voice over ip - Ústav telekomunikáciíchromy/sps2/referaty/57 - voice over... · 2010-09-02 ·...

VOICE OVER IP

Chris Roberts March 2005

Centre for Critical Infrastructure Protection

PO Box 12-209 Wellington

New Zealand

Telephone: +64 4 498-7654 Email: [email protected]

Website: http://www.ccip.govt.nz


VOICE OVER IP

What is VoIP? 5 Enablers and Barriers 5 Growth 5 Origins 6

VoIP Standards 8

H.323 8 Session Initiation Protocol (Sip) 8 Other Standards 9 Figure 1 H.323 and Related Standards 9 VoIP Protocols 10 Figure 2 VoIP Protocol Structure 11 Figure 3 VoIP Protocols 11

VoIP Basic Functions 12

Signalling 12 Database Services 12 Bearer Control 12 Codec Operations 12

VoIP Components 13

Figure 4 - Simple VoIP System 13 Call Processing Server 13 Figure 5 - Call Processing Server 14 User Devices 14 Media/VoIP Gateways 14 Figure 6 - VoIP Gateway Functions 15 IP Network 15

Quality of Service 16 Voice Quality 17

Delay 17 Figure 7 Cumulative Transmission Path Delay 17 Table 1: Encoding Standards and Conversion Delays,, 17 Jitter 18 Packet Loss 18 Voice Processing 18 Interoperability 18 Figure 8 - Typical Delay Times 19

Why Move to VoIP 20

Cost Savings 20 Improved Efficiency and Productivity 20 Simplification and Consolidation 20 Improved Capability 20

Implementing VoIP 22

Functionality 22 Fault Tolerance and Management 22 Accounting and Call Billing 22 Configuration Management 23 Addressing and Directory Management 23 Access Control, Authentication and Encryption 23 Security 23 Legal Considerations 23 Deploying a VoIP Network 23


Appendix A - Acronyms 25 Appendix B - International Organisations 26

International Telecommunication Union (Itu) 26 Internet Engineering Task Force (Ietf) 26 International Organisation For Standardisation (Iso) 27

REFERENCES 28


VOICE OVER IP (VOIP)

Although we live in an analogue world, much of the technology we use is digitally based. Examples include not only computers, but digital voice recorders, digital cameras, digital watches and so on. This digitisation applies to our telephone systems as much as any other piece of technology.

Many of the analogue PBX (Private Branch Exchange) and PABX (Private Automated Brach Exchange) installations are nearing the end of their economic life with many manufacturers discontinuing product lines in favour of IP (Internet Protocol) telephony enabled replacements1. The chances are, however, your data network is amongst the estimated 85% of networks in use today that are not ready to support IP telephony without modification2. With an industry-wide move to voice and data convergence, VoIP is likely to feature prominently in organisation’s IT strategic planning and investments over the next 5 to 10 years.

A distinction should also be made between VoIP and Voice over Internet (VoI). While both use the Internet Protocol, VoIP is considered to be the commercial provision of IP telephony and much of the traffic is expected to be carried over circuits and channels provided and managed by telecommunications organisations.

Voice over Internet, by contrast, uses the Internet as its primary channel and is subject to all the delays, variable routing and other challenges of using the Internet for voice communication.

This paper is primarily a discussion of VoIP.

Voice over Internet VoI is generally a PC to PC method of communication and uses software from sites such as Skype3, MyFreeLD.com, Dialpad.com, To-Talk Communications and Conflab.com.

PC to PC calling has some drawbacks. Both PC’s need to be online and running the same Internet phone software. The quality of your call is dependent on connection speeds, routing delays, error rates and other similar factors. If the connection is slow or there are other traffic inhibitors it may be difficult to hold a conversation4.

Some VoI software is now offering PC to normal telephone connectivity. This, however, is still subject to the difficulties outlined in PC to PC calling.


WHAT IS VOIP?

Voice over Internet Protocol, also known as Internet Telephony and IP Telephony, is the ability to make voice telephone calls, send faxes and video-conference over IP based data networks5, whether an organisation specific LAN (local area network), WAN (wide area network) or the Internet itself. VoIP converts voice (analogue signals) into digital form and organises the data into packets. Packets are then transmitted by the most convenient route to their destination and reassembled before delivery.

In contrast, traditional circuit-committed protocols of the PSTN (public switched telephone network) provide dedicated, duplex circuits and bandwidth for the duration of the call. Duplex circuits provide separate channels to allow simultaneous two-way communication.

High reliability, based on over 135 years of experience in providing telephony services has created expectations of always available, reliable, real-time and good quality voice telephony. Telephones are even expected to work when the power is off6. While this analogue system is of very high reliability and quality, it is not an efficient use of bandwidth by data transmission standards.

Enablers and Barriers Some of the factors that have been promoting VoIP include:

• Low cost/no cost software (softphone and configuration tools) for PC’s and PDA’s; • Wide availability of analogue telephone adapters; • Growing availability of broadband, wireless “hot spots” and other forms of

broadband access; and • Relative high cost of PSTN calls. Some of the barriers opposing VoIP include:

• High quality and reliability of the PSTN; • VoIP quality of service can be variable; • Some VoIP feature, service and VoIP service provider interconnection limitations; • Relative difficulty in setup and use; and • Introduction of call plans and flat rates charges by traditional PSTN operators.

Growth According to Gartner Vice President, Jeff Snyder in early 2004, “In two years we will be seeing a critical mass of applications in this area. These applications are what are going to make the compelling argument for switching to IP telephony7.

There are a variety of forecasts on the growth of VoIP, many indicating increasing growth to 2007 and exponential growth thereafter. For example

• By 2008, wholesale VoIP traffic in Europe, Middle East and Asia could reach 57 billion minutes (Frost & Sullivan, May 2002)8:;

• The IP PBX market is expected to grow to 20% of all traditional PBX sales by 2005 (Synergy Research February 2002);


• The use of IP Telephony by US firms is expected to grow to 19% by 2007 (FCC Chairman Michael Powell);

• According to The Yankee Group VoIP in the US has grown from 131,000 in 2003 to 1 million subscribers in 2004 and is forecast to serve 17.5 million US households by the end of 20089(August 2004);

• Juniper Research forecasts VoIP adoption will grow from 1% of all US broadband households in 2004 to 17% by 2009, representing 12.1 million households (September 2004)10;

• VoIP penetration will increase from 10% of US business and organisation in 2005 to 45% by the end of 2007 (Osterman Research February 2005)11;

• 1 in 6 US organisations have converged or are have substantially completed convergence of voice and data networks (Osterman Research February 2005)12;

13 One effect of the expected growth in VoIP is a growth in the Internet as there is currently insufficient bandwidth in the Internet to replace the PSTN14.

Origins In the 1990’s Internet telephony or VoIP applications from different vendors were incompatible due to fundamental differences in voice coding, silence suppression, addressing and dialling plans, call management, and other related functions. To address these problems a number of non-profit working groups of industry players were formed to develop standards, specifications and interoperability guidelines.

Founded in May 1996, the original VoIP Forum, was a consortium of major equipment vendors and technology organisations including Cisco, VocalTec, 3Com, Microsoft, US Robotics and Netspeak to promote the use of ITU-T H.323, the ITU standard for sending voice and video using IP15. In October 1996 the Forum joined the International Multimedia Teleconferencing Consortium (IMTC) and has operated under that umbrella since then16.


There are several other VoIP Forums established by government authorities, industry discussion groups and industry news sites17. For example, in 2003 the Federal Communications Commission (FCC) announce a VoIP Forum. This was a public hearing, with the purpose of gathering information on advancements, innovations, and regulatory issues related to VoIP services18.

The IMTC was formed in September 1994 through mergers with several similar industry groups. Since then several other groups have joined the IMTC acting as an industry convergence point for Voice, Data, and Video over IP19.


VOIP STANDARDS

The International Telecommunication Union (ITU) and the Internet Engineering Task Force (IETF) are the two major international organisations recommending standards for VoIP. The ITU recommends H.323 and the IETF recommends the Session Initiation Protocol (SIP). While there is some overlap of functionality there are differences in approach and terminology. In addition, some vendors are providing proprietary, product dependent implementations. Both protocols can be extended to manage new capabilities. The argument has been advanced that H.323 is more stable because of its maturity but SIP provides better support for some functionality and is easier to implement20. Fortunately the ITU and the IETF are now co-operating in developing standards in this area.

H.323 Currently in version 2, H.323 is a standard recommended by the Telecommunication Sector of the ITU. It defines real-time multimedia communications and conferencing over packet-based networks that do not provide a guaranteed Quality of Service (QoS) such as the LAN and the Internet. It is an “umbrella standard” belonging to the H.32x class of standards recommended by the ITU for videoconferencing applications21: These were amongst the earliest standards to classify and provide solutions to VoIP.

1. H.310 for conferencing over Broadband ISDN (B-ISDN); 2. H.320 for conferencing over Narrowband ISDN; 3. H.321 for conferencing over ATM; 4. H.322 for conferencing over LANs with guaranteed QoS; 5. H.324 for conferencing over Public Switched Telephone Networks. Earlier versions of H.323 had a large overhead in control signalling, particularly when establishing a session. This has presented some scalability limitations, especially when a large number of simultaneous sessions are presented. Subsequent version have focussed on addressing these issues.

Session Initiation Protocol (SIP) As the Internet developed, the IETF produced a large number of standards and protocols through the Request for Comment (RFC) process. In the VoIP area, some were based on ideas in the H.323 standard and developed through the RFC process.

SIP is a protocol to invite an individual users to participate in a point-to-point or unicast session and part of the IETF’s multimedia data and control protocol framework. It manages the setup and orderly termination of sessions which may include telephone calls, videoconferencing and multimedia distribution.

Sip is text-based and designed to be simple, efficient and extensible. It has inherited some design characteristics from the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP)22.


Other Standards Other relevant standards and recommendations include:

1. H.225 defines the lowest layer that formats the transmitted video, audio, data, and control streams for output to the network, and retrieves the corresponding streams from the network;

2. H.235 specifies the security requirements for H.323 communications. Four security services are provided: authentication, integrity, privacy, and non-repudiation;

3. H.245 specifies messages for opening and closing channels for media streams and other commands, requests and indications;

4. H.248, also known as Megaco (MEdia GAteway COntrol), is a current draft standard and a co-operative proposal from IETF and ITU. Also described in RFC 3015. It addresses the same requirements and has many similarities to MGCP (see below);

5. H.261. If video capabilities are provided, it must adhere to the H.261 protocol with QCIF as its mode;

6. H.263 specifies the CODEC for video over the PSTN; 7. Various audio CODECs are specified under G.711, G.722 G.723,G.723.1, G.726,

G.729 and G.729.a23; 8. T120 a protocol for data and conference control. Over 120 leading computer, telecommunication and technology organisations have indicated their intent to support and implement H.323 in their products and services. This wide ranging support establishes H.323 as the de facto standard for audio and video conferencing over the Internet.

Figure 1 H.323 and Related Standards24


VoIP Protocols There are a number of other protocols that may be used in VoIP applications. Although these protocols will generally interoperate with H.323 standards, some may not. Other protocols include:

1. Media Gateway Control Protocol (MGCP). A development of SGCP and IPDC protocols;

2. IP Device Control (IPDC). A group of protocols for controlling hardware devices such as control gateway devices at the boundary between the circuit- switched telephone network and the Internet. Examples of such devices include network access servers and voice-over-IP gateways.25

3. Real Time Transport Protocol (RTP). Described in IETF RFC 1889, this is a real-time, end-to-end protocol, utilising existing transport layers for data that has real-time properties;

4. RTP Control Protocol (RTCP). Described in IETF RFC 1889, a protocol to monitor QoS and carry information on the participants in a session. It also provides feedback on total performance and quality so allow modification to be made.

5. Resource Reservation Protocol (RSVP). Described in IETF RFC 2250-2209. This is a general purpose signalling protocol allowing network resources to be reserved for a connections data stream, based on receiver-controlled requests. There may be scability issues in using this protocol due to its focus and management of individual application traffic flows26;

6. Simple Gateway Control Protocol (SGCP). SGCP is a simple "remote control" protocol that the call agent uses to program gateways according to instructions received through signalling protocols such as H.323 or SIP27. Now superseded by MGCP, an IETF work in progress;

7. Session Announcement Protocol (SAP). Protocol used by multicast session managers to distribute a multicast session description to a large group;

8. Real Time Streaming Protocol (RTSP). Interface management to a server providing real-time data;

9. Session Description Protocol (SDP). Describes the session for other protocols including SAP, SIP and RTSP.

In common with many communication and data systems, the protocols used in VoIP generally follow a layered hierarchy, similar to the Open Systems Interconnect theoretical model developed by the International Organisation for Standards (OSI). There are, however, exceptions to this, for example IP over ATM28.


Figure 2 VoIP Protocol Structure

Figure 3 VoIP Protocols The following table provides an overview of the principal VoIP protocols, as described in a Cisco white paper29:

H.323 SIP MGCP/H.248/Megaco

Standards body

ITU IETF MGCP/Megaco—IETF H.248—ITU

Architecture Distributed Distributed Centralized Current version

H.323v4 RFC2543-bis07 MGCP 1.0, Megaco, H.248

Call control

Gatekeeper Proxy/Redirect Server

Call agent/media gateway controller

Endpoints Gateway, terminal User agent Media gateway Signaling transport

Transmission Control Protocol (TCP) or User Datagram Protocol (UDP)

TCP or UDP

MGCP—UDP; Megaco/H.248—both

Multimedia capable

Yes Yes Yes

DTMF-relay transport

H.245 (signaling) or RFC 2833 (media)

RFC 2833 (media) or INFO (signaling)

Signaling or RFC 2833 (media)

Fax-relay transport

T.38 T.38 T.38

Supplemental services

Provided by endpoints or call control

Provided by endpoints or call control

Provided by call agent


VOIP BASIC FUNCTIONS

There are three basic types of VoIP calls:

1. PC to PC; 2. PC to ‘phone/’phone to PC; and 3. ‘phone to ‘phone. All, however, must be able to perform the same basic functions as the PSTN network. These are:30

1. Signalling; 2. Database services; 3. Bearer Control; and 4. CODEC operations.

Signalling Depicted as layers 1 to 3 in Figure 2 above, signalling is the means by which devices communicate, initiating and managing necessary functions in order to complete a call. In a VoIP network signalling is achieved through the exchange of IP messages compared with a PSTN network where a PABX, PBX or a switch manages call connection and routing.

Database Services In a PSTN, endpoints are identified through the allocation of a ‘phone number. In a VoIP network and IP address is the primary means of identification, although an endpoint may also be allocated a ‘phone number. A call control database records and manages endpoint identifiers and mappings. It will also record transactions for billing, audit, operational and security management.

Bearer Control Simplistically, this is the call connect and call disconnect management.

CODEC Operations CODECs (coder/decoders) provide the means to convert analogue voice signals to digital signals and reverse the process on delivery. CODECs are also known as VOCODERs or voice coder/decoders.

On conversion from analogue to digital, a data stream is packetised and transported across the network. The receiving endpoint will not only have to reassemble the packets into the correct sequence, but also decode the contents. Clearly commonality of standards and CODECs is essential if the communication is to be intelligible.

Any detected signalling tones are routed around the CODEC which can modify the tones to the point it is not recognised by the device being signalled.


VOIP COMPONENTS

The major components in a VoIP network are:

1. Call processing server; 2. User devices; 3. Media/VoIP gateways; and 4. IP Network.

Figure 4 - Simple VoIP System

Call Processing Server A call processing server, also known as an IP/PBX manages all VoIP connections. These can be deployed in various configurations from a single server through to a server farm. Fault tolerance, redundancy, reliability, load, flexibility and traffic growth will factor in the choice of configuration. These deployments are based on standard hardware running specialist software applications. Call processing servers are also available on router platforms or as an appliance.

To establish a VoIP communication, control traffic establishes and manages a call while the voice traffic is carried as a VoIP payload or voice stream in a peer to peer fashion in an RTP stream as illustrated in Figure 4 below. Exceptions to this are some conferencing functionality, music-on-hold and routed voice traffic to another call server.


Figure 5 - Call Processing Server

User Devices User devices may be discrete devices, sometimes known as “hard phones’ or software based or “soft phones”. Such devices include VoIP ‘phones, PCs and other desktop or mobile VoIP devices such as laptop computers.

Media/VoIP Gateways Sometimes also known as gatekeepers which were traditionally used for call admission and control and bandwidth management31. With technology convergence, the distinction between this function and traditional gateways has fallen away and the functionality now exists within traditional gateways.

The main function of a media gateway is to create VoIP packets from analogue voice signals using CODECs. Other features such as compression, echo cancellation, silence suppression and traffic management are often incorporated into gateway functionality. Media gateways can fulfil a number of functions32:

• Trunk gateways that form the interface between a telephone and VoIP network, typically managing multiple digital circuits;

• Residential gateways that provide an analogue interface to a VoIP network. Examples include cable modems, xDSL devices and broadband wireless devices;

• Access media gateways provide an analogue or digital PBX interface to a VoIP network. Examples include small-scale (enterprise) VoIP gateways;

• Business media gateways that provide digital PBX interface or an integrated soft PBX interface to a VoIP network;

• Network access servers that connect a modem to a telephone circuit and provide data access to the Internet.

Figure 5 illustrates the functional components of gateways using the H.323 standard33.


Figure 6 - VoIP Gateway Functions

IP Network The IP network connects the, often distributed, elements of a VoIP network. As VoIP traffic is sensitive to delay, Quality of Service must be maintained. This is discussed below. The IP network prioritises VoIP traffic through Class of Service (CoS) identifiers to ensure VoIP traffic is not affected by other network traffic.


QUALITY OF SERVICE

Data networks are designed to transport data efficiently but can be susceptible to jitter, delay, packet loss, data errors, bandwidth fluctuations and “dropped” connections. While IP was designed to manage these conditions, latency and jitter can have an unacceptably detrimental effect on voice and video over IP.

The reliability of today’s PSTN allows fast call setup times, a robust feature set, and a sophisticated billing and settlement system34 This has created expectations of QoS for VoIP calls, which should be at least as easy and as good as PBX or PSTN calls (sometimes referred to as “toll quality”35 or “feature parity”36. In some cases, the fact that VoIP is perceived as an application of technology, compared with a simple telephone, creates expectations of higher capability. VoIP QoS requires careful network management, monitoring, fault resolution, call management and security management37.

QoS covers the following aspects:

• voice quality; • network performance; • call control (packet loss and voice processing); and • interoperability.


VOICE QUALITY

Delay Because voice is analogue and has a continuous nature, digitising can introduce a number of quality problems. Packet delay, also described as latency, is the time a packet takes to travel from its point of origin to its destination. Delay can be caused by a number of technical issues including call distance and routing (propagation and network transit delays), incompatible CODECs (packet encoding/decoding), bandwidth reliability and disparity, network congestion, and data errors. Echo and talker overlap are the most apparent with echo becoming problematic when the round-trip delay is more than 50 milliseconds. Most calls can manage a delay of 150 milliseconds or less. Talker overlap becomes problematic with delay of 250 milliseconds or more38 and any delay exceeding 400 milliseconds is untenable39

Figure 7 Cumulative Transmission Path Delay40

Table 1: Encoding Standards and Conversion Delays41,42,43

ITU Standard CODEC

Conversion Algorithm

Bandwidth (Kbps)

Conversion Delay (ms)

G.711 PCM 48, 56, 64 <1.0 G.721 ADPCM 16, 24, 32,

40 <1.0

G.722 Sub-band ADPCM

48, 56, 64 <2.0

G.723.1 Multirate CELP 5.3, 6.3 67-97 G.726 ADPCM 16, 24, 32,

40 60.0

G.727 EADPCM 16 ,24, 32, 40

60.0

G.728 LD-CELP 16 ~2.50 G.729 CS-ACELP 8 25-35 G.729 annex A CS-ACELP 8 25-35


This demonstrates the trade-off between bandwidth consumption and coding (compression and encoding/decoding) delays. It also indicates the encoding efficiency as part of this equation.

Jitter Jitter is delay variability or the variation in packet arrival times compared with expected arrival times. It is generally caused by transmission delays and variable packet routing. Packets can arrive in any order and removing jitter requires holding packets in a buffer until the slowest packet has arrived. Only then can that voice segment be played in the correct sequence. Jitter buffers themselves can add delay. To reduce the effect of jitter, data is buffered on receipt and replayed at a constant rate. Protocols supporting this include RTP and RTCP.

A subset of this delay variation is to describe jitter as a high frequency variation and wander as a low frequency variation. Jitter should be less than 60ms. Toll quality call have a maximum of 20ms jitter44.

Packet Loss IP is designed to work in a world where packet delivery cannot be guaranteed. Packet loss may occur where network congestion forces queue buffer overflows and network devices discard packets. Error correction and packet retransmission requests are designed to manage data errors and non-delivered packets in a data world. Again this can add delay. CODECs manage lost packets by substituting “white noise” or replaying the last successfully received packet. This may also occur when an “out of order” packet is received. This substitution is usually undetectable until a 1% packet loss threshold. Packet losses greater than 10% are generally not tolerable.

Voice Processing Voice processing transforms analogue speech into digital packets using compression and such techniques as silence suppression. This the number of packets transmitted and conserves bandwidth. A voice call may include pauses, breaths and other periods of silence which can amount to as much as 60% of the call. Since the absence of packets is interpreted at the destination as silence, CODECs generally add “white noise or “comfort noise” to substitute for the absent packets.

Interoperability This cover both equipment and service interoperability. Apart from the network equipment and desktop VoIP devices, voice and data networks require gateways. All this equipment and the services over these networks must be VoIP capable.


Figure 8 - Typical Delay Times


WHY MOVE TO VOIP

The rationale for moving to VoIP usually includes:

• cost savings; • improved efficiency and productivity; • simplification and consolidation; and • improved capability.

Cost Savings The cost savings are expected mainly from reduced toll call costs. While this may be an operational cost saving, some considerable capital expenditure may be required to achieve this. Fortunately, most new network equipment is VoIP capable so as older equipment is retired, the network can become VoIP capable with little additional capital cost. There are also opportunities to consolidate service contracts, with associated cost savings.

Real savings, particularly in the longer term, are the subject of continued debate. This is clouded by the improved equipment capability and falling cost of such items as the VoIP desktop ‘phones.

Improved Efficiency and Productivity Current VoIP system include a number of productivity tools such as integrated email, voicemail, fax and messaging45. Other features may include called party “presence” (which is not synonymous with avaiability), and a “follow-me/find-me” capability.

Customer Relationship Management (CRM) systems can integrate with VoIP systems for better customer information capture and better customer service.

Simplification and Consolidation VoIP is an opportunity to consolidate data and voice communication, part of the growing trend of technology convergence. It can reduce equipment requirements, simplify network management and lead to application convergence. Complexity can be reduced and with properly planned networks, flexibility to add, change or remove nodes improved, single points of failure eliminated and security improved. There may also be opportunities for operational efficiencies in improving communication (for example, desktop to desktop video conferencing) and consolidating enterprise systems.

Improved Capability In addition to integrating telephony and facsimile requirements, ubiquitous video conferencing, electronic whiteboarding, multimedia and multiservice applications will become feasible. VoIP can deliver numerous features including advanced call routing, computer integration, unified messaging, integrated information services, toll call bypass and encryption46. For example, the ability to integrate service centre calls (help desk) with web services and shared screens is expected to provide faster call resolution times. Other examples include voice messages delivered to multiple


mailboxes over the Internet, voice-annotated documents and “follow-me” features where a person is always contactable at a single telephone number or extension number, irrespective of physical location.

47


IMPLEMENTING VOIP

While VoIP may be justifiable for financial, operational or strategic reasons, the implementation of VoIP may be more problematic. A number of aspects will need to be accommodated, including:

• Functionality; which may include interoperability, reliability, availability and accessibility

• Fault tolerance and management; • Accounting and call billing; • Configuration management; • Addressing and directory management; • Access control, authentication and encryption; • Security; • Legal considerations; and • Network deployment.

Functionality A basic feature of VoIP is the ability for anyone to call anyone else, regardless of location or how they are connected to the network (POTS, VoIP ‘phone, wireless ‘phone, PC, facsimile or some other device). VoIP should also be at least as good as current telephony services in terms of reliability, interoperability, availability, accessibility, QoS, management and security. VoIP has the potential to grow rapidly and any implementation should also provide scalability.

Desktop functionality should not be neglected and while multimedia and multiservice applications will become commonplace, other productivity enhancers may include such features as text-to-speech/speech-to-text and voice response systems.

Fault Tolerance and Management Today’s networks are expected to provide a high reliability service and in order to do so they must be fault tolerance both in hardware and software. Fault tolerance is a function of the design, systems architecture, interoperability and quality of the system devices and software. Many existing networks will continue to handle data adequately but may require significant re-engineering and investment to provide the QoS and robustness expected in a VoIP network.

Swift identification and resolution of failures, faults and other problems is a fundamental network management task. Adequate management tools must be provided for this task.

Accounting and Call Billing Accounting and call billing must track network traffic and calls to assist in network and traffic management and to allocate and recover costs where appropriate. Call details should include such aspects as call duration, number dialled, source and destination IP address, packets sent and received and so on. Users should receive a consolidated billing for network usage. Audit and security considerations should also be taken into account when establishing the accounting and billing system.


Configuration Management An important element of network management, any configuration management system should permit live-system configuration, in addition to the authorisation, record keeping, backup and management of configuration changes. Examples of live system changes may include protocol changes, compression algorithm modification or change, dialling plans, access control, audit, recovery, fault management, port reassignment and so on.

Addressing and Directory Management Any configuration changes should be transparent to the user so, for example, an IP address reassignment should include a dynamic directory update. Multimedia PC’s may need a telephone number, as well as an IP address and directory services should include mappings between different address types.

Access Control, Authentication and Encryption Access to VoIP networks should be at least as carefully managed as access to data networks. VoIP also offers authentication and encryption capabilities which can enhance access control and call security.

Security VoIP adds a level of complexity to network operations and VoIP cannot be treated as a new function on an already secure network. VoIP may introduce new vulnerabilities and any VoIP network must be designed with particular security considerations in mind. Security is discussed in more detail below.

Legal Considerations There are a number of legal and contractual considerations which may impact the decision to implement a VoIP network. These may include:

• Caller ID and Local Number Portability (LNP); • Calling line identification restrictions; • Malicious call identification; • Legal interception; • Carrier selection; and • Emergency services (111 calls).

Deploying a VoIP Network Once the implementation of a VoIP network has been agreed, there are a number actions required. While not an exhaustive list, these include:

• Systems and network architecture design; • Selection and procurement of core infrastructure devices and cabling; • Interfaces to the PSTN; • Determination of routing plans and an internal number convention; • System resilience including automatic recovery calls flows and call rerouting rules;


• Single point of failure identification; • Disaster recovery planning and equipment acquisition; • Billing strategies and links to billing and accounting systems; • Change management; and • Technical and user training.


APPENDIX A - ACRONYMS

CODEC Coding/decoding utility CoS Class of Service DES Data Encryption Standard DoS Denial of Service FCC US Federal Communications Commission http://www.fcc.gov/ HTTP Hypertext Transfer Protocol IETF Internet Engineering Task Force http://www.ietf.org/ IP Internet Protocol IPDC IP Device Control IMTC International Multimedia Teleconferencing Consortium

http://www.imtc.org/ ISO International Organisation for Standardisation ITU International Telecommunication Union http://www.itu.int/ IVR Interactive Voice Response LAN Local Area Network LNP Local Number Portability Megaco MEdia GAteway COntrol MGCP Media Gateway Control Protocol (supersedes SGCP) ms Micro Second OSI Open Systems Interconnect (or Interconnection), a theoretical

seven layer model of network architecture and a suite of protocols to implement that architecture. Developed by ISO in 1978 as a framework for international standards for the interconnection of data systems.

PABX Private Automated Branch Exchange PBX Private Branch Exchange POTS Plain Old Telephone System PSTN Public Switched Telephone Network QoS Quality of Service RFC Request for Comment RTCP RTP Control Protocol RTSP Real-Time Streaming Protocol RTP Real-Time Transport Protocol RSVP Resource Reservation Protocol SAP Session Announcement Protocol SCTP Stream Control Transmission Protocol SDP Session Description Protocol SGCP Simple Gateway Control Protocol SIP Session Initiation Protocol SMTP Simple Mail Transfer Protocol SNMP Simple Network Management Protocol, an application layer

protocol that facilitates the exchange of management information between network devices. Part of the Transmission Control Protocol/Internet Protocol (TCP/IP) protocol suite.

SSRC Synchronisation source identifier (RTP Header) TCP/IP Transmission Control Protocol/Internet Protocol protocol suite. VoI Voice over Internet VoIP Voice over Internet Protocol (IP) WAN Wide Area Network


APPENDIX B - INTERNATIONAL ORGANISATIONS

International Telecommunication Union (ITU)

The ITU, headquartered in Geneva, Switzerland, is an international organisation within which governments and the private sector co-ordinate global telecommunications networks and services. It is also the leading publisher of telecommunication technology, regulatory and standards information.

Founded originally as the International Telegraph Union on 17 May 186548, the ITU has, over the years, absorbed a number of related international organisations. It became a UN specialised agency on 15 October 1947. In 1992, the ITU was restructured with three sectors corresponding to its three main areas of activity Telecommunication Standardisation (ITU-T), Radiocommunication (ITU-R) and Telecommunication Development (ITU-D).

The ITU has oversight from an Administrative Council, comprising a maximum of 25% of the total number of Member States. Members of the Council are elected by regular conferences with due regard to the need for equitable distribution of Council seats among the five world regions (Americas, Western Europe, Eastern Europe, Africa, Asia and Australasia). The current Council comprises 46 Member States.

Internet Engineering Task Force (IETF)

The Internet Engineering Task Force (IETF) is a large open international community of network designers, operators, vendors, researchers and other interested parties contributing to the evolution of the Internet architecture and the smooth operation of the Internet. It is open to any interested individual. The technical work of the IETF is managed through working groups, which are organised by topic into seven areas :

• Applications; • Routing, • General; • Transport;

and • Internet; • Security. • Operations and

Management;


Development of protocols and standards is publicised through Requests for Comments (RFC’s) which are developed by the working groups, published in draft form for comment and feedback and finally published as Internet Standards. The first RFC was published in 1969. The most recent, numbered 4015, published in February 200549.

International Organisation for Standardisation (ISO)

ISO is a non-governmental organisation: comprising a network of the national standards institutes of 148 countries with a Central Secretariat in Geneva, Switzerland. The ISO name is derived from the Greek isos, meaning "equal", chosen because "International Organization for Standardization" would have different abbreviations in different languages.

International standardisation began in the electrotechnical field: the International Electrotechnical Commission (IEC) was established in 1906. Pioneering work in other fields was carried out by the International Federation of the National Standardizing Associations (ISA), which was set up in 1926.

In 1946, delegates from 25 countries met in London and agreed to create a new international organisation, the object of which was "to facilitate the international co-ordination and unification of industrial standards". ISO officially began operations on 23 February 1947.

International standards are achieved through consensus. Firstly nationally amongst all the economic stakeholders concerned - suppliers, users, government regulators and other interest groups, such as consumers. Secondly, internationally amongst the delegates to ISO. Agreement is reached on specifications, criteria, terminology and in the provision of services. International Standards thus provide a reference framework and a common technological language50.


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