Transcript Convergence_Tech_v3_0_PowerPoint

Convergence
Technologies
Copyright © 2006 Prosoft Learning, a VCampus Company - All rights reserved.
Lesson 1:
Convergence Industry
Standards and Protocols
Copyright © 2006 Prosoft Learning, a VCampus Company - All rights reserved.
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)
• IEEE 802 protocols – define the relationships
between physical network interfaces and all
signaling
– 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
International
Telecommunication 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
International
Telecommunication Union (ITU)
(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
International
Telecommunication Union (ITU)
(cont'd)
• ITU T-Series protocols – discuss different types of
terminals for telephony services
– T.30
– T.37
– T.38
– T.134
International
Telecommunication Union (ITU)
(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
Agreement
o A precursor to a standard that has formed the
basis of subsequent standards
o 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)
Electronic
Industries 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 National
Standards 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)
Copyright © 2006 Prosoft Learning, a VCampus Company - All rights reserved.
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
QoS Technologies
(cont'd)
• 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
QoS Technologies
(cont'd)
• 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 Compression
and 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 Contrasting
Transmission 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 Limitations
for 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 realtime 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
Copyright © 2006 Prosoft Learning, a VCampus Company - All rights reserved.
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 of
Convergent 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
Convergence
Signaling 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
SIP
User Agent 1
user1@guest
200 OK
ACK user2@host
INVITE user2@host
SIP Server
(host)
200 OK
ACK user2@host
SIP
User Agent 2
user2@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 a
Convergent 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