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Voice Gateways and Gatekeepers
The Role of Voice GatewaysThe basic function of a gateway is to translate between different types of networks. In the data environment, a gateway might translate between a Frame Relay network and an Ethernet network, for example. In a VoIP environment, voice gateways are the interface between a VoIP network and the public switched telephone network (PSTN), a private branch exchange (PBX), or analog devices such as fax machines. In its simplest form, a voice gateway has an IP interface and a legacy telephone interface, and it handles the many tasks involved in translating between transmission formats and protocols. At least one gateway is an essential part of any IP telephony network that interacts with the PSTN or with analog devices. In addition, when gateways are properly configured, many can take over for a Cisco CallManager when it is unreachable. The gateway allows communication between the two networks by performing tasks such as these: · Interfacing with the IP network and the PSTN or PBX. · Supporting IP call control protocols, in addition to time-division multiplexing (TDM) call control protocols. · Performing call setup and teardown for calls between the VoIP and PSTN networks by terminating and reoriginating the call media and signaling. · Providing supplementary services, such as call hold and transfer. · Relaying dual tone multifrequency (DTMF) tones. · Supporting analog fax and modems over the IP network. · In a Cisco CallManager network, a gateway also needs to do the following: - Support CallManager redundancy by rehoming to alternate CallManagers. - Support call survivability when no CallManager is available. Gateways communicate with other gateways, gatekeepers, their endpoints, or their call control agents, such as Cisco CallManager or a PBX. The following are the protocols that Cisco gateways use for voice signaling and media: · Media Gateway Control Protocol (MGCP)
· H.323 · Session Initiation Protocol (SIP) · Skinny Client Control Protocol (SCCP) · Real-Time Transport Protocol (
Types of Voice GatewaysGateways contain analog or digital telephony ports, in addition to IP interfaces. Analog options include Foreign Exchange Office (FXO) ports, Foreign Exchange Station (FXS) ports, and Ear and Mouth (E&M) ports. Digital options include T1/E1 PRI ports, ISDN BRI ports, and T1 channel associated signaling (
RoutersYou can use many Cisco routers, such as 1700, 2600, 2800, 3600, 3700, 3800, 7200, and 7500 models, as voice gateways. Voice gateway routers can contain analog ports such as FXO and FXS, and digital ports such as E1 and T1. Most can use the MGCP, SCCP, SIP, and H.323 protocols. The remainder of this book focuses mainly on router-based voice gateways.
Standalone Voice GatewaysUnlike routers, standalone voice gateways are used only as voice gateways. They come in two types: digital and analog. Digital trunk gateways, such as the AS5000 series interface, connect an IP telephony network and the PSTN or a PBX via their trunk ports. Analog trunk gateways, such as AT-2 or AT-4, connect to the PSTN or a PBX via analog trunks. Analog station gateways, such as an ATA186, VG224, or VG248, connect to analog devices such as telephones, fax machines, or voice-mail systems. The signaling protocols that are available vary by gateway model. The ATA186, an analog-to-IP adapter, can support two analog devices each with its own telephone number. It has one Ethernet port that connects to the VoIP network, and two voice ports for connecting analog devices. It can be controlled by Cisco CallManager and CallManager Express, and it supports SIP and SCCP protocols. However, CallManager 5.0 does not have native SIP support for the ATA186. VG224 and VG248 are Cisco voice gateways with 24 and 48 analog FXS ports, respectively. These are line-side gatewaysthey do not interface with the PSTN, but with end-user analog devices. Both models allow a CallManager, CallManager Express (
The analog telephones, faxes, or modems that are attached to the gateway appear as individual endpoints to the CallManager. You must configure the gateway on the CallManager, and you must configure each port on CallManager with a directory number and any call features it needs. In
The VG248 gateway uses only SCCP. The VG224 gateway can use SCCP, H.323, MGCP, and SIP.
Switch ModulesModules containing analog and digital ports are available for Cisco 6500 switches, allowing them to act as voice gateways. For example, the Communication Media Module (which you can also use in the 7600 series router) can contain a combination of T1, E1, and FXS ports. It uses MGCP, H.323, and SIP protocols, and it can provide Survivable Remote Site Telephony (SRST) functionality. The Voice T1/E1 and Services Module, or 6608 blade, can contain T1 or E1 interfaces, and it can perform as an MGCP gateway.
The Role of Voice GatekeepersGatekeepers help VoIP networks scale to large sizes. Companies that have geographically dispersed voice networks, or networks that have become so large that they are unwieldy, might opt to segment their network. In a CallManager network, you can create multiple clusters. In that case, you would need to configure a full mesh of connections over the IP WAN to link all the segments or clusters. You would need to configure dial information for every remote location on every gateway and CallManager cluster. A better alternative is to use gatekeepers. In a network that has gatekeepers, trunks are needed only to the gatekeeper, and the gatekeeper maintains remote endpoint information. When you use gatekeepers, gateways and CallManagers register with their gatekeeper. Gatekeepers divide the network into "zones," or groups of devices that register with a particular gatekeeper. When an H.323 gateway receives a call that is destined to a remote phone, it queries the gatekeeper for the location of the endpoint. If the call is destined for a different zone, you can configure the gatekeeper to allow it only if sufficient bandwidth is available. In more complex networks, you can use a Directory gatekeeper to maintain information about all the zones. You can configure Cisco routers with the appropriate Cisco IOS as H.323 gatekeepers. Gatekeeper functionality is part of the H.323 standard. A voice gatekeeper provides the following services: · Address resolution A gatekeeper resolves E.164 telephone numbers and H.323 IDs to endpoint IP addresses. · Call admission control A gatekeeper permits or denies a call between clusters. · Bandwidth control A gatekeeper can refuse to admit calls that exceed the allocated bandwidth. · Zone management A gatekeeper can register and manage endpoints within its zone. · Optional Security A gatekeeper can authenticate and authorize calls on an endpoint-by-endpoint basis. · Optional call management A gatekeeper can maintain information about the endpoint call state. · Optional routing of call control signaling A gatekeeper can reroute signaling to allow endpoints to communicate directly. The Role of IP-to-IP GatewaysA voice gateway joins a VoIP network and the PSTN. A gatekeeper joins separate segments of the same VoIP network. An IP-to-IP gateway (IPIPGW), often called a Session Border Controller, joins independent VoIP or Video over IP networks. It acts as a border device, allowing users in different administrative domains to exchange voice and video using IP, rather than through the PSTN. The call media can either flow through the gateway, or directly between endpoints. For example, an Internet Telephony Service Provider (ITSP) can use IPIPGWs to route IP voice traffic through another ITSP network. An IPIPGW can provide billing information to the ITSP. IPIPGWs can allow an ITSP to offer its customers end-to-end VoIP service between each other, or between remote offices of the same company. This would allow the exchange of IP calls between CallManager, H.323, and SIP networks. Figure 1-3 shows an example of two companies that frequently conduct videoconferences between them. They use an IPIPGW to hide the details of each network, while still allowing communication. The H.323 video systems at each location communicate with the IPIPGW, rather than with each other directly. To each network, it looks as if the call signaling originates at the IPIPGW. The IPIPGW acts as a Session Border Controller (
You can install the IP-to-IP gateway Cisco IOS feature set on many Cisco multiservice routers. The following are some features of an IP-to-IP gateway: · Interconnecting segments of the same or different VoIP networks using different signaling types, such as H.323 and SIP · Interconnecting segments of the same or different VoIP networks using different media types · Billing abilities · Coder/decoder (codec) control · Call admission control · Security Introduction to Voice ProtocolsVoIP has several call signaling and control protocols available for use. The protocol that you should use depends on the type of gateway, endpoint host, and call agent, and the capabilities you need in the network. Multiple protocols might be used in different portions of the network. After you set up the call, you transmit IP voice and video traffic using a different protocol,
Media Gateway Control ProtocolMGCP is a client-server call control protocol, built on centralized control architecture. All the dial plan information resides on a separate call agent. The call agent, which controls the ports on the gateway, performs call control. The gateway does media translation between the PSTN and the VoIP networks for external calls. In a Cisco-based network, CallManagers function as the call agents. MGCP is an Internet Engineering Task Force (IETF) standard that is defined in several RFCs, including 2705 and 3435. Its capabilities can be extended by the use of "packages" that include, for example, the handling of DTMF tones, secure
An MGCP gateway is relatively easy to configure. Because the call agent has all the call-routing intelligence, you do not need to configure the gateway with all the dial peers it would otherwise need. A downside, however, is that a call agent must always be available. Cisco MGCP gateways can use SRST and MGCP fallback to allow the H.323 protocol to take over and provide local call routing in the absence of a CallManager. In that case, you must configure dial peers on the gateway for use by H.323.
H.323 is an International Telecommunications Union Telecommunication Standardization Sector (ITU-T) standard protocol. It has its roots in legacy telecommunications protocols, so it communicates well with hosts on the PSTN. H.323 is actually a suite of protocols that specify the functions involved in sending real-time voice, video, and data over packet-switched networks. Unlike MGCP, an H.323 gateway does not require a call agent; it is built on a distributed architecture model. Gateways can independently locate a remote host and establish a media stream; thus, you must configure them with call routing information. Although an H.323 gateway does not require a call agent, you can use it in a CallManager network. The CallManager directs calls that are bound for the PSTN to the gateway, which uses plain old telephone service (POTS) dial peers to route them. The gateway has a VoIP dial peer pointing to CallManager for calls that are bound inside the VoIP network. You can configure IP phones to register directly with an H.323 gateway using SRST when their CallManager is unavailable. The H.323 standard defines four components of an H.323 system: terminals, gateways, gatekeepers, and multipoint control units (MCU). · Terminals These are the user endpoints, such as a video conferencing units, that communicate with each other. · Gateways Used to communicate with terminals on other networks (primarily across the PSTN). · Gatekeepers Translate phone numbers to IP addresses, and control and route calls. · Multipoint control units (MCU) Enable multiple parties to join a videoconference.
Session Initiation ProtocolSIP, like MGCP, is an IETF standard, which is defined in a number of RFCs. Its control extends to audio, video, data, and instant messaging communications, allowing them to interoperate. SIP uses a distributed architecture based somewhat on the Internet model, using clear text request and response messages and URLs for host addressing. The protocol addresses only session initiation and teardown. It relies on other protocols, such as HTTP for message format, Session Description Protocol (
SIP uses several functional components in its call setup and teardown. Because these are logical functions, one device could serve several functions. SIP entities can act as a client or server, and some can act as both. Clients initiate requests for a service or information, and servers respond. One call might involve several requests and responses from several devices. Some SIP functions are as follows: · User agent The SIP endpoint, such as a SIP phone, which generates requests when it places a call and answers requests when it receives a call. · Proxy server The server that handles requests to and from a user agent, either responding to them or forwarding them as appropriate. · Redirect server The server that maintains routing information for remote locations and responds to requests from proxy servers for the location of remote servers. · Registrar server The server that keeps a database of user agents in its domain and responds to requests for this information. · Presence server The server that supports SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) applications. It collects and communicates user and device status, communications capabilities, and other attributes. SIP is a developing standard; therefore, interoperability between vendors and with other VoIP protocols can be a challenge. Much work is being done in this area, as SIP becomes more widely adopted.
Skinny Client Control Protocol
Cisco IP phones use the Cisco proprietary Skinny Client Control Protocol (SCCP), or "Skinny," to communicate with their call agent. As the "skinny" portion of the name implies, SCCP is a lightweight protocol that is built on a client-server model. Call control messages are sent over
Routers in SRST mode can use SCCP to communicate with the Cisco IP phones they control. Some analog gateway devices, such as a VG224 and Analog Telephone Adapter (ATA), can also use Skinny to communicate with a call agent. Real-Time Transport ProtocolMGCP, H.323, SIP, and SCCP are protocols that handle call signaling and control.
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