alternatives to define the features and benefits of each transport model, you must take the 920-803 exam. Moreover, it facilitates you with the abilities to implement knowledge of how voice is sampled and converted into IP packets to determine the appropriate CODEC and packetization interval required to meet customer VoIP bandwidth and voice quality requirements, as well as, to maintain voice quality in a frame. This Nortel certification helps you identify the traffic convergence issues. In addition, the 920-803 exam assists you in assessing the data network for VoIP.

The internet is crowded with thousands of websites about various topics including exam sites as well. These exam sites prove to be very helpful for the candidates who want to be certified in any of the fields related to computer. Among all these websites, Testking is the website that guides you all through your 920-803 Exam for Technology Standards and Protocols for IP Telephony Solutions Certification Exam. Testking proves to be very helpful for people who are wholly solely dependent on the study materials available on the internet for 920-803 Exam for Technology Standards and Protocols for IP Telephony Solutions Certification Exam.

Testking offers you with all the important information material required for taking 920-803 Exam for Technology Standards and Protocols for IP Telephony Solutions Certification Exam. Testking helps you learn fast all the essential features that a candidate must know about the 920-803 Exam for Technology Standards and Protocols for IP Telephony Solutions Certification Exam. Each TestKing certification products provides you with free updates for 90 days.

C ourse Outline: Technology Standards and Protocols for IP Telephony Solutions Certification Exam

Major Components of VoIP

In an IP network, define how VoIP fraffic flows between gateways, gatekeepers, clients, and call servers for both call setup and media path.

Media Gateways

H.323 Gatekeep functions

IP Terminals and Clients

Hard or soft telephones (portable or stationary)

Wireless devices (802.11a and 802.11b)

Call Servers

Proxy Servers

Management and converged software applications

Network Infrastructure (LAN & WAN)

Define the key infrastructure requirements needed to support the addition of VoIP traffic in a LAN/WAN environment.

IP network call setup traffic flow for H.323 wireless devices in an 802.22b wireless infrastructure

Enhancing voice performance in a VoIP network by positioning of Layer 2 and Layer 3 switches

Appropriate use of 802.1p Class of Service or Differentiated Services (DiffServ), based on switch type

Advantages of DiffServ in providing better control for mapping applications with different types of Quality of Service (QoS) behavior

Advantages of routers and switches prioritizing VoIP packets ahead of the other IP data packets

Transport Models

Compare VoIP versus VoFR and VoATM as voice over packet alternatives to define the features and benefits of each transport model, specifically how the Committed information Rate (CIR) impacts voice and the role of queuing methods.

VoIP

Layer 3 LAN/WAN protocol

Variable packet sizes

VoFR

Layer 2 WAN protocol

Variable frame sizes

VoATM

Layer 2 LAN/WAN protocol

Fixed cell sizes

Compression Standards

Compare the standard speech CODECs to define their features and benefits.

G.723.1 characteristics

G.726 characteristics

G.729A characteristics

G.107E characteristics

G.711 characteristics

Transport and Session Layer Internet Protocols

Apply knowledge of the attributes of Real-Time Protocol (RTP) to identify why it is ideal for handling packetized voice in an IP telephony environment.

Define the unique attributes of User Datagram Protocol (UPD) and Transmission Control Protocol (TCP), and explain the benefits of UPD over TCP in a real-time VoIP environment.

RTP characteristics

Real-Time Control Protocol (RTCP) characteristics

Interaction between RTP and RTCP

CODEC and Voice Packetization Selection

Apply knowledge of how voice is sampled and converted into IP packets to determine the appropriate CODEC and packetization interval required to meet customer VoIP bandwidth and voice quality requirements.

Requirements for ensuring QoS for voice quality maintenance between two points on a Frame Relay network

Identify devices that shape traffic

Appropriate use of Committed Information Rate (CIR)

Identification of OSI Layers for key functions, such as IP, ATM and Frame Relay

ATM transport requirements when transporting an IP voice packet over an ATM network

Performance Considerations

Explain how Ip network impairments (e.g., packet loss, delay, echo, etc) impact voice quality.

Given a VoIP solution with specified bandwidth requirements and multiple CODECs in the speech path, determine whether latency from multiple transcodings will result in acceptable voice quality.

Voice Quality Measurement Models

Compate the G107 E Model to the Mean Opinion Score (MOS) to define the features and benefits.

Quality of Service (QoS) Methods

Given customer voice quality requirements for a VoIP solution, choose the appropriate QoS method, e.g., RSVP, DiffServ, 802.1q/p, port-based prioritization, etc) to achieve the best voice quality.

QoS methods for achieving best voice quality in a Layer 2 and a Layer 3 technology environment

Resource ReSerVation Protocol (RSVP) characteristics and when used

Differentiated Services (DiffServ) characteristics and when used

802 Standards

Port-based Prioritization

Traffic Separation using VLANs

IP Address Prioritization

Packet Fragmentation - implementation on WAN circuits of less than 1Mbps

IP Fragmentation

Traffic Shaping

Prioritizing VoIP traffic and signaling with Per-Hop-Behaviors (PHB)

Assured Forwarding

Expedited Forwarding

Queuing Mechanisms

Identify the queuing mechanisms (e.g., First In First Out (FIFO), Priority Queuing (PQ), etc) available to handle network traffic (Layers 2 and 3 ) congestion.

Queuing definitions

Fair queuing

Priority queuing

Common Ethernet Network Issues for LAN Environments

Define the common Ethernet network issues known to be problematic for VoIP traffic in a LAN environment.

Configuration parameters that may affect VoIP traffic

Round Trip Delay (RTD)

Excessive LAN collisions

Methods for improving delay characteristics

Challenges of Low-Speed WAN Connections

Explain the issues and challenges of running VoIP over low-speed WAN connections.

Maximum Transmission Unit (MTU) setting implications

H.323 - International Telecommunications Union (ITU) Standard

Identify the components of the H.323 standard in a VoIP environment.

H.323 terminal characteristics

H.323 gateway characteristics

H.323 gatekeeper characteristics

H.323 interoperability issues

Session Initiation Protocol (SIP) - Internet Engineering Task Force (IETF) Standard

Identify how the SIP standard enables support of multimedia "sessions" for IP telephony.

Compare SIP versus H.323 within a VoIP environment to determine their respective benefits and capabilities.

SIP characteristics

How SIP supports multimedia applications

How SIP can be used to support multimedia sessions for IP telephony

How SIP uses session management

Session Description Protocol (SDP) characteristics

Use of network server types to provide network intelligence and services

Other Signaling Protocols

Identify the Media Gateway Control Protocol (MGCP) and Megaco/H.248 standards and their applicability in controlling VoIP media gateways.

Usage while SIP is being used in the same network

Ensuring interoperability with equipment that uses Megaco signaling

Use of a stimulus-based signaling model for obtaining intelligence and features from servers

Network Health Assessment (Conceptual)

Define the recommended VoIP network health assessment process and areas (e.g., link types/speeds, peak delay, packet loss, LAN/WAN platforms, etc) used to assess customer LAN networks for successful implementations of VoIP solutions.

VoIP Network Health pre-assessment questions

Network assessment steps

Develop customer recommendations for network improvements, based on sample network assessment scenarios in preparation for LAN VoIP deployments.

Critical factors in determining "consistent" voice quality

Dynamic Host Configuration Protocol (DHCP) server characteristics<il> -- VoIP Network path redundancy facilitation

Dynamic routing protocols

Multiple call servers

Reducing security risks in a Network Address Translation (NAT) and firewall environment

Network Health Assessment Tools

Discriminate the VoIP network health assessment tools (e.g., Sniffer Pro, Net IQ Chariot, IXIA Qcheck, NetIQ Vivinet Assessor, Viola Networks NetAlly VoIP, Multi-Router Traffic Grapher, etc) available to perform VoIP network health assessments.

NetIQ Chariot

Sniffer Pro

Multi-Router Traffic Grapher