Understanding Voice over IP Protocols

Understanding Voice over IP

Protocols

Cisco Systems—Service Provider Solutions Engineering February, 2002

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

Why Move to VoIP?

•

Cost savings—toll bypass

•

Open standards—H.323, SIP, MGCP

•

Multi-vendor interoperability

Cisco Packet Voice Architecture

Standards--Based Based

Packet Infrastructure Layer

Packet Infrastructure Layer

(IP, ATM) (IP, ATM)

Open Call Control Layer

Open Call Control Layer

(SIP, H.323, MGCP, etc.) (SIP, H.323, MGCP, etc.) Open Service Open Service Application Layer Application Layer

(JAIN, AIN, TAPI, (JAIN, AIN, TAPI, JTAPI, XML etc.) JTAPI, XML etc.) Open/Standard Interface Open/Standard Interface

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

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

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

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.

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

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

H.324 Terminal H.324 Terminal H.323 Gatekeeper H.323 Gatekeeper Packet Network H.323 Terminal H.323 Terminal H.323 Gateway H.323 Gateway H.323 MCU H.323 MCU Scope of H.323 PSTN ISDN V.70 Terminal V.70 Terminal Speech Terminal Speech Terminal H.320 Terminal H.320 Terminal Speech Terminal Speech Terminal

H.323 Components

Scope of H.323 Recommendation

Media (UDP) RTP Stream RTP Stream RTCP Stream RTCP Stream H.323 Endpoint A H.245 (TCP Dynamic Port) Open Logical Channel

H.225 (TCP Port 1720) Setup

Alerting / Connect

Open Logical Channel Acknowledge Capabilities Exchange / MSD RTP Stream RTP Stream H.323 Endpoint B V V V V

H.323 Signaling

Gatekeeper A Gatekeeper B RRQ/RCF ARQ RRQ/RCF LRQ IP Network Phone A Gateway A _{Gateway B} H.225 (Q.931) Setup

H.225 (Q.931) Alert and Connect H.245 RTP ACF LCF V V

Basic H.323 Call

Midwest Zone Midwest Zone Chicago GW Chicago GW Local PSTN Local PSTN

Deploying H.323 Networks

• Addition of new rate centers • Minimizes GK configuration • Addition of new zones

DGK • Addition of new NPAs

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

MGCP/H.248/Megaco—Architectures

Deploying MGCP/H.248/Megaco

Networks

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

SIP Architecture

SIP Proxy, Registrar & Redirect Servers

SIP VoIP Network Calling Party PSTN Called Party PSTN Signaling Signaling Bearer Or Media Bearer Or Media

SIP Signaling

SIP Servers/Services

SIP Servers/Services

SIP User Agents

Registrar Redirect _DatabaseLocation

SIP Proxy SIP Servers/ Services REGISTER “Here I am” INVITE

“I want to talk to another UA

Proxied INVITE

“I’ll handle it for you” “Where is this

name/phone#?”

3xx Redirection

“They moved, try this address”

SIP User

Central Zone Central Zone Chicago POP Chicago POP West Zone West Zone SF POP SF POP PSTN 312 East Zone East Zone NY POP NY POP PSTN 212 PSTN

Deploying SIP Networks

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

Voice Myths

• Networks can only be built one way

• Networks will only use one protocol

• All networks will converge

• VoIP allows centralized or distributed

architectures

• H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP

networks

• Networks will converge to IP

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

Interconnecting VoIP Networks

?

?

SIP

SIP

H.323

H.323

Connecting VoIP to SS7/C7 Networks

VoIP Interworking Issues

•

Service interworking

E.g.: H.450 <-> SIP <-> MGCP

•

Media interworking

End-to-end codec negotiation

•

Bearer interworking

VoIP Interworking

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)

IP Cloud Voice Gateway Voice Gateway G.711µ G.729 G.729 G.711µ PCM G.711µ PCM G.711µ

Passthru Simplified

What Is Modem Passthru?

•

It is the transport of modem signals

(modulation, error correction and

compression) through a packet network

using PCM encoded packets

Modem Passthru (Cont.)

•

Modem tone detection (<= V.90)

•

Switchover signaling

•

No VAD

•

EC off

•

RTP payload redundancy (10ms

packetization) RFC2198 (optional)

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

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

Fax Relay—T.38

• 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

PSTN PSTN PSTN PSTN IP T.30 UDP T.30

DTMF

•

What is DTMF

•

Why is it required?

and where is it used?

•

How do you transport

it in IP?

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

DTMF Schemes with VoIP Protocols

H.323

H.323 MGCP, H.248,

Megaco

MGCP, H.248,

Megaco SIPSIP

In-Band

In-Band In-BandIn-Band In-BandIn-Band In-BandIn-Band

Out-of-Band Out-of-Band Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833 Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833 Cisco RTP, NSE, NTE,RFC2833 Cisco RTP, NSE, NTE,RFC2833 RFC2833RFC2833

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

• 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

Crystal Ball on VoIP

•

All three protocols (or its variations)

are here for the long run

•

Changes/enhancements will be made

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