Transcript Document

VoIP Conception
and
Implementation
LANtel Telecommunication Corp.
Senior Product Manager
Jeremy Chan
Agenda
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What’s VoIP and IP Telephony (IPT)
VoIP Applications
VoIP QoS Issue
VoIP Architecture
VoIP Signaling
Fax over IP (FoIP)
What’s VoIP and IP Telephony (IPT)
VoIP
VoIP :
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VoIP, Voice over Internet Protocol, is the
technology that uses the Internet
Protocol to transmit voice conversation
over a data network.
The primary advantages of moving voice
over a data network are increased
efficiency and decrease cost.
IPT (IP Telephony)
IPT (IP Telephony) :
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An IP Communications System that
provides a high availability and
scalability telephony system.
Provide support industry standards such
as H.323, MGCP, SIP, JTAPI, TAPI, …
etc. VoIP signal protocol.
Enterprise Voice Solution
PBX
PBX
PSTN
PBX
PBX
IP WAN
Router/GW
Router/GW
Soft-Switch
Soft-Switch
IP WAN
Router/GW
Application Servers
Router/GW
Packet Voice Technology
VoIP Applications
Branch Office Application
HQ
Branch 1
IWF
Server Farm
Telephone
IWF
Packet Network
Branch N
PBX
Telephone
*IWF: Interworking function
PSTN
IWF
Telephone
Interoffice Trunking Application
Packet Network
PBX
Telephone
PBX
Telephone
Interoffice Trunking Application
VoIP QoS Issue
VoIP QoS Issue
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Delay
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Algorithmic Delay
Processing Delay
Network Delay
Jitter
Lost-Packet Compensation
Echo Compensation
Delay
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Algorithmic Delay
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Collect a frame of voice samples to be processed by
the voice coder.
G.726 adaptive differential pulse-code modulation
(ADPCM) (16, 24, 32, 40 kbps)—0.125 microseconds
G.728 LD–code excited linear prediction (CELP)(16
kbps)—2.5 milliseconds
G.729 CS–ACELP (8 kbps)—10 milliseconds
G.723.1 Multirate Coder (5.3, 6.3 kbps)—30
milliseconds
Delay
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Processing Delay
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Actual process of encoding and collecting
the encoded samples into a packet for
transmission over the packet network.
The encoding delay is a function of both
the processor execution time and the
type of algorithm used.
Delay
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Network Delay
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Physical medium and protocols used to
transmit the voice data and by the buffers
used to remove packet jitter on the receive
side.
Network delay is a function of the capacity
of the links in the network and the
processing.
Delay Causes Problems
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Echo
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Signal reflections of the speaker's voice from the
far-end telephone equipment back into the
speaker's ear.
Round-trip delay becomes greater than 50
milliseconds. (G.131)
Talker overlap
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one talker stepping on the other talker's speech
the one-way delay becomes greater than 250
milliseconds. (G.114)
Jitter
Variable interpacket timing caused by the
network a packet traverses.
 Removing jitter: collecting packets and
holding them long enough to allow the
slowest packets to arrive in time to be
played in the correct sequence.
 Causes additional delay
Lost-Packet Compensation
Lost packets can be an even more severe
problem, depending on the type of packet
network that is being used.
 Interpolate for lost speech packets by
replaying the last packet received during the
interval.
 Send redundant information.
 Use a hybrid approach with a much lower
bandwidth voice coder to provide redundant
information.
 Avoiding and Managing network congestion
Echo
Normal Telephony Call
Normal Telephony Call with an Echo
Echo Compensation
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Signal reflections generated by the hybrid
circuit that converts between a four-wire
circuit (a separate transmit and receive pair)
and a two-wire circuit (a single transmit and
receive pair).
The round-trip delay through the network is
almost always greater than 50 milliseconds.
ITU standard G.165 defines performance
requirements that are currently required for
echo cancellers.
VoIP Architecture
VoIP–Embedded Software Architecture
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Voice Packet Software Module
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Telephony-Signaling Gateway Software
Module
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Translating signaling into state changes used by the
packet protocol module to set up connections.
Packet Protocol Module
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digital-signal processor (DSP)
processes signaling information and converts it.
Network-Management Module
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Voice-management interface to configure and
maintain the other modules
VoIP Signaling
Signaling – H.323
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H.323
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Entities
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Umbrella standard covering multimedia
communications over LANs that do not provide a
guaranteed Quality of Service. (H.323 v1)
Terminals
Gateways
Gatekeepers
MCUs
Protocols
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Parts of H.225.0 - RAS, H.225 (Q.931)
H.245
RTP/RTCP
Audio/video codecs
H.323 Protocol Stack
Data
Audio Signal
G.711
G.722
G.723.1
T.127
G.728
G.729
Video Signal
H.261
H.263
Presentation
Session
Transport
RTCP
RAS
T.124
RTP
Supplementary Services
H.450.3
H.235
UDP
Network
Data Link
Physical
Control
H.225
IP
T.125/T.122
H.450.2
H.450.1
H.245
T.126
X.224.0
TCP
H.323 protocols
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H.225
Covers narrow-band visual telephone services
H.225 Annex G
H.235
Security and authentication
H.245
Negotiates channel usage and capabilities
H.450.1 Series defines Supplementary Services for H.323
H.450.2 Call Transfer supplementary service for H.323
H.450.3 Call diversion supplementary service for H.323
H.450.4 Call Hold supplementary service
H.450.5 Call Park supplementary service
H.450.6 Call Waiting supplementary service
H.450.7 Message Waiting Indication supplementary service
H.450.8 Calling Party Name Presentation supplementary service
H.450.9 Completion of Calls to Busy Subscribers supplementary service
H.450.10 Call Offer supplementary service
H.450.11 Call Intrusion supplementary service
H.450.12 ANF-CMN supplementary service
H.261
Video stream for transport using the real-time transport
H.263
Bitstream in the RTP Q.931manages call setup and termination
RAS
Manages registration, admission, status
RTCP
RTP Control protocol
RTP
Real-Time Transport
T.38
IP-based fax service maps
T.125
Multipoint Communication Service Protocol (MCS).
H.323 Architecture
Typical H.323 Deployment
Signaling – MGCP, MAGACO
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Media Gateway Control Protocol
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Entities
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Using packages model and providing an centralized
architecture where call control and services.
Controlling Telephony Gateways from external call control
elements called media gateway controllers or call agents.
MGC (Media Gateway controller / Call agent)
MG (Media Gateway)
Protocols
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MGCP v1 – RFC 2705
H.248 (H.248 / MAGACO) – RFC 3525
SDP (Session Definition Protocol) - RFC 3407
MGCP Architecture
PSTN
MGCP
Voice Gateway
T1/E1
FXO/FXS
E&M
IP Phone
( MGCP Client )
Call Agent
IP Phone
( MGCP Client )
PBX
MGCP
RTP
Signaling – SIP
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Session Initiation Protocol
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Entities
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Multimedia protocol that could take advantage of the Internet
model for building VoIP networks and applications. Using
distributed architecture.
User Agent
Gateways
Proxy Server
Redirect Server
Registrar Server
Protocols (RFC 2543 v1, RFC 3261 v2)
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SDP ( Session Definition Protocol )
URLs
DNSs
TRIP ( Telephony Routing Over IP
SIP Architecture
ENUM
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“ENUM protocol is defined by RFC 2916, aiming at
translating the numbers stemming form the ITU-T
E.164 Recommendation into Internet Domain Names;
ENUM is an opportunity for developing the
information society.”
“As a matter of fact, ENUM allows to use a traditional
telephone number in the context of different
communications media, in particular those rising from
the development of IP networks (e-mail, VoIP, …) and
therefore, could facilitate the penetration of new
applications into the mass market easily ( this market
is accustomed to E.164 numbers).”
ENUM (Cont.)
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ENUM is part of Convergence
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ENUM is part of series of technical initiatives
underway in both the IETF and ITU to develop
Internet Telephony Standards.
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Call Setup – H.323 – SIP
Quality of Service – DIFFSERV – INTSERV – MPLS
PSTN – IP Interworking H.248/MEGACO
FAX – T.37, T.38 – RFC 2503
Mobile – 3GPP related
ENUM is about new service creation
It must address naming and numbering
issues
VoIP Signaling Comparison
VoIP Signaling Comparison
Fax over IP
FAX over IP
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ITU and Internet Engineering Task
Force (IETF) are working together to
continue to evolve both the real-time
FoIP network standard (T.38) as well as
the store-and-forward FoIP network
standard (T.37).
T.38 is the fax transmission protocol
selected for H.323.
FoIP QoS
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Timing
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Jitter
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network delay
processing delay
IWF must compensate for the loss of a fixed timing
of messages over the packet network.
collect packets and hold them long enough so that
the slowest packets to arrive are still in time to be
played in the correct sequence.
Lost-Packet Compensation
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repeating information in subsequent frames
using an error-correcting protocol
Reference
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Cisco
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REDCOM
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Voice and Fax over Internet Protocol (V/FoIP)
ENUM.ORG
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H.323 Tutorial
IEC
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Introduce H.323
SIP Presentation
Study Group A Presentation on ENUM
IETF
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ftp://ftp.isi.edu/in-notes/rfc2916.txt -- ENUM Core Protocol
ftp://ftp.isi.edu/in-notes/rfc3261.txt -- SIP
ftp://ftp.isi.edu/in-notes/rfc2705.txt -- MGCP
Thank You