ip telephony deploaying voip protocols and ims infrastructure

3GPP Third Generation Partnership ProjectA-BGF Access Border Gateway function AAD Average Acknowledgement Delay AAL2 ATM Adaptation Layer 2 ACD Automatic Call Distribution ACELP Algebrai

Deploying VoIP Protocols and IMS Infrastructure, Second Edition

Olivier Hersent

CEO of Actility

A John Wiley and Sons, Ltd., Publication

Deploying VoIP Protocols and IMS Infrastructure, Second Edition

Deploying VoIP Protocols and IMS Infrastructure, Second Edition

Olivier Hersent

CEO of Actility

A John Wiley and Sons, Ltd., Publication

First edition published 2005

Registered office

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Library of Congress Cataloguing-in-Publication Data

Abbreviations ix

1.1 Transporting voice, fax, and video over a packet network 11.1.1 A Darwinian view of voice transport 11.1.2 Voice and video over IP with RTP and RTCP 5

2.1.3 Relation between H.323 and H.245 versions, H.323 annexes,

2.2.1 The ‘hello world case’: simple voice call from terminal A

2.2.2 A more complex case: calling a public phone from

2.2.4 H.323 calls across multiple zones or administrative domains 86

2.3.6 Using RAS properly and only when required 106

2.5.2 Contacting an email alias with H.323 and the DNS 115

3.1.2 From RFC 3261 to 3GPP, 3GPP2 and TISPAN 166

3.3.2 The proxy function, back to back user agents 230

3.5 Instant messaging (IM) and presence 2543.5.1 Common profile for instant messaging (CPIM) 2553.5.2 RFC 3265, Specific Event Notification 2603.5.3 RFC 3428: SIP extensions for instant messaging 266

4.1.1 Centralized value added services platforms on switched

telephone networks: the ‘tromboning’ issue 269

4.1.3 How VoIP solves the ‘tromboning’ issue The value added

4.1.4 The IMS architecture is ideal for mobile

4.4.3 Summary of SIP extensions required in an IMS network 311

5.2.4 Extensions for phone user interface control 354

6 Advanced Topics: Call Redirection 367

6.1.1 Call transfer, call forward, call deflection 367

6.1.3 Reference network configurations in the PSTN 3716.1.4 Reference network configurations with VoIP 374

6.1.6 VoIP call redirection and call routing 388

7.1 Introduction to Network Address Translation 393

7.2 Workarounds for VoIP when the network cannot be controlled 398

7.2.2 Using port forwarding to solve the wrong media

7.2.4 Other proposals: COMEDIA and TURN 4027.3 Recommended network design for service providers 4047.3.1 Avoid NAT in the customer premises for VoIP 405

3GPP Third Generation Partnership Project

A-BGF Access Border Gateway function

AAD Average Acknowledgement Delay

AAL2 ATM Adaptation Layer 2

ACD Automatic Call Distribution

ACELP Algebraic-Code-Excited Linear-Prediction

ACL Access Control List

ACM Address Complete Message

ADEV Average Delay Deviation

ADPCM Adaptive Differential Pulse Mode ModulationADSL Asymmetric Digital Subscriber Line

AES Advanced Encryption Standard

AGCF Access Gateway Control Function

AMF Access Management Function

AMR Adaptive Multi-Rate

AMR-WB Adaptive Multi-Rate (Wide Band)

AN-GW Access Network Gateway

ANDSF Access Network Discovery and Selection Function

ANSI American National Standard Institute

AoR SIP Address of Record

APDU Application Protocol Data Unit

API Application Programming Interface

ARF Access Relay Function

ASCII American Standard Code for Information Interchange

ASF Application Server Function

ASN-1 Abstract Syntax Notation One

ASP Application Service Provider

ASR Automatic Speech Recognition or Answer Seizure RatioATM Asynchronous Transfer Mode

AVT Audio/Video Transport

B2BUA Back-to-back User Agent

BASIC Beginners’ All-purpose Symbolic Instruction Code

BBERF Bearer Binding and Event Reporting Function

BGCF Breakout Control Gateway Function

BGF Border Gateway Function

BICC Bearer Independent Call Control

BTF Basic Transport Function

C-BGF Core Border Gateway function

CALEA Communication Assistance for Law Enforcement Act

CallID Call Identifier

CBC Cipher Block Chaining

CCF Charging Collector Function

CCIR Consultative Committee for International Radio (ITU)

CDMA Code Division Multiplex Access

CELP Code-excited Linear Prediction

CFU Call Forwarding Unconditional

CIC Circuit Identification Code

CID Conference Identifier

CIF Common Intermediary Format

CLEC Competitive Local Exchange Carrier

CLF Connectivity Session Location and Repository FunctionCLIP Calling Line Identity Presentation

CLIR Calling Line Identity Restriction

CMTS Cable Modem Termination System

CND Customer Network Device

CNG CalliNG; Comfort Noise Generator

CNG Customer Network Gateway

CNGCF CNG Configuration Function

CoIx Connectivity-oriented Interconnection

COMEDIA Connection-oriented Media Transport in SDP

COPS Common Open Policy Service

CPE Customer Premises Equipment

CPG Call Progress (Message)

CPIM Common Profile for Instant Messaging

CPL Call Processing Language

CPN Customer Premises Network

CPU Central Processing Unit

CRC Cyclic Redundancy Check

CRLF Carriage Return and Line Feed

CRV Call Reference Value

CS-ACELP Conjugate Structure, Algebraic Code-Excited Linear PredictionCSRC Contributing Source

CTI Computer Telephony Integration

DCME Digital Circuit Multiplication Equipment

DCS Distributed Call Signaling

DCT Discrete Cosine Transform

DES/CBC Data Encryption Standard, Cipher Block Chaining

DES Data Encryption Standard

DHCP Dynamic Host Configuration Protocol

DiffServ Differentiated Services

DIS Digital Identification Signal

DLSR Delay Since Last Sender Report

DNS Domain Name System

DNSSEC Domain Name System Security Protocol

DOCSIS Data over Cable Service Interface Specification

DSL Digital Subscriber Line

DSMIP Dual Stack Mobile IP

DSP Digital Signal Processor

DSS1 Digital Subscriber Signaling 1

ECF Elementary Control Function

EFF Elementary Forwarding Function

ENUM “Electronic Numbers” Protocol

EPCF Endpoint Configuration Command

ePDG evolved Packet Data Gateway

ETSI European Telecommunications Standardisation InstituteETSI TIPHON ETSI Telephony and Internet Protocol Harmonization Over

NetworksETTB Ethernet to the Building

ETTX Ethernet to the<anything> (Curb, Home, Building)

FEC Forward Error Correction

FIF Fax Information Field

FIFO First in First Out

FIPS PUB Federal Information Processing Standards Publication

GEF Generic Extensibility Framework

GGSN Gateway GPRS Support Node

GSM Global System for Mobile Communications

GTD Global Transparency Descriptor

GTP Generic Tunneling Protocol

HDLC High-level Data Link Control

HLR Home Location Register

HLR/AuC HLR Authentication Center

HSPA High Speed Packet Access

HSS Home Subscriber Server

HTML Hypertext Markup Language

HTTP Hypertext Transfer Protocol

I-BGF Interdomain Border Gateway function

I-CSCF Interrogating Call/Session Control Function

IAD Integrated Access Device

IAM Initial Address Message

IANA Internet Assigned Numbers Authority

IARI IMS Application Reference Identifier (IARI)

IBCF Interconnection Border Control Function

ICID IMS Charging Identifier

ICMP Internet Control Message Protocol

ICSI IMS Communication Service Identifier

IEC ISO International Electrotechnical Commission

IETF Internet Engineering Task Force

IFP Internet Fax Protocol

IFT Internet Fax Transmission protocol

ILS Internet Locator Service (Microsoft)

IMCN IP Multimedia Core Network

IMPI IP Multimedia Private Identity

IMPP Instant Messaging and Presence Protocol

IMPU IP Multimedia Public Identity

IMS IP Multimedia subsystem

IMTC International Multimedia Teleconferencing Consortium

INAP Intelligent Network Application Protocol

IntServ Integrated Services

IOI Inter Operator Identifier

IP Intelligent Peripheral

IP CAN Internet Protocol Connectivity Access Network

IP-PBX Internet Protocol–Private Branch Exchange

IPDC Internet Protocol Device Control

IPR Intellectual Property Rights

IPSec Internet Protocol Security

IRC Internet Relay Chat

IRQ Information Request

IRR Information Request Response

ISDN Integrated Service Digital Network

ISP Internet Service Provider

ISUP ISDN USER PART protocol

ITSP Internet Telephony Service Provider

ITU International Telecommunications Union

IVR Interactive Voice Response

IWF Interworking Function

JFIF JPEG File Interchange Format

JPEG Joint Photographic Experts Group

LCD Liquid Crystal Display

LD-CELP Low-delay, Code-excited Linear Prediction

LDAP Lightweight Directory Access Protocol

LNP Local Number Portability

MEGACO Media Gateway Controller

MGCF Media Gateway Control Function

MGCP Media Gateway Control Protocol

MME Mobility Management Entity

MMS Multimedia Message Service

MMUSIC Multiparty Multimedia Session Control

MP-MLQ Multipulse Maximum Likelihood Quantization

MPEG Moving Picture Experts Group

MPLS Multiprotocol Label Switching

MRFC Media (or Multimedia) Resource Function Controller

MRFP Media (or Multimedia) Resource Function Processor

MTP Message Transfer Part

MTT Minimum Transmission Time

MTU Maximum Transmission Unit

MWI Message Waiting Indication

NACF Network Access Configuration Function

NAPT Network Address and Port Translation

NAPTR Naming Authority Pointer Record

NAS Network Access Server

NASS Network Attachment Subsystem

NAT Network Address Translation

NCS Network Based Call Signaling Protocol

NFE Network Facility Extension

NTP Network Time Protocol

NTSC National Television System Committee

OSP Open Settlement Protocol

P-CSCF Proxy Call/Session Control Function

P-frame Prediction Frame

PAL Phase-alternation-line

PBDF Profile Data Base Function

PBX Private Branch Exchange

PCC Policy and Charging Control

PCEF Policy and Charging Enforcement Function

PCM Pulse Code Modulation

PCMA Pulse Code Modulation A Law

PCMU Pulse Code Modulationµ Law

PCRF Policy and Charging Rule Function

PDF Policy Decision Function

PDN-GW Packet Data Network Gateway

PEP Policy Enforcement Point

PER Packed Encoding Rules

PES PSTN/ISDN Emulation Subsystem

PGR Pages Received (Fax)

PIDF Presence Information Data Format

PIM Protocol-independent Multicast

POSIX Portable Open System Interconnect

POTS Plain Old Telephone Service

PSTN Public Switched Telephone Network

QCIF Quarter CIFV (144∗176)

QoP Quality of Protection

RACF Resource and Admission Control Function

RACS Resource and Admission Control Subsystem

RAI Resource Availability Indicator

RAS Registration, Admission, Status Protocol

RC Reception Report Count

RCEF Resource Control Enforcement Function

RED Random Early Detection

RSA Rivest, Shamir, Adleman (public key algorithm)

RSIP Restart in Progress

RSVP Resource ReserVation Protocol

RTC Return to Command

RTCP Real-time Control Protocol

RTO Retransmission Timeout

RTP/AVT Real Time Protocol under the Audio/Video Profile

RTP Real-time Transport Protocol

RTSP Real-time Streaming Protocol

S-CSCF Serving Call/Session Control Function

S/MIME Secure Multipurpose Internet Mail Extension

SAP Session Announcement Protocol

SBC Session Border Controller

SCN Switched Circuit Network

SCP Service Control Point

SCTP Stream Control Transport Protocol

SDES Source Description RTP Packet

SDL Specification and Description Language

SDP Session Description Protocol

SECAM S´equentiel Couleur `a M´emoire

SGCF Signaling Gateway Control Function

SGCP Simple Gateway Control Protocol

SGF Signaling Gateway Function

simcap Simple Capability (SDP Declaration)

SIMPLE SIP for Instant Messaging and Presence Leveraging ExtensionsSIP Session Initiation Protocol

SIPS Session Initiation Protocol Secure

SLF Subscription Locator Function

SMG Special Mobile Group (of ETSI)

SMS Short Message Service

SMTP Simple Mail Transfer Protocol

SoIx Service-oriented Interconnection

SS-CD Supplementary Service: Call Deflection

SS-CFB Supplementary Service: Call Forwarding on Busy

SS-CFNR Supplementary Service: Call Forwarding on No Reply

SS-CFU Supplementary Service: Call Forwarding UnconditionalSS-DIV All Diversion Supplementary Services

SSF Service Switching Function

SSL Secure Sockets Layer

STP Signaling Transfer Point

STUN Simple Traversal of UDP through Network Address

Translators

TAPI Microsoft Telephony API

TCAP SS-7 Transaction Capabilities

TCF Training Check Function

TCP Transport Control Protocol

TCS Terminal Capability Set

TCS=0 NullCapabilitySet Call Flow in H.323

TDM Time Division Multiplexing

TE Terminal Equipment Unit

TFTP Trivial File Transfer Protocol

TGCF Trunking Gateway Control Function

TIA Telecommunications Industry Association (USA)

TIPHON Telephony and Internet Protocol Harmonization over Networks

(ETSI)TLS Transport Layer Security

TLV Type, Length, Value Format

TPKT Transport Packet (RFC 1006)

TURN Traversal Using Relay NAT

UAAF User Access Authorization Function

UCF Unregistration Confirm

UCS Universal Character Set

UDP User Datagram Protocol

UDPTL UDP Transport Layer

UICC Universal Integrated Circuit Card

UII User Input Indication

UMTS Universal Mobile Telecommunication System

UPSF User Profile Server Function

UPT Universal Personal Telephony

URI Uniform Resource Identifier

URJ Unregistration Reject

URL Uniform Resource Locator

URN Uniform Resource Name

URQ Unregistration Request

USH Universit´e de Sherbrooke

UTF UCS Transformation Format

UTRAN UMTS Terrestrial Radio Access Network

VAD Voice Activity Detector

VASA Value Added Services Alliance

VLAN Virtual Local Area Network

VoIP Voice over Internet Protocol

VPIM Voice Profile for Internet Messaging

VPN Virtual Private Network

VSELP Vector Sum-excited Linear Prediction

WAP Wireless Application Protocol

XML eXtensible Markup Language

XMPP eXtensible Messaging and Presence Protocol

Abstract Syntax Notation-1

(ASN-1)

Defined in ITU standard X.691

Access Control List (ACL) A packet filter on a router

Admission Confirm (ACF) A RAS message defined in H.225.0

Admission Reject (ARJ) A RAS message defined in H.225.0

Admission Request (ARQ) A RAS message defined in H.225.0

Application Protocol Data

Units (APDUs)

See H.450.1

Associate Session A related session Two related sessions must be

synchronized (e.g., an audio session can specify

a video session as being related) The receivingterminal must perform lip synchronization forthose sessions

Backus–Naur Form (BNF) See RFC 2234

Bandwidth Confirm (BCF) A RAS message defined in H.225.0

Bandwidth Reject (BRJ) A RAS message defined in H.225.0

Bandwidth Request (BRQ) A RAS message defined in H.225.0

Basic Encoding Rule (BER) See ASN.1

Call Identifier (Call-ID) A globally unique call identifier

Call Reference Value (CRV) A 2-octet locally unique identifier copied in all

Q.931 messages concerning a particular call (seealso conference identifier)

Conference Identifier (CID) This is not the same as the Q.931 Call Reference

Value (CRV) or the call identifier (CID) TheCID refers to a conference which is the actualcommunication existing between the

participants In the case of a multipartyconference, if a participant joins the conference,leaves, and enters again, the CRV will change,while the CID will remain the same

The Common Intermediary

Format (CIF)

A video format which has been chosen because itcan be sampled relatively easily from both the525-line and 625-line video formats: 352× 288pixels

Contributing Source (CSRC) When an RTP stream is the result of a combination

put together by an RTP mixer of severalcontributing streams, the list of the SSRCs ofeach contributing stream is added in the RTPheader of the resulting stream as CSRCs Theresulting stream has its own SSRC

Disengage Confirm (DCF) A RAS message defined in H.225.0

Disengage Reject (DRJ) A RAS message defined in H.225.0

Disengage Request (DRQ) A RAS message defined in H.225.0

Dynamic Host Configuration

Fast-Connect A procedure to eliminate media delays after the

connection of the call introduced in H.323v2.Another name used for the same procedure isFast-Start

Fast-Start See Fast-Connect

Gatekeeper Confirm (GCF) A RAS message defined in H.225.0

Gatekeeper Request (GRQ) A RAS message defined in H.225.0

Gatekeeper Reject (GRJ) A RAS message defined in H.225.0

Information Request (IRQ) A RAS message defined in H.225.0

Information Request

Response (IRR)

A RAS message defined in H.225.0

Initial Address Message

(IAM)

SS7 ISUP message initiating a call set-up

Inter-mode Refers to a video-coding mode where compression

is achieved by reference to the previous, orsometimes the next, frame

Interactive Voice Response

server (IVR)

A machine accepting DTMF or voice commands,and executing some logic which interacts withthe user using pre-recorded prompts or syntheticvoice

Internet Fax Transmission

(IFT)

A protocol, see ITU recommendation T.38

Internet Relay Chat (IRC) The famous ‘chat’ service of the Internet, based on

a set of servers mirroring text-basedconversations in real-time

Intra-mode Refers to a video-coding mode where compression

is achieved locally (i.e., not relatively to theprevious frame)

IP-PBX Private phone switch with a VoIP wide area

network interface Most IP-PBXs have an H.323WAN interface See also IPBX

IPBX Same as IP-PBX Some use the term IPBX for

private phone switches which use only VoIP(i.e., the phones are also IP phones), whereas anIP-PBX can be a traditional PBX with analogphones and only uses a WAN VoIP interface.See IP-PBX

Jitter Statistical variance of packet interarrival time It is

the smoothed absolute value of the meandeviation in packet-spacing change between thesender and the receiver The smoothing isusually done by averaging on a sliding window

of 16 instantaneous measures

Location Confirm (LCF) A RAS message defined in H.225.0

Location Reject (LRJ) A RAS message defined in H.225.0

Location Request (LRQ) A RAS message defined in H.225.0

macroblock For the H.261 algorithm, a group of four 8∗8

Multipoint Processor (MP) The H.323 entity which processes the media

streams of the conference and does all thenecessary switching, mixing, etc

Naming Authority Pointer

This defines a standard way to format a timestamp,

by writing the number of seconds since 1/1/1900with 32 bits for the integer part and 32 bits forthe decimal part expressed as number of 1/232seconds (e.g., 0x800000000 is 0.5 s) A compactformat also exists with only 16 bits for theinteger part and 16 bits for the decimal part Thefirst 16 digits of the integer part can usually bederived from the current day, the fractional part

is simply truncated to the most significant 16digits

P-frame Prediction frame obtained by motion estimation or

otherwise, and representing only the differencebetween this image and the previous one.Packed Encoding Rules

(PER)

See ASN.1

Payload Type (PT) As defined by RTP

port An abstraction that allows the various destinations

of the packets to be distinguished on the samemachine (e.g., Transport Selectors, or TSELs, inthe OSI model, or IP ports) On the Internet,many applications have been assigned

‘well-known ports’ (e.g., a machine receiving an

IP packet on port 80 using TCP will route it tothe web server)

Prediction frame (P-frame) Obtained by motion estimation or otherwise, and

representing only the difference between thisimage and the previous one

Private Branch Exchange

(PBX)

A private phone switch

Proxy server An intermediary program that acts as both a server

and a client for the purpose of making requests

on behalf of other clients Requests are servicedinternally or by passing them on, possibly aftertranslation, to other servers A proxy interprets,and, if necessary, rewrites a request messagebefore forwarding it

Q-interface Signaling

(QSIG)

Protocol used at the Q-interface between twoswitches in a private network ECMA/ISO havedefined a set of QSIG standards

Real-time Control Protocol

Registration Confirm (RCF) A RAS message defined in H.225.0

Registration Reject (RRJ) A RAS message defined in H.225.0

Registration Request (RRQ) A RAS message defined in H.225.0

Registration, Admission,

and Status (RAS)

The name of the protocol used between thegatekeeper and a terminal, and betweengatekeepers for registration, admission, andstatus purposes Defined in H.225.0

Return To Command (RTC) Six consecutive EOLs instructing a G3 Fax to

return to command mode

Sender Report (SR) Used in RTCP and RTP

Session ID A unique RTP session identifier assigned by the

master The convention is that the value of thesession ID is 1 for a primary audio session, 2 for

a primary video session, and 3 for a primarydata session See Associate session

Single Use Device (SUD) See H.323 annex F

SIP dialog This was defined in RFC 3261 as a peer-to-peer

SIP relationship between two UAs whichpersists for some time A dialog is established

by SIP messages, such as a 2xx response to anINVITE request A dialog is identified by a callidentifier, a local tag, and a remote tag

A dialog was formerly known as a call leg inRFC 2543

SIP final response A SIP response that terminates a SIP transaction

(e.g., 2xx, 3xx, 4xx, 5xx, 6xx responses) SeeSIP provisional response

SIP provisional response A SIP response that does not terminate a SIP

transaction, as opposed to a SIP final response(1xx responses are provisional)

SIP redirect server A redirect server is a server that accepts a SIP

request, maps the address into zero or more newaddresses, and returns these addresses to theclient Unlike a proxy server, it does not initiateits own SIP request Unlike a user agent server,

it does not accept calls

SIP registrar A registrar is a server that accepts REGISTER

requests A registrar is typically co-located with

a proxy or redirect server and may offer locationservices

SIP server A server is an application program that accepts

requests in order to service requests and sendsback responses to those requests Servers areeither proxy, redirect, or user agent servers orregistrars

SIP transaction A SIP transaction occurs between a client and a

server, and comprises all messages from the firstrequest sent from the client to the server up to afinal (non-1xx) response sent from the server tothe client A transaction is identified by theCSeq sequence number within a single-call leg.The ACK request has the same CSeq number asthe corresponding INVITE request, but

comprises a transaction of its own

Stream Control Transport

Talkspurt A period during which a participant usually speaks,

as opposed to silence periods

Time Division Multiplexing

stream between two hosts, but there is nodelimitation of individual messages within thisstream RFC 1006 defines a simple TPKT packetformat to delimit such messages It consists of aversion octet (‘3’), two reserved octets (‘00’),and the total length of the message including theprevious headers (2 octets)

Transport address Combination of a network address (e.g., IP address

10.0.1.2) and port (e.g., IP port 1720) whichidentifies a transport termination point

Transport Control Protocol

(TCP)

The most widely used, reliable transport protocolfor IP networks

Transport Layer Security

Uniform Resource Identifier

(URI)

Defines a uniform syntax and semantic conventionfor any resource The URI is defined in RFC

2396 See also URL, URN

Uniform Resource Locator

A RAS message defined in H.225.0

Unregistration Reject (URJ) A RAS message defined in H.225.0

Unregistration Request

(URQ)

A RAS message defined in H.225.0

User Agent Client (UAC) Also known as a calling user agent A user agent

client is a client application that initiates the SIPrequest

User Agent Server (UAS) Also known as a called user agent A user agent

server is a server application which contacts theuser when a SIP request is received and returns

a response on behalf of the user The responseaccepts, rejects, or redirects the request

User agent A SIP end system participating in a SIP

transaction See UAC, UAS

User Datagram Protocol

(UDP)

The most widely used unreliable transport protocolfor IP networks UDP only guarantees dataintegrity by using a checksum, but anapplication using UDP has to take care of anydata recovery task

Zone An H.323 zone is the set of all H.323 end points,

MCs, MCUs, and gateways managed by a singlegatekeeper

VoIP 1998– 2004, 6 YEARS FROM R&D LABS TO LARGE SCALE DEPLOYMENTS

Since 1998 Voice over IP, in short VoIP, has been the favorite buzzword of the telecomindustry In 1998, IP was not yet as established and dominant as it is today, and mosttelecom engineers still believed that only ATM technology would be able to supportmultimedia applications Indeed at this time most of us experienced the Internet onlythrough dial-up modems and most ISPs, unable to keep-up with the exploding demandfor Internet connections, were providing a level of service that could hardly qualify evenfor ‘best effort’

But even in this context, the R&D teams that started to work on VoIP were not simplytaking a leap of faith Their bet on VoIP was backed by the last developments of packetnetworking theory, which proved that properly designed IP networks could provide anappropriate support for applications requiring quality of service Knowing this, most ofthese teams felt confident that VoIP could be deployed on a wide scale in the future,and in the mean time tried to evaluate what could be the impact of VoIP, compared toprevious technologies

It took a relatively long time to understand the reasons that would lead a service provider

to deploy VoIP instead of traditional switched voice networks Initially VoIP was presented

as a technology that could enable a service provider to transport voice ‘for free’ over theInternet, because IP transport was ‘free’, and calls could be routed to local breakout trunks

on the far end The first commercial applications of VoIP focused on prepaid telephony,which was a reasonable target given that potential buyers of prepaid card systems do careabout costs, and they are much more tolerant to quality of service issues than any othermarket segment VoIP prepaid telephony systems did have a great success—today themajority of international calling card services use VoIP—but not because of cheaper calltermination costs (which are regulated independently of the technology in most countries),

or cheaper transport costs (traditional voice compression systems are much more efficientthan VoIP systems) The reason for the success was mainly because VoIP facilitates thetrading of minutes between multiple networks without the constraint of establishing leasedlines: on the Internet, virtually all VoIP service providers ‘see’ each other and can decide

to exchange traffic immediately, or to stop as soon as better arbitrage opportunities exist

In addition the central switching system of a VoIP service provider does not processvoice streams, but only signaling messages: a call initiated from a gateway in Paris can

be routed to a gateway in London by a VoIP call controller located in NewYork with veryfew overhead costs Only signaling messages make the round trip through the Atlantic,voice packets only cross the Channel

It is now clear that the key reasons for the success of VoIP are:

– location independence: because of the unique characteristics of VoIP call controllers,

or ‘Softswitches’, many functions that previously required multiple distributed points

of presence can now be centralized, reducing administrative overheads and ing deployments

accelerat-– simplification of transport networks: in the example above, service providers no longerneed to establish leased lines dedicated to voice prior to exchanging traffic But theuse of standard IP data networks—configured appropriately—is a major breakthrough

in many other circumstances: core transport networks no longer need to maintain thededicated network that was required by SS7 signaling, enterprises moving to newoffices can save the significant expenses required by dedicated telephony wiring anduse virtual LANs instead

– the ability to establish and control multimedia communications, e.g interactive audioand video calls, data sharing sessions, etc

Because of these unique characteristics, VoIP technology is a very good choice everytime a relatively complex call control function would require multiple points of presenceclose to the end-users in traditional switched technology, and can be centralized with anapplication softswitch:

– In residential telephony, new service providers can deploy centralized VoIP call control

servers and use any IP networking technology For instance FastWeb, in Italy, serves theItalian market from just two PoPs located in Milan and Rome This is not possible withtraditional technology using traditional (TDM) switches (even with V5.2/GR303 ATMgateways used at the edge of the network), because the voice streams need to bephysically switched by the backplane of the TDM switch In addition of course, VoIPtechnology makes it possible to introduce additional media, like video communications,which differentiate the service and help maintain the ARPU1

– Informal, Distributed contact centers also become much easier and cheaper to operate

with VoIP: the centralized call distribution point no longer needs to switch the voice

1 Average Revenue Per User

streams, and therefore tromboning2 through the VoIP call distribution server is pletely eliminated, which reduces communications costs and minimizes the requiredbandwidth for the connection of the call distribution platform

com-– In general, all applications which previously required a complex intelligent network

architecture in order to minimize tromboning (call switching occurs at specific nodes

in the network, and the applications can be located elsewhere), can be significantlysimplified using a centralized call control server which controls voice signaling butoptimizes the voice path through the IP network

Today more and more service providers and enterprises, as they have become confident

in the VoIP technology and quality of service of IP networks, deploy VoIP applications inorder to enjoy the location independence and greater flexibility of the technology Withmore successful deployments, VoIP is gaining in maturity, and the cost of VoIP gatewaysand IP phones is quickly dropping with the increased volumes

SCOPE OF THIS BOOK

In IP Telephony, we will also assume, like the pioneers of VoIP, that it is possible to

carry multimedia data flows over an IP network with an appropriate quality (i.e lowlatency and low packet loss), and we will focus only on the functional aspects of VoIP.Voice coding technology is presented as a ‘black box’, with enough information for anengineer who wants to use an existing coder in an application, but without describ-

ing the technology in detail IP Telephony will be useful mainly in the lab (development

platforms, validation platforms), when designing and troubleshooting new interactive timedia applications

mul-The companion book Beyond VoIP Protocols becomes necessary when you deploy these applications in the field, over a real network with limited capacity Beyond VoIP protocols

contains an overview of the techniques that can be used to provide custom levels of quality

of service for IP data flows, and guidelines to properly dimension an IP network for voice

It also delves into the details of voice coding technology, and the influence of the selectedvoice coder and the transmission channel parameters on perceived voice quality

In theory, it is sufficient to read the VoIP standards in order to become an efficientVoIP engineer Although reading the standards is always necessary at some point, thesedocuments were never written to be read from A to Z Not only the mere volume is aproblem, hundreds of pages for each standard, but also the structure is inappropriate: allVoIP standards are written as umbrella documents, which point explicitly or implicitly

to dozens of other more detailed documents Sometimes, these documents are also leading, because some of the recommended methods were discussed in a specific context

mis-2 ‘Tromboning’ refers to a non-optimal media path through the network, compared to the shortest path It happens when the media streams have to ‘zigzag’ across multiple nodes, reminding of the shape of a bent trombone.

in the standard bodies, but this context was lost or not clearly expressed in the writtenrecommendation (see for instance the issues presented in Chapter 7 for call redirection).Last but not least most standards are the result of ‘diplomatic’ agreements between firms,which often result in multiple alternate ways of doing the same thing, very long andcumbersome documents with many ‘options’ and unclear sentences designed to preservethe agreed compromise, while in practice after a few years, the market forces lead to a

‘de-facto’ standard choice, in general adopted from the practice of the dominant players

We wrote IP Telephony because we believe it is much more efficient to gain first a

general overview on VoIP, and only then go into the details of the standard documents,but only when needed and if clarification is required on a specific item Initially this bookwas designed as an internal training tool within France Telecom, and over the years itdeveloped by capturing the accumulated experience of the authors and their colleagues, inover 50 voice over IP deployments, among which some of the largest VoIP deploymentsworldwide: Orange and its multi-million livebox deployment (well over 50% of theFrench telephony traffic now uses VoIP), FastWeb in Italy, Equant (the world’s largestVoIP virtual private network), etc

IP telephony begins by giving an overview of the techniques that can be used to encode

media streams and transmit them over an IP network (Chapter 1) It focuses on the tional requirement of transmitting an isochronous data stream over an asynchronous net-work which introduces delay variations (“jitter”) The media encoding methods themselvesare presented very briefly, with just enough details for an engineer who wants to use themand understand the main parameters required for the transmission of the resulting data.The most popular VoIP standards are presented in Chapter 2 (H.323), Chapter 3 (SIP),and Chapter 5 (MGCP) In Chapter 4 we describe the IMS (IP multimedia subsystem),which has become the standard architecture for large scale residential VoIP networks(using the TISPAN profile, also described in Chapter 3) and will be the cornerstone offuture LTE deployments These chapters do not intend to fully replace the standards, butprovide a detailed overview that should be sufficient for most engineers and pointers torelevant normative documents if further reference is required The value of these chapterscomes also from the many discussions on aspects of the standards that are still immature,and descriptions of calls flows or protocol extensions commonly used by vendors but notdescribed in standard documents

func-The ‘advanced topics’ chapters (Chapters 6 and 7), discuss two issues faced by allservice providers when deploying public VoIP services (as opposed to custom servicesdesigned for a single enterprise) The first issue comes from the incompatibility of currentVoIP protocols with Network Address Translation routers and firewalls, which changethe addresses of IP packets on the fly but without properly translating the IP addressescontained in the VoIP messages carried by these packets The second issue comes fromthe widespread confusion between private telephony techniques and public telephonytechniques for call transfers In both cases the chapter presents techniques that weredeployed successfully, and explains the pros and cons of each possible method

When the first edition of this book was published, VoIP standards were beginning tomature, at the protocol level VoIP products, which were using totally proprietary protocolsbefore the year 2000, began to interwork first using H.323 and then MGCP and SIP also.Simultaneously, some operators began to deploy huge VoIP residential networks, reachingmillions of users In 2005, most deployments used standard protocols; however, the archi-tectural details of the VoIP networks were still proprietary and specific to each VoIP net-work: network interconnection was possible, but roaming across networks was impossible.The need for a standard architecture became stronger as the size of deployments reachedmassive dimensions: the work of 3GPP on the IP Multimedia Subsystem architectureaimed at defining such a standard architecture This was quite an ambitious and difficultchallenge, but with the help of ETSI TISPAN which complemented the standard with spe-cific functions required in fixed networks, the IMS architecture, in its release 8 (“CommonIMS”), finally reached a level of maturity which makes real deployments possible

In this new edition, we dedicate a full chapter to the IMS architecture, the underlyingtransport network architecture (Enhanced Packet System), and the TISPAN specific addi-tions for fixed networks We continue, however, to present in detail, protocols such asH.323 or MGCP which are not used inside the IMS system, but as peer networks or atthe edges of the IMS network These protocols are still used intensively in existing VoIPnetworks, and are still the best candidates in some situations, e.g., videoconferencing andISDN PBX trunking for H.323, or business IP phone control for MGCP It is likely thatfuture evolutions of SIP and IMS will progressively alleviate the need for other protocols

in VoIP; however, most VoIP operators will still need to support multiple VoIP protocols

in the next 5 to 10 years

We hope that this book will help network engineers to deploy, maintain, or upgradetheir VoIP networks, using each protocol where it fits best, and with full awareness of thepotential pitfalls and difficulties

Multimedia Over Packet

1.1 TRANSPORTING VOICE, FAX, AND VIDEO OVER A PACKET NETWORK

1.1.1 A Darwinian view of voice transport

The most common telephone system on the planet today is still analog, especially at theedge of the network Analog telephony (Figure 1.1) uses the modulation of electric signalsalong a wire to transport voice

Although it is a very old technology, analog transmission has many advantages: it issimple and keeps the end-to-end delay of voice transmission very low because the signalpropagates along the wire almost at the speed of light

It is also inexpensive when there are relatively few users talking at the same time andwhen they are not too far apart But the most basic analogue technology requires onepair of wires per active conversation, which becomes rapidly unpractical and expensive.The first improvement to the basic ‘baseband’ analog technology involved multiplexingseveral conversations on the same wire, using a separate transport frequency for eachsignal But even with this hack, analog telephony has many drawbacks:

• Unless you use manual switchboards, analog switches require a lot of electromechanicalgear, which is expensive to buy and maintain

• Parasitic noise adds up at all stages of the transmission because there is no way todifferentiate the signal from the noise and the signal cannot be cleaned

For all these reasons, most countries today use digital technology for their core telephonenetwork and sometimes even at the edge (e.g., ISDN) In most cases the subscriber lineremains analogue, but the analogue signal is converted to a digital data stream in the

IP Telephony: Deploying VoIP Protocols and IMS Infrastructure, Second Edition O Hersent

 2011 John Wiley & Sons, Ltd

Figure 1.1 Analog telephony, as old as the invention of the telephone, and still in use today

at the edge of the network.

first local exchange Usually, this signal has a bitrate of 64 kbit/s or 56 kbit/s (one sampleevery 125µs)

With this digital technology, many voice channels can easily be multiplexed along the

same transmission line using a technology called time division multiplexing (TDM) In

this technology, the digital data stream which represents a single conversation is dividedinto blocks (usually an octet), and blocks from several conversations are interleaved in around robin fashion in the time slots of the transmission line, as shown in Figure 1.2

Because of digital technology, the noise that is added in the backbone does not influencethe quality of the communication because digital ‘bits’ can be recognized exactly, even inthe presence of significant noise Moreover, digital TDM makes digital switching possible.The switch just needs to copy the contents of one time slot of the incoming transmissionline into another time slot in the outgoing transmission line Therefore, this switchingfunction can be performed by computers.

However, a small delay is now introduced by each switch, because for each conversation

a time slot is only available every Tµs, and in some cases it may be necessary to wait up

to T µs to copy the contents of one time slot into another Since T equals 125 µs in all

digital telephony networks, this is usually negligible and the main delay factor is simplythe propagation time

Unless you really have a point to make, or you’re a politician, you will usually speakonly half of the time during a conversation Since we all need to think a little before wereply, each party usually talks only 35% of the time during an average conversation

If you could press a button each time you talk, then you would send data over thephone line only when you actually say something, not when you are silent In fact, most

of the techniques used to transform your voice into data (known as codecs) now have theability to detect silence With this technique, known as voice activity detection (VAD),instead of transmitting a chunk of data, voice, or silence every 125µs, as done today onTDM networks, you only transmit data when you need to, asynchronously, as illustrated

And when it comes to multiplexing several conversations on a single transmission line,instead of occupying a fraction of bandwidth all the time, ‘your’ bandwidth can be used

by someone else while you are silent This is known as ‘statistical multiplexing’

The main advantage of statistical multiplexing is that it allows the bandwidth to be

used more efficiently, especially when there are many conversations multiplexed on the

same line (see companion book, Beyond VoIP protocols Chapter 5 for more details) But

statistical multiplexing, as the name suggests, introduces uncertainty in the network Asjust mentioned, in the case of TDM a delay of up to 125µs could be introduced at eachswitch; this delay is constant throughout the conversation The situation is totally differentwith statistical multiplexing (Figure 1.4): if the transmission line is empty when you need

to send a chunk of data, it will go through immediately If on the other hand the line isfull, you may have to wait until there is some spare capacity for you

This varying delay is caller jitter, and needs to be corrected by the receiving side.

Otherwise, if the data chunks are played as soon as they are received, the original speechcan become unintelligible (see Figure 1.5)

The next generation telephone networks will use statistical multiplexing, and mix voiceand data along the same transmission lines Several technologies are good candidates(e.g., voice over frame relay, voice over ATM, and, of course, voice over IP)

Jitter and delay

Statistical multiplexer

Bandwidth optimization

‘Hello, How are you today?’

w are you today?’

‘Hello, ho