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Understanding Voice over IP

Protocols
Cisco Systems—Service Provider Solutions Engineering
February, 2002

4426_02_2002_c1 © 2002, Cisco Systems, Inc. All rights reserved. 1


Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

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Why Move to VoIP?

• Cost savings—toll bypass


• Open standards—H.323, SIP, MGCP
• Multi-vendor interoperability
• Integrated IP voice and data networks

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Cisco Packet Voice Architecture

Open Service
Application Layer
TDM/ (JAIN, AIN, TAPI,
Circuit Switch JTAPI, XML etc.)
Open/Standard
Line Call Control Interface
Switching Network

Concentration Connection Control


Features

Digital Trunk
Subsystem
Common Channel
Signaling Complex
Open Call Control Layer
(SIP, H.323, MGCP, etc.)
Administration
Maintenance
Billing

Open/Standard
Interface
Standards-Based
Standards-
Packet Infrastructure Layer
(IP, ATM)

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Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

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Early Adopters—
Advanced Services and Toll-Bypass

• Regulatory opportunities
allowed for toll-bypass
• PC-to-phone, calling-card
and international fax
services
• Cisco-based carriers used
standard protocols, but not
all carriers implemented
standards
• Inter-carrier connections
had protocol
interoperability challenges

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Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

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Making the Rules for VoIP

• IETF (Internet Engineering Task Force)


The community of engineers that standardizes the
protocols that define how the Internet and Internet
Protocols work. http://www.ietf.org/
• ITU (International Telecommunications Union)
An international organization within the United Nations
System where governments and the private sector
coordinate global telecom networks and services.
http://www.itu.int/home/index.html

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Defining the VoIP Protocols

• H.323
An ITU Recommendation that defines “Packet-based multimedia
communications systems”. H.323 defines a distributed architecture for
creating multimedia applications, including VoIP
• SIP
Defined as IETF RFC 2543. SIP defines a distributed architecture for
creating multimedia applications, including VoIP
• MGCP
Defined as IETF RFC 2705. MGCP defines a centralized architecture for
creating multimedia applications, including VoIP
• H.248
An ITU Recommendation that defines “Gateway Control Protocol”. H.248 is
the result of a joint-collaborate with the IETF. H.248 defines a centralized
architecture, and is also known as “Megaco”
• Megaco
Defined as IETF RFC 2885. Megaco defines a centralized architecture

4426_02_2002_c1 © 2002, Cisco Systems, Inc. All rights reserved. 9


Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

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H.323 Components
H.323
MCU Scope of
GK H.323
e

H.323 Packet H.323


Gatekeeper Network Terminal

H.323
Gateway

PSTN ISDN

V.70 H.324 Speech H.320 Speech


Terminal Terminal Terminal Terminal Terminal

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Scope of H.323 Recommendation

Video I/O Video Codec


Equipment H.261, H.263
Receive RTP
Pain
Delay UDP
Audio I/O Audio Codec RTCP
(Sync)
G.711, G.722,
Equipment G.723, G.728,
G.729

User Data H.225


Applications Layer IP
T.120, etc. System Control

H.245 Control TCP

System Call Control


Control H.225.0
User RAS Control
Interface H.225.0 UDP

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H.323 Signaling

V V
H.323
H.323 Endpoint B
Endpoint A Setup
Alerting / Connect H.225 (TCP Port 1720)

Capabilities Exchange / MSD


H.245 (TCP
Open Logical Channel Dynamic Port)
Open Logical Channel Acknowledge

RTP Stream
RTP Stream Media (UDP)
RTCP Stream

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Basic H.323 Call

Gatekeeper A Gatekeeper B
LRQ
LCF
ACF ACF

RRQ/RCF IP Network RRQ/RCF

ARQ H.225 (Q.931) Setup ARQ


H.225 (Q.931) Alert and Connect
H.245
V RTP V
Gateway A Gateway B

Phone A Phone B

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Deploying H.323 Networks
• Minimizes GK configuration
• Addition of new zones
• Addition of new NPAs
DGK
• Addition of new rate centers

GK GK GK

West Chicago NY
LA Zone LA GW GW
GW #1 GW #2
Rate Intra-LATA Rate Local Midwest Local East
Center #1 Toll Center #1 PSTN Zone PSTN Zone

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MGCP/H.248/Megaco—Architectures

Call Agent Call Agent


SS7

P P
IMT
S S
T T
PSTN
N N
PRI

Access
Gateway

MGCP / H.248 / Megaco


RTP

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Deploying MGCP/H.248/Megaco
Networks
OSS
CA
SS7 Billing and
Backhaul Measurement
MGCP Server
SLT and ISDN
SS7 Backhaul
MG Service
STP
MG
MG Provider's
PSTN TDM
IMTs TDM Network
Voice
VoIP
Service
ISDN/PRI Provider's
Packet
Network
NAS/VoIP
Traditional TDM Traffic
Modem Dial-up Traffic

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SIP Architecture
I
N
Application
T
3pcc Services
E eMail CPL
CPL
LDAP Oracle XML
L
L
I
G SIP Proxy, Registrar
E & Redirect Servers
N
T SIP

SIP SIP PSTN


S SIP User
E Agents (UA)
R
CAS or PRI
V
I RTP
C (Media)
E
S4426_02_2002_c1 Legacy PBX
© 2002, Cisco Systems, Inc. All rights reserved. 18
SIP Signaling

PSTN PSTN
SIP VoIP Network
INVITE
Calling Party Called Party
100 Trying INVITE
SIP Signaling
100 Trying
and SDP
Signaling 180 Ringing 180 Ringing Signaling
(UDP or TCP) 200 OK 200 OK

ACK ACK

Bearer Or
Media (UDP) Media
RTCP Stream

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SIP Servers/Services

SIP
Location Servers/
Registrar Redirect Database Services

“Where is this
name/phone#?”

3xx Redirection
“They moved,
REGISTER SIP Proxy
try this address”
“Here I am”

Proxied INVITE
“I’ll handle it for
INVITE
you”
“I want to talk
to another UA

SIP User
Agents SIP User
Agents SIP-GW

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Deploying SIP Networks
PSTN PSTN
312 212

NY
Chicago
POP
POP
Central Zone East Zone

IP Network

West Zone
SF
POP

PSTN
415

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Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

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Voice Myths

Myths Facts

• Networks can only be • VoIP allows centralized


built one way or distributed
architectures
• Networks will only use
• H.323, SIP, MGCP and
one protocol
H.248/Megaco will all be
• All networks will present in VoIP
converge networks
• Networks will converge
to IP

4426_02_2002_c1 © 2002, Cisco Systems, Inc. All rights reserved. 23


Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

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Interconnecting VoIP Networks

23
H.3 SIP
?

MGCP
H.248
Megaco
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Connecting VoIP to SS7/C7 Networks

IAM
H.225 Setup (ANI,DN)
Proceeding
H.323
H.245

CRCX
ACK
MGCP
SDP

INVITE
SIP ACK
SDP

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VoIP Interworking Issues

• Service interworking
E.g.: H.450 <-> SIP <-> MGCP
• Media interworking
End-to-end codec negotiation
• Bearer interworking
End-to-end fax, modem, DTMF

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VoIP Interworking

• Bearer level • Service translation


Modem (relay/passthru) issues
Fax (relay/passthru) Call deflection

T.38 Park/hold

T.37 • Signal issues


DTMF (relay/passthru) SDP

• Media level H.245

Codec (negotiation,
selection)

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Fax and Modem Passthru Mechanisms

• Modem and fax are control mechanisms based


on PLL (Phase Locked Loops)
• They are both time sensitive
• Highly sensitive to packet network impairments:
Jitter
Packet loss
Delay
• Susceptible to clock slew (clock sync differences
between gateways)

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Passthru Simplified
Voice Gateway

PCM G.711µ
G.711µ DSP

G.729

IP
Cloud Voice Gateway

G.711µ PCM
DSP
G.711µ

G.729

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What Is Modem Passthru?

• It is the transport of modem signals


(modulation, error correction and
compression) through a packet network
using PCM encoded packets

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Modem Passthru (Cont.)

• Modem tone detection (<= V.90)


• Switchover signaling
• No VAD
• EC off
• RTP payload redundancy (10ms
packetization) RFC2198 (optional)

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Modem Passthru Issues

• Consecutive packet drops


(loss) cause retrain
• Consecutive drops during
retrain causes disconnect
• Variation of delay (jitter) has
quite an effect
• Jitter (at 10%) is a
conservative estimate—
Since jitter mostly impacts
performance with packet loss

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What Is Modem Relay?

• Modem relay involves


demodulating the modem
signal at ingress gateway
• Passing this data as packet
data to terminating gateway
• Re-modulating the data and
passes it to the receiving
modem

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Fax Relay—T.38
T.30 UDP T.30

PSTN PSTN

IP

• Real-time
• Also called demod/remod
• Can be used in H.323/MGCP/SIP signaling
• Delivers fax data over UDP streams (uses same RTP port)—
reuses voice UDP ports
• Fallback to proprietary mode
• Method of encoding the T.30 and T.4 into packets
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DTMF

• What is DTMF
• Why is it required?
and where is it used?
• How do you transport
it in IP?
• DTMF implementation

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DTMF (Cont.)

• In TDM world, all voice traffic is sent as


uncompressed 64Kbs PCM streams; anything
sent on that circuit is an untouched stream of
bits; (e.g., voice speech, modem tones, fax
tones, and DTMF digits)
• DSP codecs designed to interpret human
speech, can distort DTMF tones (machine-tones)
• High b/w codecs less likely to distort
• Distortion causes problems with voicemail and
IVR systems

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DTMF Schemes with VoIP Protocols

MGCP, H.248,
H.323 SIP
Megaco

In-Band In-Band In-Band In-Band

Cisco RTP, Cisco RTP,


Out-of- H.245 Alphanum, NSE, NTE,RFC2833 RFC2833
Band H.245 Signal, AVT
Tones RFC2833

4426_02_2002_c1 © 2002, Cisco Systems, Inc. All rights reserved. 38


Topics to Discuss

• History of VoIP
• VoIP—Early Adopters
• VoIP—Standards and Standards Bodies
• VoIP—Making Sense of the Protocols
• “The Great Voice Myth”
• VoIP—Protocol Challenges
• Summary

4426_02_2002_c1 © 2002, Cisco Systems, Inc. All rights reserved. 39


Summary

• Understand the possibilities


and the issues
• Avoid protocol/product
based bias
• Decide on application
• Consider market and business
drivers
• Deploy what’s possible today
• Choose signaling protocol
depending on services intended to
be offered
• Many possibilities—stay tuned

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Crystal Ball on VoIP

• All three protocols (or its variations)


are here for the long run
• Changes/enhancements will be made
• IP will be the core

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Reference URLs

• ITU: www.itu.org
• IETF: www.ietf.org
• SIP: www.cs.columbia.edu/~hgs/sip/
• H.323: www.packetizer.com/iptel/h323/
• MGCP:www.softswitch.org/asp/techlibrary
_protocol.asp?page=techlibrary

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