IP telephony gateway primer

IP telephony gateways serve an important and ever-increasing role in an enterprise unified communications infrastructure, delivering both circuit-switched and newer packet-based telephony solutions to the enterprise. Initially designed to adapt analog voice, the gateways are now the conduit to bring video and data solutions to enterprise users, no matter where they may be.

By Michael Brandenburg, Technical Editor

The IP telephony gateway is the bridge that connects an enterprise's unified communications (UC) solution with the outside world. This initially consisted of support for voice calling over public switched telephone network (PTSN) lines, but the role of the gateway has been expanded to include video conferencing, as well as data services such as Web conferencing and fax over IP.

IP telephony gateways also play an important role in extending UC services to remote sites across the wide area network (WAN), in many cases serving as a mini-PBX by providing not only call routing back to headquarters but also access to local dial tone in cases of emergency or WAN outage. On legacy circuit-based PBX systems, gateways can also serve as the intermediary to packet-based telecom services such as SIP trunking, delivering modern IP services to these older solutions. IP telephony gateways are available in a range of form factors, notably as trunk cards, standalone appliances or even PCI cards within server-based communications solutions.

While gateways are typically deployed by enterprises as part of a larger UC rollout, service providers may also install gateways on-premise for the customer as a means of delivering telecom services. An IP telephony gateway interfaces with the UC solution to set up and tear down calls on traditional telephony connections and encodes the call as an audio stream, enabling transmission over an enterprise's IP network.

Common protocols, interoperability key for IP telephony gateways

To communicate with the packet-based world of IP telephony, gateways need a common protocol. The predominant standard for communicating with modern IP telephony solutions is Session Initiation Protocol (SIP). SIP is being used throughout the UC infrastructure to connect gateways not only to services but all the way down to the individual handsets. The role of SIP is to locate the user on the telephony network and then determine his availability and media capabilities.

More IP telephony
gateway resources

IP telephony gateway duties: Integrating VoIP and the PSTN

What's the difference between a gateway and a gatekeeper?

Using a gateway/router setup to bypass the PSTN

Find answers to your toughest IP telephony questions at ITKnowledge Exchange

 The last component is important for ensuring that the system is not attempting to send a video stream to a user on a traditional audio desk set, for example. One of the advantages of SIP over its predecessors is that it shares format similarities with Web protocols like HTTP. As IP telephony solutions are making their way onto corporate networks, many network administrators are suddenly finding themselves in the role of telecom managers as well, and the Web nature of SIP no doubt eases the transition into that role.

Controlling the deployed gateways on the network are the SIP proxies. These serve as intermediaries for the IP telephony gateways, helping them locate users and enforcing any policies in place for those users. The proxies will, for example, block the attempts of users to initiate a video conference if they are not allowed to do so.

SIP is quickly taking over as the key link to both UC solutions and telecom managed services, and it is supplanting the H.323 standard, which served similar roles within early IP telephony products. While still supported by most gateway vendors, it is largely considered to be a legacy protocol, kept to maintain compatibility with older voice solutions that enterprises have already deployed.

Historically, a problem with gateways was a lack of interoperability among the various manufacturers. While the ubiquity of SIP is in many ways pushing vendors to a common level of standards, a few vendors are pushing the level of interoperability further through the Unified Communications Interoperability Forum (UCIF).

Looking at the IP telephony gateway market

For enterprises, interoperability is an important element, in part because of the way many gateways are sold. Large UC incumbents like Cisco will make and sell their own telephony gateways as part of an end-to-end UC portfolio. While these vendors do offer some third-party support, their primary focus is to sell and implement the entire package for their customers.

On the opposite end of the spectrum are software-based UC solutions, such as Microsoft's Office Communications Server, which rely on partners to provide the necessary telephony hardware. These partnerships have enabled a few gateway vendors, such as AudioCodes, to rise to prominence. The partnership model does offer enterprises the opportunity to pick and choose best-of-breed components for their telephony infrastructure, but the organization is still tied to the primary vendor's compatibility list and certification programs. IP telephony gateway vendors also sell to the service providers, enabling the carriers to roll out gateways as customer premises equipment (CPE) to support their managed telecom services.

This was last published in August 2010

Dig Deeper on IP Telephony Systems

Start the conversation

Send me notifications when other members comment.

Please create a username to comment.

-ADS BY GOOGLE

SearchCRM

SearchNetworking

SearchITChannel

Close