Aarti Iyengar

Solutions Architect
Polycom

Introduction
VoIP Network Components/Protocols
Performance Parameters
Regulations
Security
Economics
Applications
Evolution Scenarios

 Legacy Telephony

TDM/SS7 based infrastructure

Traditional Class 5/Class 4 switches

Voice over IP

IP-based packet infrastructure for PSTN voice transport

New elements that collectively perform traditional
functions and more

And what is IP Telephony?

Voice + Messaging + Video + Data over IP networks = IP

Telephony
Public Internet : Best Effort Service
Managed IP Network : SLA based Service

Filter
lower/higher
frequencies

Analog Voice
(400-4000kHz)

Echo Cancellation(if
reqd.)
Digitized Voice Packet
encapsulated within
RTP packet

Sample at 8000
samples per
second

Silence Suppression
(Optional). Yields ~50%
bandwidth gains
RTP packet
encapsulated within
UDP packet

Quantization

Compression (Optional).
Yields further bandwidth
gains
UDP packet
encapsulated within IP
packet

Voice Over IP Packet

Voice Samples
RTP Payload

RTP Header

UDP Payload

UDP Header

IP Payload

IP Header

Digital
Encoding
into bytes

SS7
signaling

SS7 network

Legacy
Class 4/5
Switch

Legacy
Class 4/5
Switch

Legacy
Class 4/5
Switch

TDM network
TDM
bearer

Call Control,
Signaling,
Bearer/Media
and Features

SS7
signaling

IP signaling
+
IP bearer

SS7 network

Soft IP Phones

Application
Server

Signaling
Signaling
Gateway

TDM
bearer

Media
Gateway
Controller

IP network

Media
Gateway

Features

Call
Control

IP Phones

Media
Server

Media
(conferencing)

Media
Gateway

TDM network

Bearer/
Media

!!
Terminals or Endpoints
IP Phones
Soft Phones/PC Phones

Media converter
Media Gateway/PSTN Gateway

Call Processor
Media Gateway Controller or Gatekeeper or Proxy Server or
Softswitch

Signaling Gateway
Application Server
Media Server

"
Central
Intelligence

Centralized
Model

Distributed
Model Intelligent

Distributed
Intelligence

Master
Controller

Intelligent
Server
Dumb
Slave
GWs

Dumb
Slave
GWs

Intelligent
Client
Dumb
Slave
GWs

Server

Dumb
Slave
GWs

Dumb
Slave
GWs

Intelligent
Client

Intelligent
Intelligent Server
Intelligent
Client
Server

H.323

MGCP

SIP

Megaco/H.248

Call Control/Signaling
TRI
P
EN
M

More .

Call Control/Signaling
Gateway Control

RTP

DP

SIP-T

Bearer

BICC
Softswitch-Softswitch

"

"

# $ !% & %

ITU-T defined standard

Originally developed for ISDN based multimedia services
over LAN
Distributed protocol model
Consists of

Terminals
Gatekeepers
Gateways
Multipoint control units

Umbrella protocol comprising of several other protocols

like H.225, H.245, T.120 etc. defining RAS, capability
negotiation etc.
Binary ASN.1 encoding
H.323v4 currently implemented everywhere
Future H.323v5

"

"

# $ !% & %

Gatekeeper

Gateway

H.323 endpoint

IP network

Gateway

H.323 endpoint

"

"

IETF RFC 3265 (obsoletes RFC 2543)

Developed for multimedia services over IP
networks based on http model
Designed to employ existing popular
Internet protocols like DNS, SDP etc.
Distributed model consisting of User agents
and Servers
Text-based implementation is perceived to
be simpler, modular, easily adaptable to the
www

"

"

Registrar

Proxy
Server

SIP Phone

Redirect
Server

IP network

Location
Server

Proxy
Server

SIP User Agent

$ !% & %

#'

ITU protocols more tightly defined; IETF looks for looser

working code
H.323 older and more established; SIP relatively newer but fast
catching up
H.323 widely deployed today; SIP is being widely adopted by
large players
Importance of the Internet and web-based applications
increasing
SIP capable of giving service providers greater control of
services, extensibility and interoperability with the www;
hence, may eventually win the race
For a long time however, both these protocols need to co-exist
Robust standards must be developed to define interoperability to
make things easier

"

Source: Hughes Software Systems

"

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"

"

#* )

IETF informational RFC 3661

Provides call control services in a packet network
Early implementation of Master/Slave protocol
Consists of media gateways and call agents
Call Agents-> centralized intelligent entities handling call
control and signaling
Media Gateways-> dumb devices handling media
Call Agent communicates with Media Gateway via MGCP

Now a closed effort from standards perspective

MGCP implementations do exist today. MGCP variants
NCS/TGCP are adopted by Packetcable.

"

"

#* "

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Enhances MGCP
Joint effort by ITU and IETF (IETF nomenclatureMegaco/RFC 3525, ITU nomenclature- H.248)
Provides call control services in a packet network
Adopts the Centralized model
Supports IP/ATM networks
MGC-MG communication via Megaco/H.248
Deals with contexts and terminations

decouples physical terminations from logical (ephemeral)

ones
more suited to handling multimedia

More complete and robust, standard allowing for multivendor interoperability

IETF RFC 3372

Defines a framework to interface SIP with ISUP
To maintain feature transparency in the SIP network
w.r.t PSTN to support IN services not supported in SIP
To deliver SS7 information (in its entirety ) to some
trusted SIP elements

Integration methods

Encapsulation of ISUP within SIP using MIME

Translation of ISUP parameters to SIP header
Provision to transmit mid-call ISUP signaling messages
through INFO method

Implemented at SIP-PSTN boundary

gateways
Carried end to end

SIP-T is relevant in the following scenarios

PSTN origination, IP termination
IP origination, PSTN termination
PSTN origination, PSTN termination with IP
transit
IP origination, IP termination : SIP-T is not
required

#/

Development triggered by a need for a packetbased PSTN replacement

Functional separation of call and bearer signaling
protocols in a broadband network

IP/ATM bearers in addition to TDM bearer

Uses SS7 signaling (with extensions to ISUP)
Binding information allows correlation between call
control and bearer

BICC defines three capability sets

CS1: supports ATM-based (AAL1/AAL2) bearer

CS2: supports IP-based bearer
CS3: still in works to support advanced services and
interoperability with SIP

/
SIP-T
IETF defined
Defined to maintain feature
transparency across SIP
networks (deliver ISUP to
SIP endpoints)
Packet-based signaling and
bearer

SIP signaling
IP bearer; ATM supported
through RFC 3108 (may be
some issues, but defined)

Provides a transition to pure

advanced multimedia
services based SIP network

BICC
ITU defined
Defined to separate call
control from bearer (extends
ISUP to handle packet
bearers)
SS7 signaling, Packet bearer
Network is SS7 (ISUP)
signaled
TDM/ATM/IP bearers
Intended for packet-based
next-generation network
supporting all existing
legacy services.

# '
IETF RFC 3550 (obsoletes RFC 1889)
End-to-end network transport services for
multimedia applications
Services include payload type identification,
sequence numbering, time stamping and delivery
monitoring
Control protocol (RTCP) to monitor data delivery
Can be used with any transport protocol
Depends upon underlying transport layer for QoS
Applications typically use RTP over UDP

" 0

"

"

Protocol selection is a strategic decision depending on existing network and future services planned
Ultimately, one winner will make it easy for all !

CP
MG

i
ed

tra
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GC
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P
TC
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TP
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IP Phone (H.323/SIP)

IP Network
M

dia
Me

P
TC
R
/
RTP

Signaling
Gateway
A
3U
M
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T P)
C
/S

(SS7)
PSTN
(TDM)

48

Media
Gateway

7
SS

SIP-T
/BICC

Media
Gateway
Controller

248
/H.

Media
Gateway
Controller

Application
Server
SIP

Media
Server

TDM

2 (

Coding Algorithms
Echo Cancellation
Latency
Jitter/Jitter Buffer
Packet loss
Transcoding/Tandeming
QoS
Reliability/ Availability
Quality

" 0 "
Compression

What codec is used and their corresponding bit rates

Greater the compression, more the encoding delay
Determining appropriate packetization times and packet
length

MOS score of codec determines perceived quality

VAD and CNG

At the transmitter

Detection of voice activity

Suppression of silence

At the receiver
Comfort Noise generation
Voice playback

 Echo detection and cancellation

Availability and echo signal return loss
quality
Adjustments to loudness rating
Tail length is MG role dependent

( +4

Packetization Delay
Propagation Delay
Network Processing Delay
Jitter buffer delay and speech playback
PLCs add about 5ms delay
PSTN benchmark for toll quality voice is
150ms RTT (ITU G.114)
Delay greater than 300ms is completely
unacceptable for toll quality
An occasional packet loss is tolerable, but
latency beyond 200 ms RT is not.

+5

Jitter = Delay Variation

Jitter Buffer compensates for jitter on the
receiver side
Jitter buffer size should be optimally chosen
Rule of thumb: Jitter Buffer size = atleast 2 x
speech frame size
Absolute jitter buffer size = end-to-end delay
variation + some safety margin
Used by Gateways that have more processing power

3
Packet loss should be below 1% for acceptable
quality
Use Codecs with packet loss concealment
algorithms

E.g G.729, G.723.1 have built in PLC; add-on PLCs have to

be used with G.711 and G.726

PLC algorithms compensate about 40 ms of missing

speech.
Delay >40ms & <200 ms, speech is clipped
Delay >200ms, speech dropouts

Packet loss is mostly bursty in nature. Hence,

packet loss performance is directly related to packet
size, the shorter the better

"+

"

Transcoding: Two or more encodings of a signal

through different types of non-G.711 codecs
separated by G.711 e.g G.726 to G.711 to G.729A
Tandeming: Two or more encodings of a signal
through same types of non-G.711 codecs separated
by G.711 e.g G.729A to G.711 to G.729A
Transcoding increases distortion and delay
Only one transcode can be tolerated before the
network performance drops to unacceptable levels
for most combinations of non-G711 codecs

(
Means to prioritize voice packets
Real time voice packets receive higher
priority than non-real time data packets
Helps improve performance by decreasing
delay/jitter for voice packets
Significant delay/jitter events can be avoided
only by implementing a proper QoS Strategy

"(

Best Effort

Prioritized Queuing

A class of service in which the

network provides no
guarantees to the edge
equipment
Differentiation in the queuing
of traffic for various classes of
traffic
Assigns a priority or
classification to every IP
packet
Packets are sent in order of
priority

Traffic Engineered Tunnels

ConstraintConstraint-based (traffic
sensitive) connectionconnection-oriented
paths through a routed
network
MPLS Label Path
ATM VC

Speech quality important for

Monitoring/fault-finding
Service level agreements
Optimisation of network

Quality will remain an issue so

long as bandwidth or
processing power are limited

e.g. mobile, leased capacity

Factors that affect

quality

Background noise
Silence suppression
Low bit-rate coding
Errors (mobile, packet)
Delay
Echo
Handsets/access
network

Quality measures: MOS; PSQM (Perceptual Speech Quality

Measure); PAMS (perceptual analysis measurement system);
PESQ (Perceptual Evaluation of Speech quality)
End-to-end speech quality is the key measure of voice QoS

/ ""

"

The VoIP network must:

-provide a customers service preferences anywhere
- securely
-adjusting to the constraints of the networks and access
methods used: Wireline, Wireless, 3rd Party, Customer
Owned, or Public Internet
Clearly a list of codecs, packet sizes, loss rate and jitter targets
are needed to ensure voice quality to a defined level of
acceptability.

' 7

Viewing packets as part of sessions

Policies are required for Sessions
New IP Services are enabled to handle this
Routing sessions between different networks,
carriers and domains
Session packet flow anchoring
Detect failures and reroute
Usage based Billing/reporting at session flow
level
Session aware borders for security

 Two way VoIP communication impeded by

NAT/NAPT
Signaling messages can be exchanged on defined port
Bearer messages are a problem

Special tags in SIP message to permit two way

communication
Simple Traversal of UDP messages over NAT
(STUN)
Creates NAT awareness in Clients
To modify SDP messages

/
Applicatio
n Service Session
Provider
Border
Controller

Softswitch

IP
Backbone

PSTN
Gateway
PSTN

Session
Aware
Firewall

Enterprise

Session
Border
Controller
IP Service
Provider

VoIP Firewall Traversal Solution for

Carrier to Carrier Peering
Integrated SIP Application Layer
Gateway (ALG)
Modify signaling traffic to
accommodate NAT/NAPT
Dynamic pinhole opening/closing
Topology Hiding
Provide an address normalization
boundary
VoIP Media Anchoring Solution
VoIP Session QoS / Service Level
Agreement Solution
Per session based policing
Guaranteed service in congested
environments
VoIP Session Admission Control
Solution

' "

' "
Numbering Services

Rate Centre Association of Numbers

Impact on Number Conservation
Number Portability Compliance for VoIP providers?

Information service versus Telecommunications

service

Access charges at Origination and Termination points?

CALEA

Requires North American telecommunications carriers to

modify their equipment, facilities, and services to ensure
that they are able to comply with authorized electronic
surveillance.
Similar requirement for the VoIP?

 What Service Providers want

Decrease expense
Lower capex on new infrastructure elements introduced
Lower opex on maintenance of existing infrastructure

Increase ARPU
Increase talk-time and use of Value added services
(Centrex)

Ease of feature incorporation

Ease of Application development and innovation
Distributed architecture, fewer elements

""

In 2003, IP accounted
for the majority - 58% of
network traffic - TIA
International VOIP calls
grew from 0.2% of
telephone traffic in 1998
to 10.4% in 2002 Broadwatch News
Analysis
Total Internet protocol
(IP) revenues expected
to grow in 2004 by 7.8
percent, achieving a
total of $13.9 billion
TIA
3

Major VoIP Providers

Vonage - 70,000 subscribers
Free World Dial-up - 80,000 subscribers
Yahoo BB - Over 3 Million subscribers
FastWeb - 400K subscribers

IP PBX
IP Centrex
Enhanced IP Telephony
Class 4 Replacement
Class 5 Replacement
And more..

/8
A CPE based IP telephony service that replaces a traditional
TDM PBX

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, '
)Scenario
)ILECs, CLECs, IXCs, Large Corporations
)Benefits
)By-pass traditional long distance toll network
(Class 4) carriers and their per-minute usage rates
and run their voice traffic over IP networks for a
reduced cost.
)Lower costs with higher bandwidth efficiency
)Issues
)Traffic engineering of IP network for PSTN QoS
)Migration from Circuit to Packet-based Network

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: '
Softswitch
PSTN

Telephone

IAD

Broadband
Network
Media
Gateway

IP Network

Scenario
Out of Region and Greenfield deployments
Benefits
Flexibility - Enable Rapid Deployment of New Services
Distributed Architecture rather than Hierarchical Class Model
Issues
Maturity of softswitch technology
Ability to support all legacy systems supported by a Class 5
switch

* "
CLASS 5 DERIVED

Broadband
Network

IAD

Media
Gateway

Class 5 Switch

CLASS 5 + SOFTSWITCH

IAD

SOFTSWITCH ONLY
Softswitch
PSTN

IAD

Broadband
Network
Media
Gateway

IP Network

Softswitch

Broadband
Network
Media
Gateway

Class 5 Switch

PSTN/SS7
Network

SS7/PSTN
Network
Softswitch

Softswitch

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8

SIP

Voice Over Broadband

48
.2

DSL Cable Wireless

IP Network
Gateway

Gateway
TDM Network
Class 4

Class 5
Class 4

PSTN/SS7
Network

GR-303 Interface

Gateway

.)
C2P Packet Core

Central Office
Line
GW

IP

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IP

Trunk
GW

TDM

Digital
Centrex

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TD

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IXC

AS

Single IP Pipe
for Voice,Data
and Multi-Media

LD
Network

IP
Campus
Network

IP

Wireless
Device

IP

Enhanced IP
Telephony
(SIP)

IP

Data PC

IP

IP

Multi Service
Edge Switch

TDM

IP

Analog Centrex

Integrated
Desktop
(SIP Soft Client)

PSTN

IP

/W

AP

Internet
Mobile Network

PDA or
Laptop

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