copyright © 2006 prosoft learning, a vcampus company - all rights reserved. convergence...

Copyright © 2006 Prosoft Learning, a VCampus Company - All rights reserved.

Convergence Technologies


Lesson 1:Convergence Industry

Standards and Protocols

Objectives

• Discuss the various standards agencies in the telecommunications industry

• Discuss the major industry standards in convergence technologies

• Identify and define the various IEEE 802 protocols• Identify and define the various ITU protocols• Discuss Requests for Comments (RFCs) used in

convergence technologies

Defining Convergence

• Convergence – The integration of telephony and data technologies

• Benefits of convergence:– Deliver services to any IP-enabled device– Allow for a more flexible use of data– Enable toll-free calling– Handle voice calls for multimedia

communication– Enable the use of PCs as phones– Use existing infrastructure– Lower ownership costs

Common VoIP Applications

• Toll bypass• Fax over the Internet• PC phone to PC phone• IP-based public phone service• Call-center IP telephony• IP local line doubling

Common VoIP Protocols

• Session Initiation Protocol (SIP)– Initiates and manages sessions between two or more

participants– Defines how end devices create, modify and terminate a

connection• H.323

– Manages setup, tear down and call control– Defines components in a conferencing network

• MGCP and Megaco/H.248– A standard protocol for handling signaling and session

management during a multimedia conference or VoIP call• Megaco is the IETF name for MGCP• H.248 is the ITU-T name for MGCP

Governing Organizations in Convergence Technologies

• IEEE – a nonprofit association based in the U.S. concerned with data communication standards

• ITU – an agency within the UN that coordinates the establishment and operation of global telecommunication networks and services

• IETF – an international community of operators, vendors, network designers and researchers concerned with the evolution of Internet architecture and the operation of the Internet

• EIA – a U.S. electronics manufacturer organization that has published a number of standards related to telecommunication and computer communication

• TIA – vendors, service providers and organizations involved in all aspects of modern communication networks

• ANSI – an organization that defines coding standards and signaling schemes in the U.S.

• Telcordia – provides engineering, administrative, software and telecommunications consulting services to telecommunications companies

Institute of Electrical and Electronics Engineers (IEEE)

– 802.1 – Internetwork– 802.2 – Logical Link

Control (LLC)– 802.3 – CSMA/CD and

Ethernet– 802.5 – Token Ring

Networks

– 802.6 – Metropolitan Area Networks

• Switched Multimegabit Data Service (SMDS)

– 802.9 – Integrated Data and Voice Networks

– 802.11 – Wireless LANs– 802.12 – 100VG-AnyLAN– 802.14 – Cable Modem

• IEEE 802 protocols – define the relationships between physical network interfaces and all signaling

InternationalTelecommunication Union (ITU)

• ITU H-Series protocols – define the structure and use of protocols for audiovisual and multimedia systems– H.225– H.235– H.245– H.248 (Megaco)– H.261– H.263– H.323– H.450


(cont'd)

• ITU G-Series protocols – identify the data rate of VoIP connections in the network– G.711 – Toll Quality– G.723.1– G.726– G.728– G.729


(cont'd)

• ITU T-Series protocols – discuss different types of terminals for telephony services– T.30– T.37– T.38– T.134


(cont'd)

• ITU Q-Series protocols – address issues of switching and signaling– Q.931

• ITU X-Series protocols – address issues related to data networks and open system communication– X.200– X.300

• Global System for Mobile Communications (GSM)– Wireless network system used primarily in

Europe

VoIP and Interoperability

• International Multimedia Teleconferencing Consortium (IMTC) ensures that VoIP gateways and PC-based phones follow the same standards

• VoIP Forum – subgroup of the IMTC– Created the VoIP Interoperability Implementation

Agreemento A precursor to a standard that has formed the

basis of subsequent standardso Widely supported in the industry

Internet Engineering Task Force (IETF)

• IETF Requests for Comments (RFCs)– RFC 3261 – Session Initiation Protocol (SIP)– RFC 3550 – RTP: A Transport Protocol for Real Time

Applications– RFC 2205 – Resource Reservation Protocol

• RFC 2750• RFC 3936

– RFC 3802 – Toll-Quality Voice 32 Kbps ADPCM Registration

– RFC 2805 – Media Gateway Control Protocol Architecture and Requirements

– RFC 3494 – Lightweight Directory Access Protocol– RFCs 2236 and 3376 – Internet Group Management

Protocol (IGMP)

ElectronicIndustries Alliance (EIA)

EIA TIA

Commercial Building Telecommunications Wiring Standards

Telecommunications Industry Association (TIA)

• TIA/EIA standards– TIA/EIA – 810-A Telecommunications– TIA/EIA/IS – 811 Telecommunications

American NationalStandards Institute (ANSI)

• ANSI standards– T1.240 – Generic Network Information Model for

Interfaces Between Operation Systems and Network Elements

– T1.520 – Internet Protocol (IP) Data Communication Service – IP Packet Transfer and Availability Performance Parameters

Telcordia(Formerly Bellcore)

• Telcordia Generic Requirements standards– GR-301 – Public Packet Switched Network– GR-303 – Integrated Digital Loop Carrier System– GR-1504 – Wireless Service Provider Automatic

Message Accounting– GR-3058 – Voice over Packet: Next Generation

Networks (NGN) Accounting Management– GR-2804 – Universal Network to Server Access

Method (UNAM)– Radio Free Ethernet (RFE)

Summary

Discuss the various standards agencies in the telecommunications industry

Discuss the major industry standards in convergence technologies

Identify and define the various IEEE 802 protocols Identify and define the various ITU protocols Discuss Requests for Comments (RFCs) used in

convergence technologies


Lesson 2:Enabling Voice over IP (VoIP)

Objectives

• Discuss the functions of gatekeepers• Discuss the functions of gateways• Define delay, latency, jitter and wander, and

identify their impact on real-time communications• Identify the importance of a jitter buffer• Identify the impact of large data frames on

real-time communications• Recognize the need for Quality of Service (QoS) for

converged networks• Identify QoS technologies for converged networks

Objectives (cont'd)

• Identify common codecs and their bandwidth requirements in a converged environment

• Describe the impact of compressing voice in a network

• Compare and contrast the use of T1, E1 and J1 trunks for data and voice

• Identify the factors that affect the bandwidth of packetized voice

• Identify requirements for transporting modem and fax transmissions through a converged solution

Investigating VoIP

• Asynchronous transfer mode (ATM) – a connection-oriented technology that supports real-time video, voice and data for LANs and WANs

• Frame relay – a packet-switching technology that supports data and voice for LANs and WANs

• Comparing VoIP and standard PSTN connections

Investigating Gatekeepers and Gateways

• 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

Gateway Functionality and Types

• Gateways connect two different networks• Gateway types:

– Signaling gateway – translates call control and administrative signals present on the circuit-switched PSTN into either SIP or H.323

– Media gateway – packetizes telephony information for transmission across the Internet or an intranet

VoIP gateways

Registering with a Gatekeeper

• To use a gatekeeper, you must register your VoIP terminal

• To register, a client must configure the terminal to search for and register with the gatekeeper

Troubleshooting VoIP

• 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

• Relationship of perceived connection quality to one-way latency experienced in the connection:– Excellent 0 to 150 ms– Good 150 to 300 ms– Acceptable 300 to 450 ms– Unacceptable 450 ms or greater

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

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– Stratum 2 – clocks that receive timing

information from stratum 1 servers

Large Data Frames and Delay Budgets

• Frame – in VoIP, voice information embedded inside a UDP or TCP packet

• Multiple voice frames can be compressed into a single packet

• Compression:– Improves bandwidth efficiency– Increases latency

• Compression techniques:– G.723 standard– G.729 standard– G.711 standard

Calculating a Delay Budget

• If data packets are too large, a sudden burst of calls may exceed the bandwidth you have allocated

• To protect against this problem, you must create a delay budget to determine:– The type of data placed on the network– The number of trunks in use– The average number of calls, and amount of

bandwidth and line numbers required– The peak number of calls

Quality of Service (QoS) Issues

• QoS involves the ability to differentiate between voice and data IP packets, then route them accordingly

• The most common problem related to QoS is voice signal degradation

• In convergent networks, QoS involves routing IP packets according to information contained in the packet headers

QoS Technologies

• One way to prioritize VoIP traffic is to use the Type of Service (ToS) header in IPv4 packets

• Another way to prioritize VoIP traffic is to use the Differentiated Services (DiffServ) ToS header, which can distinguish among data types and assign priorities to data streams by marking packets

QoS Technologies (cont'd)

• Internet Integrated Services (IntServ)– Enables an application to determine the level of

delivery service for its data packets from among several defined choices

– Requires each network element to support QoS mechanisms

– Requires a means for communicating QoS requirements to each network element on the data stream's path

– Requires an end-to-end control message, provided by RSVP


• Resource Reservation Protocol (RSVP)– RSVP allows an application to request the QoS it

needs by sending end-to-end control messages along the data's path

– RSVP takes advantage of existing Internet routing protocols and algorithms to carry its messages

– RSVP and IntServ operate by reserving capacity in the network, based on the needs of a session, before the session is set up


• 802.1p– An IEEE signaling standard that prioritizes

network traffic at the MAC sublayer of the OSI data link layer by adding priority messages to packet frame headers

• 802.1q– An IEEE signaling standard, similar to 802.1p,

that was created for implementation in virtual local area networks (VLANs)

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

Voice Compressionand Decompression

• Talk spurts– Voice signal divided into short fragments of

20 to 40 bytes– Prevents delay of the voice transmission

• Voice compression standards– ITU G-Series protocols

• G.711• G.728• G.729• G.729A• G.723.1

Comparing and ContrastingTransmission Media

• T1 carrier– A North American high-speed digital carrier– Transmits data at 1.544 Mbps– Time division multiplexing (TDM) device creates

24 channels of 64-Kbps data streams• E1 carrier

– A European high-speed digital carrier– Transmits data at 2.048 Mbps– TDM device creates 32 channels of 64-Kbps

data streams• J1 carrier

– Japanese equivalent of T1 carrier

Bandwidth Limitationsfor Voice Traffic

• T1 and J1 trunks provide real-time voice traffic for 24 users

• An E1 trunk provides real-time voice traffic for 32 users

• If a T1 (or J1) and E1 line are connected:– Only about 80 percent of the E1 line would be

available – Converting the signaling between the two lines

would slow the connection • Modem and fax signaling requirements

– All voice and data digital signals enter and exit a network by manipulating the dial tone

VoIP Software and Hardware

• VoIP software and hardware allows users to conduct telephone calls between their computers and other VoIP-enabled computers

• Line doubling – transmitting data and placing a phone call at the same time using a dial-up connection with IP telephony and the H.323 standard

• Advantages of line doubling:– Reduced telephony costs– Off-site workers with access to only one phone

line can transmit data and make calls at the same time

VoIP Software and Hardware (cont'd)

• Advantages of using VoIP phone technology:– More efficient use of network wiring – Easier relocation and rearrangement of IP-based

hard or soft phones – Easier relocation to another building, state or

country — anywhere a suitable Internet, intranet or VPN connection is available

• Precautions to observe:– Increased traffic on the LAN– Provision of power

Common VoIP Applications

• Microsoft NetMeeting – for end point communications on Microsoft Windows systems only

• GnomeMeeting – for end point communications on Linux systems

• ICUII – for end point communications on various platforms

• OpenPhone – a client for Windows that supports end point communications with various clients, including NetMeeting, GnomeMeeting and ICUII

Summary

Discuss the functions of gatekeepers Discuss the functions of gateways Define delay, latency, jitter and wander, and

identify their impact on real-time communications Identify the importance of a jitter buffer Identify the impact of large data frames on real-

time communications Recognize the need for Quality of Service (QoS) for

converged networks Identify QoS technologies for converged networks

Summary (cont'd)

Identify common codecs and their bandwidth requirements in a converged environment

Describe the impact of compressing voice in a network

Compare and contrast the use of T1, E1 and J1 trunks for data and voice

Identify the factors that affect the bandwidth of packetized voice

Identify requirements for transporting modem and fax transmissions through a converged solution


Lesson 3:Network Convergence

Objectives

• Identify characteristics of circuit-switched and packet-switched technologies

• Identify the differences between the call flow in convergence-based calls and the call flow in circuit-based calls

• Identify the types of signaling protocols for converged networks

Characteristics ofConvergent Networks

• Integrated Services Digital Network (ISDN) – one of the first attempts to integrate voice and data onto a single network

• Three basic voice packet technologies in converged communication networks:– Voice over IP (VoIP)– Voice over Frame Relay (VoFR)– Voice over Asynchronous Transfer Mode (VoATM)

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

ConvergenceSignaling Protocols

• Types of signaling protocols used in converged networks:– International Telecommunications Union (ITU) H-Series

protocols:• H.320• H.323• H.225• H.245• H.450

– Session Initiation Protocol (SIP)– Media Gateway Control Protocol (MGCP)– Network Call Signaling (NCS)

H-Series Protocols

• The H Series of ITU recommendations discusses protocols and mechanisms for audiovisual and multimedia systems – H.320 – governs the basic concepts of videoconferencing

that combine telephony, video and graphical communications

– H.323 – defines the components, procedures, protocols and services for multimedia communication over both LANs and WANs

– H.225 – provides call signaling and media stream packetization for packet-based multimedia systems

– H.245 – uses TCP to ensure reliable data and teleconferencing communication

– H.450 – offers supplementary services for converged networks

Session Initiation Protocol (SIP)

• Session Initiation Protocol (SIP)– An IETF signaling protocol used for Internet

conferencing and telephony– An alternative to H.323

• SIP ports– UDP port 5060 (default)– TCP port 5060

• SIP names and addresses (examples)– sip:user@host– sip:[email protected]

Session Initiation Protocol (SIP)(cont'd)

• SIP components– User agents (UAs)– Network servers

• Proxy servers• Redirect servers• User agent servers• Registration servers

• SIP messages– Requests – issued by clients– Responses – issued by servers

SIP Transactions

INVITE user2@host

ACK user2@host

200 OK

INVITE user2@host

ACK user2@host

200 OKSIPUser Agent 1user1@guest

SIP Server(host)

SIPUser Agent 2user2@host

SIP Standards

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 commands:– Create Connection (CRCX)– Modify Connection (MDCX)– Delete Connection (DLCX)– Notification Request (RQNT)

• MGCP user connections – the media gateway controller creates connections on each end point involved in the call

Network Call Signaling (NCS)

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

Bandwidth Concerns

• Client configuration– Silence suppression

• Calls/faxes include periods of silence • By suspending the sending of packets when

pauses occur, bandwidth is conserved– Fast start

• Skipping H.323 steps so connections can occur more quickly

• Enables billing messages to be sent from a gatekeeper or remote client

Bandwidth Concerns (cont'd)

• Network configuration– Trunk duty cycle

• Periods of silence occur when a gateway trunk is inactive

• By increasing the maximum average percent of time a trunk is active, bandwidth consumption is reduced

– Carrying capacity• VoIP trunks carry real-time and non-real-time data• By using silence suppression and increasing the duty

cycle, bandwidth consumption is reduced• Telephony capacity – the number of real-time

calls/faxes that can be accommodated within the total access bandwidth

VoIP Service Providers

• VoIP service providers generally provide:– Software that allows you to place calls using a

personal computer – The ability to call people who use standard

analog telephones – SIP and H.323 support – The choice of the most common codecs

• VoIP service providers may not provide:– Number portability– Location information for 911 emergency

services

VoIP and Firewalls

• Network Address Translation (NAT)– The process of translating one IP address into another– Required for computers with private IP addresses to use

the Internet– May cause problems for VoIP protocols (may drop the

media stream portion of a call)• Types of NAT:

– Port address translation (PAT)– Static address translation– Dynamic address translation

• NAT variations:– Full cone– Restricted cone– Port-restricted cone– Symmetric

VoIP and Firewalls (cont'd)

• Simple Traversal of UDP through Network Address Translators (STUN)– Helps VoIP clients traverse non-symmetric NAT-enabled

routers and firewalls– STUN consists of:

• STUN client• STUN server

– STUN benefits:• Do not have to change NAT setup or proxy server• STUN may provide QoS and decrease latency

– STUN drawback:• Does not resolve issues involving routers and firewalls

that perform symmetric NAT

VoIP and Firewalls (cont'd)

• Symmetric NAT causes the most connection challenges, especially with calls that use SIP

• To solve symmetric NAT problems, consider:– Port forwarding – forwards packets received at a

VoIP port to an internal IP address– Using TCP instead of UDP– Implementing vendor-specific solutions

Planning aConvergent Network

• Determine bandwidth required• Identify network use• Review infrastructure• Identify protocols• Determine cost• Use a test network• Create implementation plan• Deploy incrementally• Identify router problems• Identify challenges

Summary

Identify characteristics of circuit-switched and packet-switched technologies

Identify the differences between the call flow in convergence-based calls and the call flow in circuit-based calls

Identify the types of signaling protocols for converged networks

Convergence Technologies

Convergence Industry Standards and Protocols Enabling Voice over IP (VoIP) Network Convergence

copyright © 2006 prosoft learning, a vcampus company - all rights reserved. convergence...

Documents

convergence convergence

convergence technologies

mgcp slide

convergence technologies

various itu protocols

convergence industry

voice networks

various ieee