This chapter provides technical background information about the protocols and components used in IP Telephony. It introduces the relevant component types, gives detailed information about H.323, SIP and RTP as well as information about media gateway control and vendor
-specific protocols.
} 2.1 Components
An IP Telephony infrastructure usually consists of different types of components.This section gives an overview of typical components without describing them in a protocol-specific context.
} 2.1.1 Terminal
A terminal is a communication endpoint that terminates calls and their media streams. Most commonly, this is either a hardware or a software telephone or videophone, possibly enhanced with data capabilities.There are terminals that are intended for user interaction and others that are automated, e.g., answering machines.
An IP Telephony terminal is located on at least one IP address.There may well be multiple terminals on the same IP address but they are treated independently. Most of the time, a terminal has been assigned one or more addresses (see Section 2.1.5), which others will use to dial to it.
If IP Telephony servers are used, a terminal registers the addresses with its server.
} 2.1.2 Server
Placing an IP Telephony call requires at least two terminals, and the knowledge of the IP address and port number of the terminal to call. Obviously, forcing the user to remember and use IP
addresses for placing calls is not ideal and dynamic IP addressing schemes (DHCP) make this requirement even more intolerable.
As mentioned before, terminals usually register their addresses with a server.The server stores these telephone addresses along with the IP addresses of the respective terminals, and is thus able to map a telephone address to a host.
When a telephone user dials an address, the server tries to resolve the given address into a network address.To do so, the server may interact with other telephony servers or services.
It may also provide further call routing mechanisms like CPL (Call Processing Language) scripts P.11
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or skill-based routing (e.g., route calls to ‘WWW-Support’ to a list of persons who are tagged to be responsible for this subject).
Finally, a telephony server is responsible for authenticating registrations, authorising calling parties and performing the accounting
{ 2.1.3 Gateway
Gateways are telephony endpoints that facilitate calls between endpoints that usually would not interoperate. Usually this means that a gateway translates one signalling protocol into another (e.g.
SIP/ISDN signalling gateways), but translating between different network addresses (IPv4/IPv6) or codecs (media gateways) can be considered gatewaying as well. Of course, it is possible that multiple functionalities exist in a single gateway.
Finding gateways between VoIP and a traditional PBX is usually quite simple. Gateways that translate different VoIP protocols are harder to find. Most of them are limited to basic call functionality.
{ 2.1.4 Conference bridge
Conference bridges provide the means to have 3-point or multi-point conferences that can either be ad-hoc or scheduled. Because of the high resource requirements, conference bridges are usually dedicated servers with special media hardware.
{ 2.1.5 Addressing
A user willing to use a communication service needs an identifier to describe himself and the called party. Ideally, such an identifier should be independent of the user's physical location.The network should be then responsible for finding the current location of the called party. A specific user may define to be reached by multiple contact address identifiers.
Regular telephony systems use E.164 numbers (the international public telecommunication numbering plan). An identifier is composed of up to fifteen digits with a leading plus sign, for example, +1234565789123.When dialling, the leading plus is normally replaced by the international access code, usually double zero (00).This is followed by a country code and a subscriber number.
The first IP Telephony systems used the IP addresses of end-point devices as user identifiers.
Sometimes they are still used now. However, IP addresses are not location-independent (even if IPv6 is used) and they are hard to remember (especially if IPv6 is used) so they are not suitable as user identifiers.
Current IP Telephony systems use two kinds of identifiers:
- URIs (RFC2396);
- Numbers (E.164).
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Some systems tried to use names (alpha-numeric strings), but this led to a flat naming space and thus limited zones of applicability.
A Universal Resource Identifier (URI) uses a registered naming space to describe a resource in a location-independent way. Resources are available under a variety of naming schemes and access methods including e-mail addresses (mailto), SIP identifiers (sip), H.323 identifiers (h.323, RFC3508) or telephone numbers (draft-ietf-iptel-rfc2806bis-02). E-mail-like identifiers have several advantages.They are easy to remember, nearly every Internet user already has an e-mail address and a new service can be added using the same identifier.The user location can be found with a Domain Name System (DNS).The disadvantage of URIs is that they are difficult or impossible to dial on some user devices (phones).
If we want to integrate a regular telephony system with IP Telephony, we must deal with phone number identifiers even on the IP Telephony-side.The numbers are not well suited for an Internet world relying on domain names.Therefore, the ENUM system was invented, using adapted phone numbers as domain names. ENUM is described in Chapter 7.
{ 2.2. Protocols
{ 2.2.1 H.323
The H.323 Series of Recommendations evolved out of the ITU-T's work on video telephony and multimedia conferencing. After completing standardisation on video telephony and videoconferencing for ISDN at up to 2 Mbit/s in the H.320 series, the ITU-T took on work on similar multimedia communication over ATM networks (H.310, H.321), over the analogue Public Switched Telephone Network (PSTN) using modem technology (H.324), and over the stillborn Isochronous Ethernet (H.322).The most widely-adopted and hence most promising network infrastructure - and the one bearing the largest difficulties to achieve well-defined Quality of Service - was addressed in the beginning of 1995 in H.323: Local Area Networks, with the focus on IP as the network layer protocol.The primary goal was to interface multimedia communication equipment on LANs to the reasonably well-established base on circuit-switched networks.
The initial version of H.323 was approved by the ITU-T about one year later, in June 1996, thereby providing a base on which the industry could converge.The initial focus was clearly on local network environments, because QoS mechanisms for IP-based wide area networks, such as the Internet, were not well established at this point. In early 1996, Internet-wide deployment of H.323 was already explicitly included in the scope, as was the aim to support voice-only applications and, thus, the foundations to use H.323 for IP Telephony were laid. H.323 has continuously evolved towards becoming a technically sound and functionally rich protocol platform for IP Telephony applications.The first major additions to this end were included in H.323 version 2, approved by the ITU-T in January 1998. In September 1999, H.323v3 was approved by the ITU-T, incorporating numerous further functional and conceptual extensions to enable H.323 to serve as a basis for IP Telephony on a global scale and as well as making it meet requirements in enterprise environments. Moreover, many new enhancements were introduced into the H.323 protocol.Version 4 was approved on November 17, 2000 and contains enhancements in a number of important areas, including reliability, scalability, and flexibility.
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New features help facilitate more scalable Gateway and MCU solutions to meet the growing market requirements. H.323 has been the undisputed leader in voice, video, and data conferencing on packet networks, and Version 4 endeavours to keep H.323 ahead of the competition.
{ 2.2.1.1 Scope
As stated before, the scope of H.323 encompasses multimedia communication in IP-based networks, with significant consideration given to gatewaying to circuit-switched networks (in particular to ISDN-based video telephony and to PSTN/ISDN/GSM for voice communication).
Internet / Intranet
ISDN
H.323
H.320
Terminal
H.323
Gatekeeper
PSTN
H.324
H.323
MCU
H.323
ATM