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VoIP WG Working Document Final version.doc

Working Group on VoIP

Working Document on VoIP – final version

Final Version : 04/06/03



Table of contents

1 Introduction..................................................................................................................................4

1.1 Scope ..................................................................................................................................................... 4

1.2 References............................................................................................................................................. 4

1.3 Abbreviations ....................................................................................................................................... 5

1.4 Definitions............................................................................................................................................. 6

2 VoIP short description .................................................................................................................8

2.1 Interconnection general principles ..................................................................................................... 8

2.2 IP Technology and networks............................................................................................................... 82.2.1 IP Access .......................................................................................................................................................92.2.2 IP Interconnection .........................................................................................................................................92.2.3 TDM Interconnection ..................................................................................................................................10

2.3 Protocols and signalling..................................................................................................................... 10

3 Economic and market prospective.............................................................................................12

3.1 State of VoIP services in European countries ................................................................................. 12

3.2 Actual state in Switzerland ............................................................................................................... 13

3.3 Existing applications.......................................................................................................................... 13

3.4 VoIP in managed networks ............................................................................................................... 13

3.5 VoIP on public Internet..................................................................................................................... 14

3.6 Existing products on the market ...................................................................................................... 14

3.7 Potential economic prospective......................................................................................................... 14

4 Telecommunication legislation..................................................................................................17

4.1 IP voice services as telecommunications services............................................................................ 17

4.2 IP voice services as public telephone services.................................................................................. 184.2.1 Freedom of connection................................................................................................................................184.2.2 Real-time voice transport ............................................................................................................................184.2.3 Ancillary criteria..........................................................................................................................................194.2.4 Rights and obligations .................................................................................................................................19

5 Service description, functions integrated in and associated to VoIP.......................................20

5.1 Listing of most common functionalities ........................................................................................... 20

5.2 Telephony Services Types ................................................................................................................. 205.2.1 Universal Service [US]................................................................................................................................205.2.2 Public Telephony Service [PTS] .................................................................................................................215.2.3 Internet telephony (Voice over Internet) .....................................................................................................22

5.3 Public telephony, the user perception .............................................................................................. 235.3.1 Historical approach......................................................................................................................................235.3.2 Technology neutral approach ......................................................................................................................235.3.3 Minimum requirements for VoIP to meet public telephony specs ..............................................................24

5.4 Interoperability .................................................................................................................................. 245.4.1 General ........................................................................................................................................................245.4.2 Numbering...................................................................................................................................................25

5.4.2.1 E.164 Numbering and Internet Naming..............................................................................................255.4.2.2 Access to subscribers of fixed and mobile services............................................................................255.4.2.3 Access to value-added services...........................................................................................................255.4.2.4 Access to short numbers .....................................................................................................................25



5.4.2.5 Carrier selection..................................................................................................................................265.4.3 Interoperability Scenarios............................................................................................................................26

5.4.3.1 Scenario A - Interconnection between carriers IP-trunking backbone ...............................................265.4.3.2 Scenario B – Interconnection of NextGen Service Providers to PSTN Providers..............................285.4.3.3 Scenario C – Interconnection of regional VoIP Service Providers.....................................................295.4.3.4 Combined Scenarios ...........................................................................................................................295.4.3.5 Icons and associated functionality ......................................................................................................30

6 Quality of Service (QoS) ............................................................................................................31

6.1 Real time notion ................................................................................................................................. 31

6.2 QoS quantification, Mean Opinion Score (MOS) ........................................................................... 316.2.1 Aspects other than voice transmission.........................................................................................................326.2.2 Voice transmission ......................................................................................................................................326.2.3 Specific aspects of compression algorithms and delay................................................................................33

6.2.3.1 Most common compression algorithms..............................................................................................336.2.4 Building the delay budget............................................................................................................................34

7 Conclusions ................................................................................................................................35

Annex A : Network Termination Point (NTP) ............................................................................36

Annex B : "non-voice" QoS parameters ......................................................................................37

Annex C : Compression algorithms and delay ............................................................................38C-1 Multiple Compression Cycles.............................................................................................................38C-2 Basic Voice Flow................................................................................................................................38C-3 How Voice Compression works .........................................................................................................39C-4 Standards For Delay Limits ................................................................................................................39C-5 Sources of Delay.................................................................................................................................39C-6 Coder (Processing) Delay (χn) ............................................................................................................40C-7 Packetization delay (πn)......................................................................................................................40C-8 Serialization delay (σn) .......................................................................................................................40C-9 Queuing/Buffering Delay (βn) ............................................................................................................40C-10 Network Switching Delay (ωn) ......................................................................................................41C-11 De-jitter Delay (∆n) ........................................................................................................................41

Annex D : "Itot".............................................................................................................................42



1 Introduction

Technology and liberalisation have led to many changes in telecommunications during the lastdecade but greater changes are in prospect for the close future.

The Internet Protocol is a common transport for data and voice in the future. The public Internetwith its email and world wide web information services have become part of everyday life althoughfew people had used either ten years ago. The process of moving voice services onto IP is on theway. The VoIP actors in Switzerland have the opportunity to participate in the elaboration oftechnical and administrative regulations facilitating this migration.

1.1 ScopeThis document studies the implementations of Public Telephony Services in Switzerland based onVoIP technologies, including legal, technical, market and regulatory aspects. It may facilitate theintroduction of new offers on the market with such technologies.

The document gives a brief summary of the current state of VoIP-based services in Switzerland. Itwill enable the various members of the VoIP Working Group to have an equal understanding of thetechnologies, the problems and the issues that need to be tackled. This document will help toidentify the key points that need to be discussed in more detail, such as, in particular, "Qualitycriteria for the provision of the public telephone service", "Compliance with legal obligations arisingfrom the provision of the public telephone service" and "IP interconnection interfaces". As a secondpriority, these key points will be discussed in detail in a new document entitled “Final document”,which will be the basis for preparing or amending technical and administrative regulations relatingto VoIP technologies.

Another aim of this document is to identify the main references or standards issued by standardsagencies, fora, the EU, etc.

1.2 ReferencesDefinitions & frame regulatory treatment from EU Directives (E-communication, Universal Service,base definitions

• E-communication 2002/21/EC• Universal Service 2002/22/EC• base definitions of a QoS EG 201 769-1 of a QoS

E-model & applications from ITU-T and ETSI• ITU-T G.107, G.108, G.109, G.1000• EG 201 050, EG 202 086

ETSI documents (TIPHON, SPAN) covering End-to-end QoS and Services• TR 101 329-1:"General aspects of Quality of Service (QoS)";• TS 101 329-2: "Definition of speech Quality of Service (QoS) classes";• TS 101 329-3: "Signalling and control of end-to-end Quality of Service (QoS)";• TS 101 329-5: "Quality of Service (QoS) measurement methodologies";• TR 101 329-6:"Actual measurements of network and terminal characteristics and performance

parameters in TIPHON networks and their influence on voice quality";• TR 101 329-7:"Design guide for elements of a TIPHON connection from an end-to-end speech

transmission performance point of view".• ETSI TS 101 909 documents serie• ETSI TS 101 909-6: "Access and Terminals (AT); Digital Broadband Cable Access to the Public

Telecommunications Network; IP Multimedia Time Critical Services;• ETSI TS 101 909-16: "Digital Broadband Cable Access to the Public Telecommunications Network;

IP Multimedia Time Critical Services;



• ETSI TS 102 141: "Services and Protocols for Advanced Networks (SPAN); MTP/SCCP/SSCOP andSIGTRAN; M2UA [Endorsement of RFC 3331 (2002), modified]".

• ETSI TS 102 142: "Services and Protocols for Advanced Networks (SPAN); MTP/SCCP/SSCOP andSIGTRAN; M3UA; [Endorsement of RFC 3332 (2002), modified]".

• ETSI TS 102 143: "Services and Protocols for Advanced Networks (SPAN); MTP/SCCP/SSCOP andSIGTRAN; SUA, [Endorsement of SIGTRAN-SUA-14 (Dec. 2002), modified]".

IETF document on security for IP• IETF RFC 2401: "Security Architecture for the Internet Protocol".

Security (availability, privacy,…)• Release 3; Abstract Architecture and Reference Points Definition; Network Architecture and

Reference Points".

1.3 AbbreviationsATA Analogue Telephone Adapter

ATM Asynchronous Transfer Mode

BICC Bearer Independent Call Control

CODEC Coder/Decoder

ComCom Federal Communications Commission

DiffServ Differentiated Services

DOCSIS Data Over Cable Service Interface Specification

EC European Commission

ETSI European Telecommunications Standardisation Institute

EU European Union

H.xxx H series of the ITU-T recommendations

IETF Internet Engineering Task Force

INAP Intelligent Network Application Protocol

IP Internet Protocol

ISDN Integrated Services Digital Network

ISP Internet Service Provider

ISUP ISDN User Part

ITSP Internet Telephony Service Provider

ITU International Telecommunication Union

LAN Local Access Network

LTC Law on Telecommunications

Megaco Media Gateway Control Protocol

MGCP Media Gateway Control Protocol

ORAT Decree on Addressing Resources in the Telecommunications Sector

NA(P)T Network Address and Port Translation

NCS Network Call Signaling

NTP Network Termination Point

OSP Open Settlement Protocol



OST Ordonnance sur les Services de Télécommunication (RS 784.101.1)

PBX Private Branch Exchange

POTS Plain Old Telephone Service

PPP Point-to-Point Protocol

PSTN Public Switched Telephone Network

PTS Public Telephony Services

QoS Quality of Service

RSVP Reservation Protocol

RTCP Real Time Control Protocol

RTP Real Time Protocol

SCN Switched Circuit Network

SCTP Stream Control Transport Protocol

SDP Session Description Protocol

SIGTRAN Signalling Transportation

SIP Session Initiation Protocol

SIP-T Session Initiation Protocol for Telephony

SLA Service Level Agreement

SPAN Services and Protocols over Advanced Networks

SS7 Signalling System No. 7

STUN Simple Tunnelling Protocol

TDM Time Division Multiplexing

TIPHON Telecommunications and Internet Protocol Harmonisation Over Network

TSP Telecom Service Provider

VAD Voice Activity Detection

VoIP Voice over IP

VPN Virtual Private Network

xDSL Family of Digital Subscriber Line technologies

1.4 DefinitionsITSP Internet Telephony Service Provider. The provider who offers a VoIP service.

Softswitch, Gatekeeper, Call Agent, Telephony Server: All terms are used interchangeably tospecify an entity in charge to control a call.

A Codec is responsible to code and decode the analogue speech to digital samples. Codecs differin bandwidth requirements, processing power, quality and delay. There exist a lot of differentcodecs. Just to give a few examples:

G.711 64 kbps The most commonly used codec

GSM-EFR 12.2 kbps Enhanced Full Rate: used in today’s GSM networks

G.729a 8 kbps Used in VoIP media gateways

G.723 6.4 kbps Acceptable but not recommended.



VAD Voice activity detection. A method to reduce the required bandwidth independent of the usedcodec. Experiments show that in a conversation only 40% is speech and 60% is silence. So,during this 60% of the time it is not necessary to transmit voice packets.

VoIP VPN-off/on-net A Virtual Private Network can be based on different technologies: forexample Voice over IP. On-net refers to the fact, that traffic is originated and terminatedwithin the Virtual Private Network, whereas an off-net call is only originated in the VPN butterminated in another network.

Definitions in accordance with RS 784.101.113 / 1.4:

Telecommunication network interfaces Physical network termination point (NTP) or serviceaccess point through which users have access to a telecommunications service (see art. 2,para. 1d OIT1) [3].

Network Termination Point (NTP) In accordance with art. 2, para. 1d, OIT, the networktermination point is the connection point through which users obtain access to atelecommunications network which is used in whole or in part to provide telecommunicationsservices. See Annex A.

1 OIT: ordonnance du 14 juin 2002 sur les installations de télécommunication – Decree of 14 June 2002 onTelecommunications Equipment



2 VoIP short description

Generally, VoIP includes all mechanisms to transport voice in a packetized manner over theinternet protocol.

Roughly, two aspects have to be distinguished:

• Interface protocols to setup a connection, to authenticate and authorise the usage of theservice, to route and charge a call.

• Interface protocols how to code and packetize the voice and how to transport it over an IPnetwork with an acceptable quality.

2.1 Interconnection general principlesAn interconnection agreement is established between two operators or service providers. In caseof difficulties, OFCOM will moderate the negotiation to achieve the best compromise and facilitatea solution.

OFCOM has published and, based on acquired experience, will revise when appropriate thecatalogue of recommended interfaces for interconnection between the various telecommunicationoperators and services providers. The catalogue can be found on:(http://www.bakom.ch/en/telekommunikation/interkonnektion/ag_technik/unterseite13/index.html)

Two particular telecommunication services providers may agree to set out any additional oralternative requirements not included in the catalogue.

2.2 IP Technology and networksThe three interfaces an ITSP is confronted with are

1) IP access: UNI = User-Network-Interface. All protocols and interfaces to connect an enduser terminal to the ITSP

2) IP interconnection: NNI = Network-Network-Interface. All protocols, interfaces and SLAs tointerconnect two ITSPs.

3) TDM interconnection: The (known) interconnection between SCN and ITSP.

Figure 2-1: The 3 interfaces an ITSP is confronted with



2.2.1 IP Access

This subject will not be treated in the present Working Document as there are no interconnectioninterfaces available now, no major impairment is expected and standardisation work is not matureenough.

2.2.2 IP Interconnection

The following slide shows a possible architecture of an IP interconnection.

A few functions of the Border Gate:

• Protect the ITSP against attacks

• Allows/blocks RTP-streams under control of the Softswitch

• ...

Figure 2-2: IP interconnection

The Access cloud is the Access provider (e.g. Swisscom or Cablecom) who manages the wire tothe end user (copper /coax/fibre/...).

The ISP cloud is the Internet Access Provider (e.g. Bluewin or Swissonline) who is in charge inproviding an IP address (Layer 3) to the end user. He is responsible to connect the end user to theworld wide Internet.

The ITSP cloud is the Internet telephone provider. He is in charge in providing a telephonic ID tothe end user (E.164 or name)

Logically, the three functions (Access/ISP/ITSP) are separate. It is possible that certain functionsare offered by the same provider.

The interconnection agreement is in charge of authentication/authorisation/billing of the VoIP trafficbetween the ITSPs. A possible means to guarantee the agreement is the OSP protocol. OSPprovides you with a authorisation Key. This key is inserted in the call handling messages (SIP /H.323) to prove that the traffic is covered by the agreement. See also section 5.4.3.1.

The transport SLA guarantees the end-to-end quality of the VoIP call. A possible means toguarantee the SLA, is a call-by-call bandwidth reservation between A and B by a protocol calledRSVP. This transport SLA is between an ITSP and the underlying intermediate IP transportprovider (ISP). See also section 5.4.3.3.



2.2.3 TDM Interconnection

OFCOM currently recommends the ISUP v3 i signalling protocol (see the specific characteristics inthe catalogue of recommended interfaces for interconnection on the OFCOM web site athttp://www.ofcom.ch/en/telekommunikation/interkonnektion/ag_technik/unterseite13/index.html).The next version of this catalogue may include alternative or complementary additionalrecommendations. See also section 5.4.3.2.

2.3 Protocols and signallingIn this alphabetical list, all lower layer transport protocols are excluded. “Lower layer” means allprotocols below IP (Ethernet, ATM, DOCSIS, xDSL, MPLS, PPP, IPSec…). Partly, these lowerlayers offer features relevant to quality.

BICC Bearer Independent Call Control. Standardised by ITU-T. Modifications andenhancements to ISUP: The setup of the connection and the setup of the speechpath are separated.

COPS Common Open Policy Service. Standardised by IETF. Protocol between an entity thatenforces QoS (policy client, e.g. BorderGate) and an entity that controls QoS (policyserver, e.g. Softswitch).

DiffServ Mechanism to tag IP headers for prioritised handling in routers. Standardised byIETF. Allows a simple form of QoS in IP networks.

H.248 Standardised by ITU-T. See Megaco.

H.323 The first VoIP framework. Standardised by ITU-T. A set of relevant standards to beused to implement VoIP. Registration/Authorisation: H.225.0; Signaling: Q.931; MediaTransport: RTP; Codec: G.711. Components of the H.323 architecture include:Terminal, Gatekeeper, Gateway.

ISUP ISDN User Part. Standardised by ITU-T. Signaling protocol on top of SS7 to setupand release connections and to transport supplementary service and charging relatedinfo. Addressing used: E.164.

MGCP Media Gateway Control Protocol. Standardised by IETF.

Megaco Media Gateway Control Protocol. Standardised by IETF. A common approach of ITU-T and IETF to standardise the protocol between softswitch and media gateway. Sameas H.248

NA(P)T Network Address and Port Translation. Standardised by IETF. A mechanism in arouter to map IP addresses to other IP addresses. Used to map private IP addressesto public IP addresses to circumvent the IPv4 address shortage. NAT is one of themain hindrances in public VoIP deployments. See STUN.

NCS Network call signaling. Standardised by CableLabs. An MGCP based spin-off used inCable networks.

OSP Open Settlement Protocol. Standardised by ETSI. Protocol to handle inter-networkauthentication and usage recording independent of used call setup protocol (H.323,SIP)

RSVP Resource Reservation Setup Protocol. Standardised by IETF. Protocol to guaranteeEnd to End quality by reservation of resources along the path from A to B.

RTCP Real Time Control Protocol. Standardised by IETF. Protocol to control RTP.Transports timestamps and other QoS related values. Monitors jitter and delay toinfluence the packetisation and coding mechanisms of RTP.



RTP Real Time Protocol. Standardised by IETF. Protocol to transport real time information(voice, audio, video…) in a packetized manner over an IP network. Standardizes thecodecs and packetisation schemes used per media content (PCM, JPEG, MPEG…).

SCTP Stream Control Transport Protocol. Standardised by IETF. Protocol used bySIGTRAN.

SDP Session Description Protocol. Standardised by IETF. Description of the connectionSIP wants to establish. Describes used codecs, endpoint’s addresses. Comparable toBearer Capability and Channel ID in DSS1 or Transmission Medium Requirementand Circuit Identification Code in ISUP.

SIGTRAN Signaling Transportation. Standardised by IETF. A generalized mechanism totransport traditional telephony protocols on top of IP. It consists of SCTP and anadaptation layer per protocol to be transported. Some examples of differentadaptation layers: M2UA, M2PA, M3UA, IUA, SUA.

SIP Session Initiation Protocol. Standardised by IETF. Protocol to setup a connectionbetween two IP terminals. For media information SDP is used. Components of theSIP architectures include: Registrar, Redirector, Proxy.

SIP-T Session Initiation Protocol for Telephony. Standardised by IETF. Additions to SIP totransport unmapped ISUP parameters transparently between softswitches.

SS7 Signaling System No.7. Standardised by ITU-T/ETSI. Network architecture to deliverpackets from A to B. Comparable to IP. Addressing used: SPC (signaling pointcodes).

STUN Simple tunnelling protocol. Standardised by IETF. Mechanism to allow detection andtraversal of NAT devices between endpoint and softswitch.



3 Economic and market prospective

Traditional means of transporting voice required the use of circuit-switched (also known as time-division multiplexing or TDM) equipment.

VoIP promises better line utilisation and lower network control costs for ITSP. It promises majorbenefits for corporate customers in simplifying network structures and in introducing new computertelephony integration. VoIP will pave the way for the introduction of a whole new generation ofsmart devices.

Applications for VoIP include LAN Telephony, IP Enabled/Converged PBX, Local & Long DistanceTransport, Calling Card Services, and Subscriber (referred to as Class 5) and Transit (referred toas Class 4) Central Office (CO) Switch augmentation and replacement.

3.1 State of VoIP services in European countriesAs VoIP is not explicitly regulated in the new EC framework, each European country has its ownapproach to VoIP in term of regulation, depending of the development of this technology.

Some examples are given thereafter as to illustrate the different approach of VoIP services inEurope.

In Austria the incumbent has launched a trial concept for a service called "personal number".Based on IP-oriented transmission techniques, a called party with a E.164 phone number shouldalways be reachable independently from the telecom appliance (fixed/mobile terminal, PC) usedand from where the party would actually stay. The customer would configure the service via awebsite; interaction between SS7 and IP networks would be via gateways. The incumbent plannedto offer this trial service only within a closed user group which is not subject to licensing. However,this service has not been offered to the public yet. In January 2003 the incumbent announced a co-operation with an US-based VoIP carrier to establish an interconnection to the IP network of thecarrier, enabling them to route international traffic to more than 90 countries using VoIP. PTS overVoIP would be subject to voice telephony licensing requirements if managed VoIP traffic forms partof the core network of a PTS operator (who normally already has got a license).

In Germany the incumbent DTAG offers VoIP services for business clients and uses thistechnology for traffic transport between its switched circuit network and international networks. Theregulatory authority RegTP will not regulate VoIP services due to the real time problematic.

In Denmark VoIP services are not officially offered on the market. In general IP telephony over theInternet is not covered by the Danish Telecom regulation. The communication in such a case isconsidered as content. Other forms of IP telephony, which would be a telecommunications servicecan be provided under the Danish telecom regulation without any licence requirements.Nevertheless, the provision of all kinds of telecommunications services for end users must follow anumber of consumer protection related rules.

Estonia has 31 registered VoIP service providers registered. VoIP is regarded as telephone serviceand general requirements from law and decrees extend to VoIP.

Only France conducted a public call for comments on VoIP, in 1999. The regulatory analysis whichwas established then is still valid today. IP technologies are used to transport voice by operatorsbut there is not an end-to-end VoIP product available for residential. France Telecom e.g. offerseConf, a multimedia (VoIP) solution to end users, and calls to Portugal and Senegal via a VoIPbased network. Some VoIP products are available for businesses but generally limited to theirinternal network.

In Finland the regulation is technology neutral, meaning that same rules apply both to VoIP andany other technology.



PST based on VoIP is currently available in Portugal. Under the licensing regime, the figures of"Voice on the Internet" and "VoIP" have been defined. Based on the description of the serviceoffered, "Voice on the Internet" or "VoIP" may fulfil the requisites, defined in national regulations,for the Fixed Telephone Service (FTS).

In Sweden VoIP services are offered to companies and residential subscribers by differentoperators.

In Italy Telecom Italia already uses VoIP in their National and International Transit. Fastwebprovides voice services over Ethernet and ADSL for more than 100’000 subscribers.

Tele2 are providing a Carrier Select services based on VoIP in multiple European countriesincluding Spain and France, enabling low cost telephony for residential and business users via aTele2 carrier select dialling code. This is one of the largest pan-European softswitch deployments.

Nextra interconnects Germany, Czech and Slovakia via an international VoIP network.

The Netherlands offer calls to the Middle East via a VoIP based network.

Many National and International ISPs offer additional PC-to-Phone services.

3.2 Actual state in SwitzerlandToday, Swisscom doesn’t offer Public Voice Services based on VoIP Technology. Several VoIPbased services are available for Business customers only. Swisscom Mobile uses SS7 Signalingtransport over IP internally. The public operator assisted directory service offered by Swisscom isbased on VoIP technology.

Cablecom launched the first VoIP based Public Voice Service in Switzerland “digital phone” inFebruary 2003.

Plans to offer combinations of Data, Voice and Video Services over IP in the next 1 or 2 years existfor many new and traditional operators in Switzerland. Different Broadband Access Technologieslike Cable, xDSL, Ethernet and Wireless will be used to offer IP connectivity to the subscribers.

3.3 Existing applicationsIn addition to Basic Telephony Services, applications like IP based Centrex, IP Contact Center,Voice Mail, Fax, Unified Communication, Conferencing and Callback Services can be offered withVoIP technology.

3.4 VoIP in managed networksIn some cases the benefits of basic and new VoIP Telephony applications on IP transport servicesmade the VoIP Technology successful for managed, hosted and privately owned Business VoiceServices. The VoIP Technology allows also Business Critical usage. Critical factors for VoIP basedTelephony Services are Voice Prioritisation, IP Queuing, Call Admission Control, Redundancy,Echo cancellation and optimised Voice Codecs.

Increasingly popular Data VPNs give users guaranteed access and bandwidth over the publicnetwork and allow carriers to manage private network traffic more easily as a portion of theirbroader networks. They allow users and carriers to negotiate service level agreements (SLAs) toensure that certain ranges of service quality are maintained. The growth of the VPN market willcertainly help push the convergence of voice and data networks. Corporate end users need onlytry out the technology to realise that there is a world of difference in quality of service betweenVoIP over the public Internet and VoIP over a managed VPN. In some instances, VoIP over VPNscan be compared to cellular phone quality. This quality of service level can be offered by usingnetwork management tools to ensure a consistent level of bandwidth for voice traffic over the VPN.



This is known as offering a SLA. On the other hand, quality of service over the public Internetcannot be guaranteed, leading to packet loss and poor sound quality.

3.5 VoIP on public InternetVoIP is being used as a technology to enable PC-to-PC, PC-to-Phone and Phone-to-Phonecommunication with the Public Internet as a Transport Network. Internet phone is attractingattention because international phone can be free and independent of time duration. The potentialto significantly reduce the cost of long distance charges is however strongly reduced by the poorVoice Quality. This is the price to pay when calls become free.

See section 5.2.3.

3.6 Existing products on the marketProducts range from Softswitch, IP PBX, Application Servers, Voice Gateways, SignalingGateways, Residential Gateways to IP Phones and Softphones for Service Provider and Enterpriseapplications.

Figure 3-1: Existing products on the market

3.7 Potential economic prospectiveFirst of all, it is important to point out that any forecast, and this holds in particular true foreconomic forecast, is uncertain. Therefore the forecasts themselves, carried out by reputableanalyst and consultants, differ over a wide array (see Figure 3-2).



Figure 3-2: Forecasts of the market share of telephony on IP of global international voice traffic

Not only that the forecasts differ widely, but it turned out that forecasts made in the past were toooptimistic. Whereas Analysys, a major Telecom analyst, predicted a market share of about 25% forglobal international voice traffic only a couple of years ago, a report published in May 2002 byAnalysys says, that “Voice on IP is already being used to carry an estimated 6% of internationaltraffic on routes where competition is limited.” 2 It is obvious, that the forecasts have been revisedand in consequence lowered.

Nonetheless, it is possible to list the key economic drivers determining future market evolution:

• Capital and operational expenditure compared to PSTN

- For the time being, it is still controversial whether VoIP is cheaper. Economies ofscale have yet to be achieved.

- Network costs depend on quality and complexity, VoIP is not a mature technology,problems still arise when systems are installed

• How companies / residential users perceive the level of security? Basically VoIP is exposedto the same sort of attacks and problems as all IP based application3. According toComputerworld, this threat explains why most companies today hesitate to migrate fromPSTN to a IP based telecommunication network.

Despite the uncertainty about forecasts in general, there is hardly any doubt, that VoIP will becomemore and more important. Hence, the growing traffic over IP has an impact on the carriers’revenue stream. It has to be borne in mind that figure 3-3 depicts only the development forcorporate voice revenue as the discussion is mostly related to corporate customers. There isbarely any information available about residential users and the corresponding evolution.

2 Analysys, May 2002, http://research.analysys.com/default.asp?mode=lcw6.3 Computerworld: Über IP lässt sich reden, Januar 2003, Seite 42.



Figure 3-3: Total corporate fixed voice revenue for Western Europe2

Figure 3-3 shows that Virtual Private Networks (VPN) will become more and more important forcorporate clients. VPN based on VoIP (depicted in green and blue) will have a small sharecompared to VPN based on other technologies than VoIP. Corporate VoIP-based revenues areexpected to reach at least EUR 2.5 billion in 2007 (see Figure 3-3). They could be as much asEUR 7.4 billion, if the benefits of VoIP are demonstrated. The revenue stream will be dominated byVoIP VPN off-net calls, because the principal attraction of installing a VPN for the customer is thelarge discounts on internal calls that can be achieved even between distant or international sites.

The cannibalisation of PSTN revenues by VoIP-based services is inevitable, but carriers have anopportunity to compensate for this loss by developing value-added services such as hosted VoIPsystems and conferencing services. However, to succeed in this area they will have to competewith other players active within the corporate network, particularly the systems integrators.



4 Telecommunication legislation

Swiss legislation on telecommunications is technologically neutral and does not contain anyspecific definitions referring to voice transmission over IP. Regulation of such voice transmissionservices is therefore based on the existing legislation on telecommunications. Accordingly, it isimportant to first verify whether a voice transmission service over IP is actually to be classified as atelecommunications service.

4.1 IP voice services as telecommunications servicesAccording to article 3b of the LTC (Law on Telecommunications), a telecommunications service isunderstood as “transmission of information for third parties using telecommunications technology”.Consequently, three preconditions must be fulfilled cumulatively:

a) “transmission using telecommunications technology”, i.e. electrical, magnetic, optical orother electromagnetic transmission or reception of information over lines or by radio (article3 lit. c LTC);

b) “of information”, i.e. signs, signals, letters, images, sounds or representations of any kindfor people, other life forms or machines (article 3 lit. a LTC);

c) “for third parties”, i.e. not for oneself (one’s own consumption), but for other legal entities ornatural persons. No such third-party relations exist within one and the same company,between parent companies and subsidiaries or within a group (article 2 lit. c, Decree onTelecommunications Services of 6 October 1997 (OST)). However, in so far as suchbusiness structures or other business relationships or user groups are pursuing theexclusive or predominant aim of evading the obligation to register or obtain a licence, athird-party relationship is to be assumed.

If these prior conditions are fulfilled, a voice transmission service over IP is considered as atelecommunications service and the provider is therefore and in particular:

• subject to registration or licensing and subject to the statutory administrative and licensingfees (article 4 in conjunction with article 38ff LTC);

• entitled to interconnection with regard to the market-dominant telecommunication servicesproviders or providers of services forming part of the universal service (article 11 para. 1and 2 LTC);

• obliged to guarantee interconnection, if it is market-dominant for the service in question(article 11 para. 1 LTC);

• entitled to demand access to the subscriber directories of the providers of services formingpart of the universal service;

• entitled to allocation of addressing resources;

• obliged to comply with confidentiality of telecommunications (art. 43 LTC) and to ensurelegal interception (art. 44 LTC) according to the Federal Law of 6 October 2000 on thesurveillance of postal and telecommunications services (RS 780.1) and the Decree of31 October 2001 on the surveillance of postal and telecommunications services (RS780.11);

• obliged to assist in overcoming extraordinary situations (article 47 LTC);

• entitled to claim rights of expropriation and co-use (article 36 LTC);

• obliged to provide information (article 59 LTC).



4.2 IP voice services as public telephone servicesIn a second step, the question is posed as to whether a voice transmission service over IP canalso be considered as a public telephony service, which is defined in article 16 para. 1a LTC asfollows:

“transmission of speech in real time by means of telecommunications techniques, includingtransmission of data employing transfer rates compatible with the channels for transmittingspeech.”

The message of the Federal Council of 10 June 1996 relating to the revised Law onTelecommunications states the following in this context:

The term telephone service means real-time voice transmission including the transmission of data,such as takes place, for example, in today’s analogue fixed network with a modem over the voicechannel, without restriction of freedom of connection, from one network connection point to otherdomestic or foreign network connection points using the user’s own subscriber equipment (voiceterminal, modem or fax). The term "real time" relates to voice transmission and means that it is amatter of a simultaneous, bi-directional communication. Insignificant delays caused bytransmission paths, switching equipment or air interfaces are irrelevant in this context. In the law,technical development requires if possible a non-time-related, technology-independent definition ofthe telephone service, which guarantees the user the scope of the services (voice, data and faxtransmission), as currently available in the public telephone network.

The two most important elements of the definition of the public telephone service according to art.16 para. 1 lit. a LTC are accordingly freedom of connection and real time.

4.2.1 Freedom of connection

In Switzerland, freedom of connection is to be interpreted in such a way that voice transportbetween any domestic or foreign network termination points must be possible; a networktermination point is associated to subscriber numbers from the national numbering plan – inSwitzerland based on the E.164 ITU numbering plan. IP voice transmission services can certainlybe designed so that freedom of connection is guaranteed. In the case of constellations ofconnections in which network termination points are located in the traditional PSTN network,experience is already available regarding the use of gateways between this access network and acore network based on IP. In the case of end-to-end IP network infrastructures, in which the entireconnection between subscriber equipment is made using IP packet switching, the problem of“translation” of (dynamic or static) IP addresses into subscriber numbers from the nationalnumbering plan can also be solved technically.

4.2.2 Real-time voice transport

According to the Federal Council’s message concerning the revised Law on Telecommunicationsthe term “real time” relates to simultaneous, bi-directional communication. Insignificant delayscaused by transport paths, switching equipment or air interfaces are irrelevant in this context.

When it comes to the European Union, the European Commission lays down the following: "Realtime applies when internet telephony is able to provide the same reliability and speech quality asthe public telephone service over the PSTN” (communication of 10.01.1998 on the status of voicetransport over the internet). This stance was confirmed in principle in a corresponding secondcommunication from the European Commission of 22.12.2000.

On the basis of the Federal Council’s message and taking into account the communications of theEuropean Commission, the term real time can therefore be construed as follows:

“Transport of speech in real time must permit simultaneous communication in both directions withminimal time differences, offering a level of reliability and speech quality identical to the public



telephone service offered on the PSTN. Reliability and speech quality does not exceed fixedvalues.”

One of the main tasks of the “VoIP” working group is now to specify and try to quantify thisconcept of real time using appropriate criteria. In this respect it would be most appropriatefor it to use as a basis, as far as possible, existing European and international references.

4.2.3 Ancillary criteria

The conclusion that a voice transmission service is to be considered as a public telephony serviceaccording to article 16, para. 1a of the LTC may be reinforced by one or more of the followingancillary criteria:

• a service is marketed as a substitute for the traditional public telephone service of the PSTNtype;

• the service appears to the subscriber as a substitute for the PSTN public telephone service;• the service represents the sole means for the subscriber of accessing the PSTN public

telephone service;• the service clearly does not represent a supplementary service or a secondary service to an

existing PSTN public telephone service.

4.2.4 Rights and obligations

If an IP voice transmission service can be assessed as a public telephone service, additional rightsand obligations must be taken into account in comparison with “simple” telecommunicationservices providers.

The additional rights include:1. the entitlement to demand interoperability (art. 11 para. 2 LTC);

2. the entitlement to receive "carrier selection codes" (art. 10 para. 1 of the ComCom decreeon the Law on Telecommunications) and to be selected as an alternative provider fornational and international calls;

3. the entitlement to benefit from number portability as a "new provider" (art. 3 of the ComComdecree on the Law on Telecommunications).

The additional obligations include:1. the obligation to ensure access to emergency call services (art. 20 LTC);2. the obligation to allow access to the directories of its subscribers (art. 21 para. 2 LTC);3. the obligation to offer subscribers the possibility of selecting a provider for national and

international calls (art. 9 para. 1 of the ComCom decree on the Law onTelecommunications);

4. the obligation to ensure interoperability between all users of services forming part of theuniversal service (art. 11 para. 2 LTC);

5. the obligation to guarantee number portability between telecommunication servicesproviders (art. 3 of the ComCom decree on the Law on Telecommunications).



5 Service description, functions integrated in and associated to VoIP

This chapter should be understood as a bridge between chapter 4 (more legal aspects) andtechnical specifications of the most common market offers and requests.

5.1 Listing of most common functionalitiesTR 101 973 offers a listing of most common functionalities for POTS, ISDN terminals andassociated functions and functionalities offered in the European market.

The following sections in the present chapter 5 discuss functionalities associated with TelephonyServices. This is a base to analyse the different components of the service that may determine thelevel of the user’s satisfaction.

5.2 Telephony Services Types

5.2.1 Universal Service [US]

In Switzerland, the functionalities to be offered within the scope of the universal service are definedmainly in articles 19 to 27 of the Decree on telecommunication services (OST).

Connection (art. 19, para. 1a, and art. 20 OST):

• provision of a network termination point enabling subscribers to make and receive, in realtime, national and international telephone calls plus fax communications and datacommunications, at reasonable data transmission rates in order to permit internet access;

• entries in the directories of subscribers to the public telephone service (art. 29, para. 2,OST);

• provision of services forming part of the universal service by means of an analogue ordigital fixed interface (ISDN or equivalent).

Additional services (art. 19, para. 1b OST):

• information on malicious calls;• call forwarding;• suppression of calling line identification;• advice of charge;• itemised billing;• bars on outgoing calls.

Emergency calls (art. 19, para. 1c OST):

• call routing to the competent alarm centres (numbers 112, 117, 118, 143, 144,147);• data needed to identify the place of origin (location) of the call.

Directories (art. 19, para. 1d OST):

• access in the three official languages, in return for payment, to the subscriber directories ofall the providers of services forming part of the universal service in Switzerland, where theuser is free to choose between electronic access or access via an information service.

Public payphones (art. 19, para. 1e OST):

• the provision, 24 hours a day, of a sufficient number of public payphones to enable nationalcalls to be made and received in real time, international calls to be made in real time and to



provide access to the emergency call services and to the subscriber directories of allproviders of services forming part of the universal service in Switzerland, in the three officiallanguages.

Transcription service for the hearing impaired (art. 19, para. 1f OST):

• provision of a transcription service for the hearing impaired, including emergency calls, 24hours a day.

Directory and communication service for the visually impaired (art. 19, para. 1g OST):

• access, in the form of an information service in the three official languages, to thesubscriber directories of all providers of services forming part of the universal service inSwitzerland;

• provision of a communication service for the visually impaired.

Building entry point (art. 21, para. 1, OST):

• provision of the telecommunications equipment needed to provide the universal service tothe building entry point (excluding domestic installations).

Bars on outgoing calls (art. 23 OST)

Quality of service (art. 25 OST) concerning:

• connection (art. 25, para. 1a OST);• voice communication (art. 25, para. 1b OST);• data and fax communications (art. 25, para. 1c OST);• other obligations (art. 25, para. 1d OST).

Upper price limits (art. 26 OST):

• Upper price limits apply to connections, communications, public payphones, transcriptionservices, etc.

Unpaid bills (art. 27 OST):

• Rules are drawn up for dealing with unpaid bills.

Other obligations: see item 5.2.2 below.

Note: The use of VoIP technologies should not be impairment for a service provider to obtain anauthorisation from OFCOM to provide US.

The list above is not exhaustive and contains only the most important services andfunctionalities in order to facilitate the creation of a similar set to be used by networkoperators or service providers using VoIP technologies and not applying to the US provisionauthorisation in a first phase.

5.2.2 Public Telephony Service [PTS]

The following obligations (art. 28 – 32 OST) apply both to the universal service licensee and thetelecommunication services providers (SCNs) offering the services forming part of the universalservice:

Emergency calls (art. 28 OST):

• Access to emergency call services (numbers 112, 117, 118, 143, 144 and 147) must beguaranteed from any telephone connection, including public payphones. Access tonumbers 112, 117, 118 and 144 must be free of charge and must be possible without using



any form of payment method (coins or cards). A fixed fee of CHF 0.20 and the supplementin accordance with art. 26, para. 1c can be charged for numbers 143 and 147;

• Providers of mobile telecommunication services by satellite which form part of the universalservice to which addressing resources are allocated by the InternationalTelecommunication Union need only guarantee free access to number 112;

• Where the chosen technology allows this, on-line call identification (location) must beguaranteed for numbers 112, 117, 118 and 144. It must also be guaranteed for subscriberswho have chosen not to appear in a public directory (art. 21, para. 3, Law onTelecommunications (LTC)); If requested, the office can designate other numbers for theexclusive use of the emergency services (police, fire service, ambulance service andrescue service) and call identification (location) must be guaranteed for these services. Itshall publish the list of these numbers;

• The universal service licensee, in conjunction with the other providers of services formingpart of the universal service, provides alarm centres with a service permitting theidentification (location) of all users of services forming part of the universal service. Thisservice, which is remunerated, must also be accessible to the alarm centres which are notconnected to the universal service licensee. Collaboration between the latter and the otherproviders of services forming part of the universal service is governed by the principles ofcost alignment in the sense of art. 45. If there are several universal service licensees, theawarding authority may require one of them to provide the identification (location) service.

Directories (art. 29, OST):

• This article requires, in particular, any telecommunication services provider permittingaccess to the public telephone service by means of a connection identified by an E.164number to hold a directory of its subscribers to the public telephone service provided thesehave given their consent to appear in this directory.

Services for the hearing and visually impaired (art. 30 OST):

• The services mentioned in art. 19, para. 1f and g of the OST must be free of charge and theproviders of services forming part of the universal service must themselves offer these tothe hearing and visually impaired or give the latter access to the services of third-parties;

• The communications prices invoiced to the hearing and visually impaired in the context ofthese services must not be discriminatory compared with the tariffs applied to other users.

Bars on outgoing calls to services with erotic or pornographic content (art. 31 OST):

• The providers of services forming part of the universal service must offer, free of charge,the option of barring outgoing communications to services with erotic or pornographiccontent.

Advice of charge (art. 32 OST):

• The office can draw up technical and administrative regulations on the transmissionbetween the providers of services forming part of the universal service of informationneeded to show the charges to the user (advice of charge).

5.2.3 Internet telephony (Voice over Internet)

Internet telephony, called sometimes "PC-PC" telephony, uses the Internet naming system(user@domain) and is used mainly for services to informal groups and communities. Theparticularity of Internet telephony is to use IP data networks to transport end-to-end traffic with the"best-effort" strategy, the one that is widely used today within the Internet. A network device treatsuser packets as they arrive and congestion over communication routes may lead to packet loss ordelay. The consequence is that the level of availability, the level of quality and the universality ofthe called number currently cannot be guaranteed.

The Internet telephony is out of the present scope and will not be treated in the Working Group.



5.3 Public telephony, the user perceptionTelephony Services should be considered from end-to-end, i.e. from the mouth of one user to theear of another user linked by the Service. For the purpose of the present section, since theterminals market is ruled by other principles and the home installations are also not covered by themost relevant legislation in the present document, the focus will be put on aspects related tonetworks, i.e. from one NTP to another NTP (see clarifications on NTP in Annex A).

Apart the legal definition associated to the rights and obligations specified in chapter 4, the user’sperception of the PTS is associated with

• the number of functionalities referred to in section 5.2.2

• the level of quality (QoS) that this document refers to in chapter 6.

The word public refers to the offer of the service, which is addressing any consumer. The wordpublic is also due to the fact that any consumer may, using the appropriate address number, dialand contact another individual connected to PTS.

Telephony is associated with the fact that the transport of the messages was dimensioned for a3 kHz (0.3 … 3.4 kHz) frequency band, which was the range of signals considered in the earlydays of the telephony essential for the conversational speech functionality.

The “real time” concept is not explicit in the PTS designation, but was introduced by the law todistinguish services like “voice mail” or “push-to-talk” from those supporting conversational speechfunctionality.

5.3.1 Historical approach

From the History point of view the most relevant impairments of the PTS were probably the VoiceTelephony transmission Impairments. A short list with the commonly identified as majorimpairments for the most used technologies are listed below:

Technologies Impairments

PSTN analogue Tx “POTS” Long/bad lines attenuation, X-talk, etc

PSTN digital/ ISDN 1st step digital: “no-loss Tx”, delay & echo

Packet-switch, IP Digital + packet loss, variable path, variable delay/jitter, variableecho, new codecs…

Any type of Voice Tx? All the above

With the increasing complexity of the technologies many other parameters and not only for voicetelephony transmission are relevant, e.g. the introduction of “follow me” facilities may restrict thepossibility of reaching all the network users or may require new numbers to be dialled in order toreach the same partner.

5.3.2 Technology neutral approach

To reach this goal it is proposed to use as far as possible international reference documents notapplying to specific technologies or applying to a large number of known technologies. Mainreferences, sources proposed are, apart Swiss legal documents:

– Definitions & frame regulatory treatment from EU Directives (E-communication, UniversalService, base definitions

§ E-communication 2002/21/EC

§ Universal Service 2002/22/EC

§ base definitions of a QoS EG 201 769-1



– E-model & applications from ITU-T and ETSI

§ ITU-T G.107, G.108, G.109, G.1000

§ EG 201 050, EG 202 086

– Other information, Tx plan, market needs, from ITU-T and ETSI

§ EG 201 377-2, G.101, G.175 (transmission plan)

§ TR 101 963 (IPCablecom, market needs)

5.3.3 Minimum requirements for VoIP to meet public telephony specs

The PTS should be considered from User to User:

• Voice service connects 2 users over Network

• PTS aspects shall be considered from user to user

• User facilities are under user’s own control, will not be discussed in detail in this document

• To treat the PTS from NTP to NTP is the goal of the working group

How to share responsibilities in case of interconnection is one of the most relevant items in thedocument

5.4 Interoperability

5.4.1 General

According to art. 11 para. 2 LTC, the provider of services forming part of the universal service mustguarantee the capacity for communications between the users of these services and is thereforealso obliged to offer interconnection, even if he does not hold a dominant position in the market oris not the universal service licensee. The aim of this rule was expressed in these terms at the timeof the drafting of the LTC: It ensures that all the customers of a provider can always communicatefully with all the customers of another provider. In other words, those wanting to offer theircustomers a public telephone service must also allow them to have access to all customersinvolved in the public telephone service, irrespective of the providers to which they are subscribers.From the point of view of the customer, it is virtually only necessary to have a single networkpermitting each person to communicate with all others. Not only subscribers to fixed and mobileservices should be regarded as customers, but also service providers via service identificationnumbers (08XX/090X) or via short numbers (1XX), who are also allocated individual numbersdirectly by OFCOM.

To date interconnection arrangements and standards have addressed circuit switched to circuitswitched interconnection. With the introduction of IP technology it will be necessary to cover IP-IPinterconnection in equivalent detail. An important issue will be how to extend the current peeringarrangements to cover guaranteed QoS, new addressing schemes and to provide appropriatecarrier-to-carrier billing. Where networks that need to interconnect use different technologies, oneof these operators (circuit switched or IP) will need to provide the necessary inter-workingfunctions. Inter-working functions also may be needed between IP-based networks if thosenetworks use different protocols, e.g. one uses SIP and the other H.323



5.4.2 Numbering

5.4.2.1 E.164 Numbering and Internet Naming"Internet named telephony" (service that supports conversational voice and uses Internet namesfor the identification of the called party) is not considered except representation is in the ENUMURL-format (RFC 2916 by which an E.164 number can be expressed as a fully qualified domainname in "e164.arpa").

The reason for this is that in order to fulfil the obligation of the universal service, particularly withrespect to freedom of connection, VoIP subscribers need to be allocated an E.164 number to usein parallel with their Internet name. Furthermore, this facilitates the migration from SCN providersto NextGen VoIP providers (no impact on numbering).

There is no legal requirement for additional provisioning to enable interworking between suchInternet named telephony services and the traditional public telephony service.

Associations between E.164 numbers already allocated for the public telephony service and theInternet name used for VoIP subscribers can be provided by using ENUM-based applications.

5.4.2.2 Access to subscribers of fixed and mobile servicesIt follows from Article 11, para.2, LTC, that all subscribers to the public telephone service must beable to communicate with all others, including mobile telephony service subscribers.

However, it should be noted that as radio-messaging is not a service forming part of the universalservice (no voice transmission in real time), access to numbers from blocks dependent on theprefix 074 does not, in principle, have to be guaranteed (no interoperability obligation according toArticle 11, para. 2, LTC). Pagers, however, are often used in emergencies. Access to thesenumbers from any telecommunications service provider would therefore be desirable. Furthermore,the caller uses the telephone network to send his message.

5.4.2.3 Access to value-added servicesAccording to Article 43, para. 1 d, OST, access to 08xx and 09xx value-added services forms partof the basic supply as regards interconnection and must therefore be guaranteed by all providersof services forming part of the universal service (Art. 48 a, OST).

The main problem arising for this type of number is due to their special charging scale and the factthat the value-added service provider receives some of the charges. However, this must not be anobstacle to the principle of interoperability. It is incumbent on telecommunications servicesproviders to find solutions to this problem in their interconnection agreements. In the event ofdisputes regarding this subject, ComCom shall be brought in to arbitrate.

5.4.2.4 Access to short numbersAccording to Article 25, para. 1, ORAT (Decree on Addressing Resources in theTelecommunications Sector), OFCOM assigns a short number for one of the services mentioned inArticles 28 to 31a (emergency services, rescue services, breakdown and social welfare services,security information services, mass information services, directory information services), oncondition that they are accessible and provided at all times, in the three official languages(German, French, Italian) and throughout the whole of Switzerland. Access to the servicenevertheless depends to a great extent on the providers of the public telephone service. Like allother numbers, a short number must in fact be integrated into the network of a telecommunicationsservices provider. If the call is generated from a third party network, it must be routed to the hostnetwork and interoperability must be guaranteed. In case of short numbers used for the provisionof emergency services (origin dependent services), Article 28, para. 4, OST, obliges the universal



service licensee to operate, in co-operation with the other universal service providers and for thebenefit of the Alarm Headquarters, a call-tracing service for all universal service subscribers.

The solution differs more for short numbers that can be used without formal allocation (Art. 32ORAT). If the obligations of interoperability apply to numbers used to provide access to subscriberdirectory information services (111, 1151, 1152, 1153, 1154 and 1159), this is not necessarily thecase for numbers that can be used for internal administrative requirements (171xx and 176xx) andfor supporting the clientele (113, 1141, 1144 and 175). The first must in fact not be made publicknowledge (Art. 32, para. 2, 2nd sentence, ORAT), whereas the second primarily concerns accessproviders (service to clientele, manual switching and maintenance engineers). For the latter,nevertheless, one cannot exclude a priori the possibility of expecting corresponding services fromanother provider by means of the call-by-call procedure with a free choice of provider (calls arefiltered in the case of preselection; see below under carrier selection). As regards the number 1145(switching and information services for the blind and partially-sighted), even though it is part of theclientele support services category, it is mentioned explicitly as a service forming part of theuniversal service in Article 19, para. 1 g, OST and therefore indisputably falls under the obligationfor interoperability according to Article 11, para. 2, LTC. In other words, the providers of the publictelephone service must allow their customers access to short numbers that can be used withoutformal allocation (as far as they can be brought to the attention of the public), whether they providethe relevant services themselves or route calls to the provider offering such services based on aninterconnection agreement.

5.4.2.5 Carrier selectionThe preceding remarks are understood to be without the utilisation by the caller of a free choice ofprovider procedure (call-by-call or preselection). Appendix 2 to the ComCom decree lays downcases where calls are effectively routed to the selected provider and those where they are filteredby the access provider. Whether they are filtered or not, the principle remains, however, the same:calls must be routed to destination numbers relevant to interoperable services. The obligation forinteroperability therefore not only affects access providers, but also retransmission providers.

5.4.3 Interoperability Scenarios

There are several scenarios under which voice is carried on IP networks – often implying differenttreatment from a policy or regularity perspective. With regard to IP-interconnection the scenariosoutlined may pose different requirements for settlement between carriers involved. Numbering forexample may be fully compatible with existing regulatory directives for some scenario (E.164numbering) while in other scenarios it may be one of the main issues to be resolved.

IP-packetised voice carried solely across the public Internet is not addressed any further forreasons as outlined in section 4.2 of this document.

5.4.3.1 Scenario A - Interconnection between carriers IP-trunking backboneThis scenario is also referred to as the SCN to IP to SCN scenario or long distance bypass Carrierinterconnect their SCN’s through IP, either directly or via intermediate IP backbone providers. Amotivation to do so may be that existing carriers migrate their SCN-transit network to a managed(QoS aware) IP-backbone. The SCN access network (local exchange infrastructure) as well asend-user equipment are unchanged. (Phone-to-Phone service) Signaling and network intelligenceremain within SCN-switches (IP used as underlaying transport technology).



Scenario A Interconnection between carriers IP-trunking backbone

managed IP backbonemanaged IP backbone

IP-Interconnect

SCN(LE layer)

Carrier 1 Carrier 2

SCN(LE layer)

Figure 5-1: Scenario A



5.4.3.2 Scenario B – Interconnection of NextGen Service Providers to PSTNProviders

This scenario is also referred to as the IP to SCN respectively SCN to IP scenario and may covertwo options:

• Option 1 - Phone-to-Phone serviceThe NextGen provider’s (ITSP) target is support of legacy voice (use of existing phones).The term NextGen refers rather to the technology chosen for network infrastructure (voiceover cable, voice over DSL) and not to the service

• Option 2 – PC-to-Phone serviceIn this option, the NextGen provider (ITSP) sets focus on NextGen voice as a new type ofservice using VoIP CPE’s. (PC-clients, IP-phones).

Scenario BInterconnection of NextGen Service Provider to PSTN Providers

IP-Interconnect

NextGen Net SCN

Carrier 1 Carrier 2

TDM-Interconnect

NextGen Net

Option 2

Option 2

Option 1

Option 1

SCN

Figure 5-2: Scenario B

Note:IP-Interconnect for Scenario B is shown only for the purpose of completeness.This scenario is not intended for further elaboration.



5.4.3.3 Scenario C – Interconnection of regional VoIP Service Providers

This scenario is referred to as the IP to IP scenario. Regional network providers offering VoIP PC-to-PC telephony are favourably interconnected through an IP-backbone provider. In this scenario,telephony may be considered as end-to-end VoIP service, using new “soft-switch” technology tomanage network call control.

Scenario CInterconnection of regional VoIP Service Providers

IP-Interconnect

NextGen NetRegion x

SCN

Regional VoIP Service Provider Regional VoIP Service Provider

IP-Interconnect

NextGen NetRegion y

IP Backbone Provider

managed IPbackbone

Figure 5-3: Scenario C

Note: Interconnection of regional VoIP service providers through TDM-Interconnect, commonlyreferred to as the IP to SCN to IP scenario is not addressed because it is not considered aspriority scenario (limited market relevance).

5.4.3.4 Combined ScenariosIt is obvious, that scenarios are likely to be mixed.For example

• a regional VoIP service provider that interconnects to other regional VoIP networks throughan IP backbone provider (scenario C) will certainly have to support capabilities for calls toPSTN providers (scenario B) or

• a carrier supporting IP-trunking for phone-to-phone services (scenario A) may at a laterstage extend it’s offering to include VoIP terminals (scenario B)



5.4.3.5 Icons and associated functionalityFigure 5-4 represents icons and their functionality used in sections 5.4.3.1 to 5.4.3.4

Figure 5-4: Icons and associated functionality



6 Quality of Service (QoS)

The sections in the chapter 5 discussed functionalities associated with Telephony Services. This isa base to analyse the different components of the service that may determine the level of theuser’s satisfaction.

The Quality of Service is the level of satisfaction the users of a Service enjoy, having as referencetheir expectances.

The sections of the present chapter discuss relevant items for the QoS of Telephony Service. It isassumed that the QoS planned is at the same level

• for the overall service

• and for each one of the relevant functionalities and components associated to that service.

This means that, e.g. in a case where the planned QoS for a PTS is 3.0 MOS, it is assumed thatthe overall voice transmission quality and all other components of that service are planned to fulfilindividually a QoS of at least 3.0 MOS.

The obligation to respect a specific QoS level (MOS ≥ 3.6 for 95 % of connections, see RS784.101.113/1.2) is only applicable to the provision of universal service (art. 25 OST).

6.1 Real time notionSwiss law refers to the Real time notion, but this is, strictly speaking a past, technical limitedconcept, today enhanced by the user satisfaction evaluations, i.e. QoS.

Delay is still a digital voice transmission major issue, but other parameters need to be included inQoS for voice transmission evaluation and as referred to above voice transmission performance isnot the only component of the PTS to be evaluated.

6.2 QoS quantification, Mean Opinion Score (MOS)The Mean Opinion Score is the most common method to quantify the QoS in Telecom services. Itis based in statistical tools and aims to represent the user perception as follows:

Table 6-1: Mean Opinion Score

Quality Score (MOS)

Excellent 5

Good 4

Fair 3

Poor 2

Bad 1

Initially the QoS was only quantified “a posteriori”, since the user could only express the opinionafter using the service.

A number of studies were made in the mean time to establish a reasonable relationship betweenMOS and the physical parameters determining that user’s perception quantification (MOS). Thisallows at the present, during the planning/ dimensioning of the network parameters to foresee withreasonable accuracy the user’s opinion.

In this context the “E-model” as described in ITU-T Recommendation G.107 and ETR 250establish with reasonable accuracy a relationship between physical parameters and MOS in the



area of voice transmission. EG 201 050 implements these studies for the purpose of networkplanning.

For purposes other than voice transmission quality EG 201 769-01, edition 1.1.1, offers a wideguidance, referred to by the CEU in the chapter II of the List of standards 2002/C 331/04, published in2002.12.31.

6.2.1 Aspects other than voice transmission

As underlined above not only real time and voice transmission performance need to be quantified.

Parameters like the time the user needs to set up a call or unsuccessful call rates (congestions orother reasons) and others listed in Annex B "non-voice QoS parameters" are of the highestrelevance for the user evaluation and should be quantified in terms of MOS. At the present, themost relevant of these parameters are studied in EG 201 769-1 where the quantification of QoS isspecified.

Also legal obligations like number portability might be associated with levels of performanceperceived by the user. These obligations are nevertheless commonly seen in a "binary" basis, i.e.the obligation is whether fulfilled or not. This document will not propose a quantification method,since a simple qualification is enough.

The Question 1 group will analyse in further detail the functionalities and legal requirementsapplicable to voice services and will try to establish more detailed guidance on how to quantifyQoS.

6.2.2 Voice transmission

EG 201 050 using some essential ITU-T recommendations in this area, elaborates on therelationship between user satisfaction and some technical parameters of central importance as ameans for network planning. A careful study of this document is recommended.

Below are copied tables of the above-mentioned document section 6.2.

Table 6-2: Relation between Communication Quality and Total Impairment Value Itot (G.113)

ItotUpper limit

MOS GoB PoW Speech Communication Quality

5 4,32 96,6 ∼0 Very Good10 4,17 93,5 ∼0 Good20 3,79 81,3 3,4 Adequate30 3,32 60,5 11,4 Limiting Case45 2,54 25,1 39,5 Exceptional Limiting Case55 2,03 9,7 64,0 Customers likely to react strongly

For further progress on technical studies, it is assumed that for a wide acceptance of the PTS, aQoS of 3.0 MOS or lower is not recommended. This corresponds to approximately a totalimpairment factor [Itot] value of 36.

Note: this value needs to be better studied and justified (question 1 and later) and may be changedaccordingly.

Also in section 11.6 of EG 201 050, the following picture is shown:



high quality

mediumquality

low quality

0 10 20 30 40 50

linear quality scale

TotalImpairmentvalue Itot

Area notpermitted

Figure 6-1: user’s judgement of a Connection on a linear Quality Scale

Based on this value, graphics and explanations in chapter 7 of EG 201 050 help to understand therelationship between several impairments linked to well-defined physical parameters and users’acceptance.

A number of default and “permitted” range values is identified in the table 6 of section 9.7 ofEG 201 050 (copied in Annex D "Itot"). In this context, “permitted” range should be understood as arange of acceptable values, i.e. it is strongly recommended not to go beyond that range. There is interms of PTS at present no legal mandatory provision.

6.2.3 Specific aspects of compression algorithms and delay

The present section offers a first analysis overview of most relevant parameters associated withcompression algorithms and delay which are among the most relevant items to determine the QoSin VoIP based systems.

A more detailed discussion is available in Annex C “Compression algorithms and delay”.

6.2.3.1 Most common compression algorithmsOverall voice quality is a function of many factors including the compression algorithm, errors andframe loss, echo cancellation, and delay.

The following table lists common compression algorithms, their nominal bandwidths and MOSscore.

Table 6-3: Compression Methods and Their Respective MOS Scores

Compression Method Bit Rate(kbps)

MOS Score

G.711 PCM 64 4.1

G.726 ADPCM 32 3.85

G.728 LD-CELP 16 3.61

G.729 CS-ACELP 8 3.92

G.729 x2 Encodings 8 3.27

G.729 x3 Encodings 8 2.68

G.729a CS-ACELP 8 3.7

G.723.1 MP-MLQ 6.3 3.9

G.723.1 ACELP 5.3 3.65

The ‘Nominal bandwidth’ is how much bandwidth the voice stream requires if it were on the wire byitself, without packet headers or flags. Technically, PCM is according to figure C-2 (see Annex C)



an encoding method and not a compression algorithm. As noted in the previous section, thecompression algorithms require PCM streams as the input format.

6.2.4 Building the delay budget

According to above discussions and references documents, the generally-accepted limit for good-quality voice connection delay is 200 ms one-way (or 250 ms as a limit). A PTS may go lower inQoS as explained above. As delays rise over this figure, talkers and listeners becomeunsynchronised, and often they speak at the same time, or both wait for the other to speak. This iscondition is commonly called, “talker overlap.” While the overall voice quality is acceptable, usersmay find the stilted nature of the conversation unacceptably annoying. Talker overlap may beobserved on international telephone calls which travel over satellite connections (satellite delay isin the order of 500 ms, 250 ms up and 250 ms down).



7 Conclusions

This document enabled the members of the working group to better determine the problemsrelating to VoIP technologies and to adopt the same approach and understanding.

The first clear conclusion is that regulation is technology-independent and therefore legalmeasures applicable to VoIP are essentially the same as those for other technologies.

It was particularly apparent that users of such services expect a reasonable level of quality ofservice, which will be discussed in Q.1 (QoS).

As the legal requirements regarding the quality of the service offered do not currently exist forservices forming part of the universal service, one of the next tasks of the working group will be todefine acceptable QoS criteria for VoIP technologies and to suggest how to evaluate these in orderto prepare possible technical and administrative regulations. The technical feasibility with regard tothe requirements outlined in chapter 4 of this document (Telecommunication legislation) for apublic VoIP service will also be evaluated.

The working group will thus have the possibility of devising and proposing solutions in co-operationwith OFCOM.



Annex A : Network Termination Point (NTP)

Switzerland has implemented the European Union’s R&TTE Directive (Directive 1999/5/EC of theEuropean Parliament and of the Council of 9 March 1999 on radio equipment andtelecommunications terminal equipment and the mutual recognition of their conformity).

The aim of these provisions is to obtain prior notification and publication of the technicalspecifications of the network interfaces through which telecommunication services are transmitted.The specifications for this must be sufficiently detailed to permit the development of terminalequipment which is capable of being used for all the services provided by the correspondinginterfaces.

For this purpose, OFCOM has published technical and administrative regulations covering thenotification of telecommunication network interfaces (RS 784.101.113 / 1.4) which, as part ofliberalisation, must help support the development of the telecommunications market and permit themanufacture of terminals that can be used for all the services provided through the correspondingaccess point.

Key to diagram:

Interface (NTP) (NTP) Interface

Terminal Terminal

Installation domestique Domestic installation

Raccordement Connection

Réseau d’accès Access network

Réseau de base Base network



Annex B : "non-voice" QoS parameters

The following parameters were selected from EG 201 769-01 and are at the present considered asthe most relevant for the quantification of QoS. Definitions, measurement methods and furtherconsiderations are available in EG 201 769-01.

• Supply time for initial connection

• Faults rate per access line

• Fault repair time

• Unsuccessful call ratio

• Call set up time

• Response times for operator services

• Response times for directory enquiry services

• Proportion of card and coin operated public pay-telephones in working order

• Bill correctness complaints



Annex C : Compression algorithms and delay

C-1 Multiple Compression CyclesThe CS-ACELP compression algorithms are not deterministic, this means the input data streamisn’t exactly the same as the output data stream. A small amount of distortion is introduced witheach compression cycle as shown in Figure C-1.

One ACELPCompression &Decompression

Cycle

Original Signal

flowInput

Close Approximation+ Small Distortion

Output

Figure C-1: Compression Effects

Consequently, multiple CS-ACELP compression cycles quickly introduce significant levels ofdistortion. This additive distortion effect is not as pronounced with ADPCM algorithms.

Voice quality is subjective, but most users find two compression cycles still provide adequate voicequality. A third compression cycle usually results in noticeable degradation, which may beunacceptable to some users.

C-2 Basic Voice FlowThe flow of a compressed voice circuit is shown below. The analogue signal from the telephone isdigitised into PCM signals by the voice CODEC. The PCM samples are then passed to thecompression algorithm that compresses the voice into a packet format for transmission across theWAN. On the far side of the cloud the exact same functions are performed in reverse order. Theentire flow is shown in Figure C-2.

Telephone Telephone

CodecAnalog to PCM

Conversion

CodecPCM to Analog

Conversion

CompressionAlgorithm

PCM to Frame

De-compression

Algorithm

Frame to PCM

WAN

Flow

Figure C-2: End-to-End Voice Flow

Depending on how the network is configured, the gateway can perform both the CODEC andcompression functions or only one of them. For example, if an analogue voice system is used, thenthe gateway performs the CODEC function and the compression function as shown in Figure C-3.

Telephone

CodecAnalog to PCM

Conversion


PCM to Frame

WAN

Router

Flow

V

Figure C-3: CODEC Function in Gateway



If instead, a digital PBX is used, the PBX performs the CODEC function, and the Gateway justprocesses the PCM samples passed to it by the PBX. An example is shown in Figure C-4.

Telephone

CodecAnalog to PCM

Conversion


PCM to Frame

WAN

RouterPBX

Flow

V

Figure C-4: CODEC Function in PBX

C-3 How Voice Compression worksThe high complexity compression algorithms used in a gateways work by analysing a block ofPCM samples delivered by the Voice CODEC. These blocks vary in length depending on thecoder.

C-4 Standards For Delay LimitsThe ITU considers network delay for voice applications in Recommendation G.114. Thisrecommendation defines three bands of one-way delay as show in Table C-1.

Table C-1: Delay Specifications

Range inmilliseconds

Description

0-150 Acceptable for most user applications.

150- 400 Acceptable provided that administrators are aware of thetransmission time and it’s impact on the transmission qualityof user applications.

Above 400 Unacceptable for general network planning purposes,however, it is recognised that in some exceptional cases thislimit will be exceeded.

Note that these recommendations are for “connections with echo adequately controlled”, whichimplies that echo cancellers are used. Echo cancellers are required when one-way delay exceeds25 ms (G.131).

These recommendations are oriented for national telecom administrations (PTTs), and thereforeare more stringent than would normally be applied in private voice networks. When the locationand business needs of end users are well known to the network designer, more delay may proveacceptable. For private networks 200 ms of delay is a reasonable goal and 250 ms a limit, but allnetworks should be engineered such that the maximum expected voice connection delay is knownand minimised.

C-5 Sources of DelayThere are two distinct types of delay: fixed and variable.



• Fixed delay components add directly to the overall delay on the connection.

• Variable delays arise from queuing delays in the egress trunk buffers on the serial portconnected to the WAN. These buffers create variable delays, called jitter, across thenetwork. Variable delays are handled via the de-jitter buffer at the receiving gateway. Thede-jitter buffer is described in Section C-11.

Figure C-5 identifies all the fixed and variable delay sources in the network. Each source isdescribed in detail in the following sections.

Fixed:SwitchDelay

β 2 β 3 β 4

ω 1ω 2 ω 3

E1E1

64 Kbps

64 Kbps

Variable:Output

QueuingDelay

β 1

Fixed:De-JitterBuffer

∆ 1

Packet Flow

Fixed:Packetization

Delayπ 1

Fixed:Serialization

Delayσ 1

GatewayGateway

σ 2

Fixed:CoderDelay

χ 1

VV

Figure C-5: Delay Sources

C-6 Coder (Processing) Delay (χn)Also called processing delay, coder delay is the time taken by the DSP to compress a block ofPCM samples. Because different coders work in different ways, this delay varies with the voicecoder used and processor speed. For example, ACELP algorithms work by analysing a 10 msblock of PCM samples, plus and then compressing them.

C-7 Packetization delay (πn)Packetization delay is the time taken to fill a packet payload with encoded/compressed speech.This delay is a function of the sample block size required by the vocoder and the number of blocksplaced in a single frame. Packetization delay may also be called Accumulation delay, as the voicesamples accumulate in a buffer before being released.

C-8 Serialization delay (σn)Serialisation delay is the fixed delay required to clock a voice or data frame onto the networkinterface, and it is directly related to the clock rate on the trunk.

C-9 Queuing/Buffering Delay (βn)After the compressed voice payload is built, a header is added and the frame is queued fortransmission on the network connection. Because voice should have absolute priority in thegateway, a voice frame must only wait for either a data frame already playing out, or for other voiceframes ahead of it. Essentially the voice frame is waiting for the serialisation delay of anypreceding frames in the output queue. Queuing delay is a variable delay and is dependent on thetrunk speed and the state of the queue. Clearly there are random elements associated with thequeuing delay.



C-10 Network Switching Delay (ωn)The public frame relay or ATM network interconnecting the endpoint locations is the source of thelargest delays for voice connections. These delays are also the most difficult to quantify.

If wide-area connectivity is provided by Networking equipment, it is possible to identify theindividual components of delay. In general, the fixed components are from propagation delays onthe trunks within the network, and variable delays are from queuing delays clocking frames intoand out of intermediate switches. To estimate propagation delay, a popular estimate of 10 micro-seconds/mile or 6 micro-seconds/km (G.114) is widely used, although intermediate multiplexingequipment, backhauling, microwave links, and other factors found in carrier networks create manyexceptions.

C-11 De-jitter Delay (∆n)Because speech is a constant bit-rate service, the jitter from all the variable delays must beremoved before the signal leaves the network. In gateways this is accomplished with a de-jitterbuffer at the far-end (receiving) gateway. The de-jitter buffer transforms the variable delay into afixed delay, by holding the first sample received for a period of time before playing it out. Thisholding period is known as the initial play out delay.

Voice FramesFrom Network

Voice FramesFrom Network

Over Flow:Queue fills if voice

frame arrive too fast

VariableArrival Rate

= Codec FrameRate +/- ∆

V VV VV V V VV V VVV V VV V VV

Fixed PlayoutRate

= Codec FrameRate

Under Flow:Queue empties if voice frame arrive

too slow

Voice Framesto DSP decode

Voice Framesto DSP decode

Quiescent Point: Specified in mSec

Normal OperatingRange

V V V V V

Figure C-6: De-Jitter Buffer Operation

The optimum initial play out delay for the de-jitter buffer is equal to the total variable delay alongthe connection, this is shown in Figure C-7.

β 2 β 3 β 4

E1E1

64 Kbps

64 Kbps

Variable:Output

QueuingDelay

β 1

De-JitterBuffer

∆ 1

Β = (β 1+β 2+β 3+β 4)

Packet Flow

GatewayGateway

∆ 1 = 1.5(Β )

VV

Figure C-7: Variable Delay and the De-Jitter Buffer



Annex D : "Itot"

The table below offers, as explained in section 6.2.2, a relationship between the most relevantvoice transmission parameters determining QoS and recommended values.

Table D-1: Default Values and permitted Ranges for the Parameters

Parameter Abbr. Unit DefaultValue

permitted range Note

Send Loudness Rating SLRS dB +7 0 to +18 notes 1 and 4

Receive Loudness Rating RLRR dB +3 -5 to +14 notes 1 and 4

Sidetone Masking Rating STMR dB 15 10 to 20 note 2Listener Sidetone Rating LSTR dB 18 13 to 23 note 2D-Value of Telephone, Send Side Ds - 3 -3 to +3D-Value of Telephone Receive Side Dr - 3 -3 to +3 note 2Talker Echo Loudness Rating TELR dB 65 5 to 65Weighted Echo Path Loss WEPL dB 110 5 to 110Mean one-way Delay of the Echo Path T ms 0 0 to 500Round Trip Delay in a 4-wire Loop Tr ms 0 0 to 1 000Absolute Delay in echofree Connections Ta ms 0 0 to 500Number of Quantization Distortion Units qdu - 1 1 to 14Equipment Impairment Factor Ie - 0 0 to 40Circuit Noise referred to 0 dBr-point Nc dBm0p -70 -80 to -40Noise Floor at the Receive Side Nfor dBmp -64 - note 3Room Noise at the Send Side Ps dB(A) 35 35 to 85Room Noise at the Receive Side Pr dB(A) 35 35 to 85Expectation Factor A - 0 0 to 20NOTE 1: Total Values between microphone or receiver and 0 dBr-point.NOTE 2: Fixed Relation: LSTR = STMR + D.NOTE 3: This value shall not be modified.NOTE 4: Default values for North American use are SLRS = +8 dB, RLRR = +2 dB.

The graphic may correspond to a too detailed level of information for the purpose of the presentdocument but gives a perception on how to measure voice transmission parameters with artificiallygenerated voice signals (ITU-T rec. P.50)

Reference Codec G.711, G.726 or GSM FR

Network Simulator Electrical Part

ITU-T test sequences (P.50, P.501)

Electrical Part

Acoustic Part

TIPHON Terminal 0 dBr Reference point

electrical access

ITU-T test sequences (P.50, P.501)

acoustical access

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