5.4 Gateways

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Gateways are used to interconnect applications on dissimilar network environments. These systems operate at the highest layer of the OSI reference model—the application layer (see Figure 5.6). A gateway consists of protocol conversion software that usually resides in a server, switch, or front-end processor. Gateways process the various protocols used on each network so that information from the sender is intelligible to the receiver, despite differences in technology, network protocols, or platforms.

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Figure 5.6: Gateway functionality in reference to the OSI model.

For example, when a systems network architecture (SNA) gateway is used to connect an asynchronous PC to a synchronous IBM SNA mainframe, the gateway acts as both a conduit through which the computers communicate and as a translator between the various protocol layers. The translation process consumes considerable processing power, resulting in relatively slow transmission rates when compared with other interconnection methods—hundreds of packets per second for a gateway versus tens of thousands of packets per second for a bridge or router.

In addition to its translation capabilities, a gateway can check on the various protocols being used, ensuring that there is enough protocol processing power available for any given application. It also can ensure that the network links maintain a level of reliability for handling applications in conformance with predefined error rate thresholds.

5.4.1 Gateway Applications

Gateways have a variety of applications. They can facilitate LAN workstation connections to various legacy host environments, such as IBM’s SNA systems and mid-range systems. Gateways can be used to consolidate hardware and software. An SNA 3270 gateway shared among multiple networked PCs, for example, can be used in place of IBM’s 3270 information display system or many individual 3270 emulation products. Although the IBM systems offer a standard means of achieving the PC-host connection, it is expensive when used to attach a large number of standalone PCs. The relatively high connection cost per computer discourages host access for occasional users and limits the central control of information.

If the PCs are attached to a LAN, however, one gateway can emulate a cluster controller and thereby provide all workstations with host access at a very low cost. Cluster controller emulators use an RS-232C or compatible serial interface to a host adapter or communications controller, such as an IBM 3720 or 3745. They can support up to 254 simultaneous sessions.

A gateway can also be used between TCP/IP workstations and a host. Instead of installing and managing the TCP/IP stack on the host—which can be very expensive —IT departments can take advantage of the protocol already in use on the host and connect the TCP/IP workstations through the gateway. Instead of using 3270 terminals in the IBM environment, for example, users can retrieve host data via the Internet using Web browsers on their PCs. This can result in lower connection costs for remote users who need access to information on the host.

Other applications of gateways include the interconnection of various e-mail systems, enabling mail to be exchanged between normally incompatible formats. The gateway function is actually provided by servers equipped with the X.400 international messaging protocol.

Cellular service providers that offer the short messaging service (SMS) also offer public SMS gateways, which allow anyone to compose and send messages from the service provider’s Web site. A number of independently operated message gateways also exist on the Internet. Some gateways have more features than others. A few let users compose messages to more than one recipient, create group lists, manage messages, and send preset or customized replies.

One of the newest uses of gateways is to provide interconnectivity between the Internet and the PSTN, enabling users to place phone calls from their multimedia PCs or conventional telephones over the Internet or a carrier’s managed IP data network and vice versa. This arrangement allows users to save on long distance and international call charges. Examples of gateways for voice applications include the following:

  • Trunking gateways that interface between the telephone network and an IP network (such gateways typically manage a large number of digital circuits);

  • Voice over ATM gateways, which operate much the same way as VoIP trunking gateways, except that they interface to an ATM network;

  • Residential gateways that provide a traditional analog (RJ11) interface to a VoIP network (examples include cable modem/cable set-top boxes, xDSL devices, and broadband wireless devices);

  • Access gateways that provide a traditional analog (RJ11) or digital private branch exchange (PBX) interface to a VoIP network (examples of access gateways include small-scale voice over IP gateways);

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

  • Circuit switches, or packet switches, that offer a control interface to an external call control element;

  • Interactive voice response (IVR)/announcement servers that can provide IVR and announcements to VoIP networks.

A telephony gateway, in summary, is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.

5.4.2 Facilitative Protocols

With VoIP becoming more prevalent, there is growing interest in IP-PSTN gateways. Two of the primary standards in use today for joining IP nets and the PSTN are H.323 and the Session Initiation Protocol (SIP). The International Telecommunication Union (ITU) established H.323 as the first communications protocol for real-time multimedia communication over IP. Later, the IETF developed SIP for voice and video over IP. Both work well and support a comprehensive set of telephony features, but of the two, H.323 gateways are more prevalent, largely because they have been around longer. Another standard, the Media Gateway Control Protocol (MGCP), is also used. Some vendors, such as Cisco, support all three in their VoIP products.

H.323

This is actually a suite of protocols (see Figure 5.7) that defines how audio and videoconferencing systems communicate over packet-switched networks, such as the public Internet, and private intranets, which do not guarantee QoS. The H.323 protocol suite addresses call control and management for both point-to-point and multipoint conferences. The main components of an H.323 network (see Figure 5.8) include:

  • Terminals: These are any client device, hardware or software, capable of communicating with one or more other devices.

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    Figure 5.7: The H.323 protocol stack.

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    Figure 5.8: Basic H.323 network topology.

  • Gatekeepers: For each zone, these systems provide network-based services for the terminals such as registration, admission, and status (RAS) functions.

  • Multicast units (MCUs): These devices provide conference capabilities among H.323 terminals

  • Gateways: These systems convert signals and data transmission protocols so communication can take place between users on different types of networks.

When a standard voice call is received at a near-end gateway, the analog voice signal-is digitized, compressed, and packetized for transmission over an IP network. At the far-end gateway, the process is reversed, with the packets decompressed and returned to their original digital form for delivery to the nearest Class 5 central office.

The H.323 gateways support one or more of the internationally recognized G.7xx voice codec specifications for toll-quality voice. The most commonly supported codec specifications are the following:

  • G.711: Describes the requirements for a codec using pulse code modulation (PCM) of voice frequencies to achieve 64 Kbps, providing toll quality voice on managed IP networks with sufficient available bandwidth.

  • G.723.1: Describes the requirements for a dual-rate speech codec for multimedia communications (e.g., videoconferencing) transmitting at 5.3 Kbps and 6.3 Kbps. This codec provides near toll-quality voice on managed IP networks. [1 ]

  • G.729A: Describes the requirements for a low-complexity codec that transmits digitized and compressed voice at 8 Kbps. This codec provides toll-quality voice on managed IP networks.

The specific codec to be used is negotiated on a call-by-call basis between the gateways using the H.245 control protocol, a component of the H.323 protocol stack. Among other things, the H.245 protocol provides for capability exchange, enabling the gateways to implement the same codec at the time the call is placed. The gateways may be configured to implement a specific codec at the time the call is established, based on predefined criteria, such as the following:

  • Use G.711 only—in which case, the G.711 codec will be used for all calls.

  • Use G.729 (A) only—in which case, the G.729 (A) codec will be used for all calls.

  • Use highest common bit-rate codec—in which case, the codec that will provide the best voice quality is selected.

  • Use lowest common bit-rate codec—in which case, the codec that will provide the lowest packet bandwidth requirement is selected.

This capability exchange feature provides carriers and ISPs with the flexibility to offer different quality voice services at different price points. It also allows corporate customers to specify a preferred proprietary codec to support voice or a voice-enabled application through an intranet or IP-based VPN.

Session Initiation Protocol

Session Initiation Protocol (SIP) is an Internet protocol that provides simple application layer signaling for setting up, maintaining, and terminating multimedia sessions, including voice calls, videoconferences, and even instant messaging sessions. SIP performs many of the functions of the ITU H.323 multimedia conferencing standard, which was largely specified by telecommunication carriers. SIP, on the other hand, takes an Internet-oriented approach.

SIP requires only a datagram service and is independent of the packet layer. It can provide out-of-band call setup services in which the SIP exchanges take place over User Datagram Protocol (UDP) or TCP, but the actual voice conversation takes place over the public telephone network. Although most deployments of SIP will be over IP networks, it can also be used for non-IP networks such as ATM. In addition, because it communicates the capabilities exchange of the session, it can be used for voice, video, text messaging, or instant messaging.

A SIP-based network is made up of the following components:

  • SIP user agent: A network endpoint that can originate or terminate a session. This might include a SIP-enabled telephone, a SIP client (also known as a “soft-phone” installed on a PC), or a SIP-enabled gateway.

  • SIP proxy server: A call-control device that provides many services, such as routing of SIP messages between SIP user agents.

  • SIP redirect server: A call-control device that provides routing information to user agents when requested, giving the user agent an alternate uniform resource identifier (URI) or destination user-agent server (UAS).

  • SIP registrar server: A device that stores the logical location of user agents within the domain or sub-domain. This data is dynamically updated via REGISTER messages from the network.

  • SIP location services: Additional functionality that can be used by proxy, redirect, and registrar servers to find the identity (with a unique URI) and logical location of user agents within the network.

  • Back-to-back user agent: A call-control device that provides routing similar to a proxy server but allows centralized control of the network call flows. This device allows SIP networks to replicate certain traditional telephony services that require centralized knowledge of device state, such as call-park and pickup.

  • SIP-aware network devices: Devices that have knowledge of the SIP and allow the network to function more efficiently. This type of device might be a firewall or NAT device that can allow SIP traffic to cross network borders, or a load-balancing switch that allows requests to SIP servers to be more efficiently handled.

At one time, the H.323 call-setup process was considered unnecessarily cumbersome, requiring more than a dozen messages to traverse the network before voice communication could begin. SIP, on the other hand, requires only four messages before the start of voice communication. Recognizing this as a serious limitation, the ITU developed a “fast start” signaling process that cut the number of H.323 messages by half. Although H.323 sets up calls faster than before, SIP is still the simpler protocol, requiring much less code to implement than H.323. This is reflected in lower fixed and dynamic memory requirements, for both the SIP stack itself and the host application. Because of the smaller number of messages that the processor needs to handle to set up a call, SIP is still faster than H.323, which means SIP can process more calls per second—very important for call center environments—and requires a less powerful CPU to handle the same number of calls.

Media Gateway Control Protocol

Media Gateway Control Protocol (MGCP) is a protocol for the control of VoIP calls by external call-control elements known as media gateway controllers (MGCs) or call agents (CAs). MGCP assumes a call control architecture where the call control intelligence is outside the gateways and handled by external call control elements. The MGCP assumes that these call control elements, or CAs, will synchronize with each other to send commands to the gateways under their control. MGCP does not define a mechanism for synchronizing CAs; in essence, it is a master/slave protocol, where the gateways are expected to execute commands sent by the CAs.

MGCP assumes a connection model where the basic constructs are endpoints and connections. Endpoints are sources or destinations of data and could be physical or virtual. Examples of physical endpoints are the following:

  • An interface on a gateway that terminates a trunk connected to a PSTN switch;

  • An interface on a gateway that terminates an analog line to a phone, key system, PBX, or automated call distributor (ACD).

An example of a virtual endpoint is an audio source in an audio-content server. Creation of physical endpoints requires hardware installation, while creation of virtual endpoints is done by software.

Connections may be either point-to-point or multipoint. Once this association is established for both endpoints, data transfer between these endpoints can take place. A multipoint connection is established by connecting the endpoint to a multipoint session. Connections can be established over several types of networks, including the following:

  • Transmission of audio packets using the Real-Time Protocol (RTP) and UDP over an IP network;

  • Transmission of audio packets using AAL2, or another ATM adaptation layer, over an ATM network;

  • Transmission of packets over an internal connection; for example, the time division multiplexer (TDM) backplane or the interconnection bus of a gateway. This might be used for “hairpin” connections that terminate in a gateway but are immediately rerouted over the telephone network.

MGCP is designed as an internal protocol within a distributed system that appears to the outside as a single VoIP gateway. This system is composed of a CA, which may or may not be distributed over several computer platforms, and of a set of gateways. In a typical configuration, this distributed gateway system stands between one or more telephony circuit switches and H.323-conformant systems.

In the MGCP model, the gateways focus on the audio signal translation function, while the CA handles the signaling and call processing functions. Consequently, the CA implements the signaling layers of the H.323 standard and presents itself as an H.323 gatekeeper or as one or more H.323 endpoints to the H.323 systems. The distributed gateway systems and MGCP enable PSTN telephony users to access sessions set up using other protocols such as SIP. The CA provides for signaling conversion.

MGCP was developed by the telecommunications community to address signaling-system 7 (SS7) and VoIP integration issues. The H.323 initiative had originated out of the LAN environment and had trouble scaling to public network proportions. The H.323 architecture was incompatible with public telephony services, which are typified by multiple gateways and SS7. To address this problem, MGCP dispensed with the gatekeeper model and removed signaling control functions from the gateway, putting them in a media gateway controller or soft switch, which controls multiple media gateways. MGCP is the protocol used to communicate between the soft switch and the media gateways.

[1 ]The mean opinion score (MOS) used to rate the quality of speech codecs measures toll-quality voice as having a top score of 4.0. With G.723.1, voice quality is rated at 3.98, which is only 2% less than that of analog telephone.



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LANs to WANs(c) The Complete Management Guide
LANs to WANs: The Complete Management Guide
ISBN: 1580535720
EAN: 2147483647
Year: 2003
Pages: 184

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