If you’re going to make the most of unified communications (UC), it’s helpful to understand one of the technologies that is crucial to making it work effectively: the Session Initiation Protocol, or SIP.
As its name implies, SIP is used to set up and terminate sessions, or conversations, in a UC environment. But it does far more than that, says Mykola Konrad, VP of Cloud and Strategic Alliances for Sonus Networks, Inc.
Four Fundamental SIP Functions
For starters, SIP handles four fundamental functions in a UC environment:
- Availability: SIP enables users to tell the UC system whether or not they’re available at any given time, giving rise to functions such as presence.
- Device capabilities: SIP can determine what kind of device the end user is employing and what sorts of communication the device can support, so it can set up the call appropriately. For example, one user may want to initiate a video call with three others, but one of them is using a device that only supports voice. SIP will initiate a video call for the three who can handle it and a voice call to the fourth participant.
- Session setup: SIP establishes the ground rules or parameters for a call between two or more participants.
- Management: SIP also handles management of the call, including any transfers, modifications (such as graduating from audio to video) and call termination.
“Those are the core capabilities but SIP has also become the lingua franca – or bridge language – of the VoIP, audio and, increasingly, the video world,” Konrad says. “It’s the glue holding the UC infrastructure together.”
By that he means SIP enables UC applications from different vendors to communicate and integrate with one another, whether it’s presence, instant messaging, voice or video. “In the end SIP enables productivity enhancements for the enterprise by enabling them to take advantage of all of these UC applications,” he says.
SIP Trunking Offers Cost Savings and a Foundation for UC
From a network infrastructure perspective, a technology known as SIP trunking can also enable significant cost savings and efficiencies. For starters, simply replacing a T1 line with a SIP trunk will likely save money because of the way carriers charge for the lines, Konrad says. T1s and PRI lines, for example, are sold in chunks of maybe four, eight or 24 channels, which translate to the number of simultaneous calls the line can handle. But many smaller offices only need one or two sessions, so the remaining capacity is wasted. “SIP is one-to-one,” he says. “If you only need to support one call into the office at a time, you can get that.”
The next step up is for the enterprise to replace the assorted T1 and analog lines it has coming into its various buildings with a single SIP-based trunk that comes into its main data center. From there, traffic of all sorts – voice, data and video – are carried over the company’s own wide-area network to its various branch and other locations.
“Then you really get economies of scale because you don’t have equipment in branches to terminate calls and sessions, and you’ve only got a single site to manage,” Konrad says. What’s more, SIP trunking can improve disaster recovery because it makes it a simpler proposition to have multiple carriers servicing your main data center; if one goes down, you can fail over to the other.
With a centralized SIP trunking architecture in place, the enterprise is now poised to add UC applications such as video, collaboration, messaging and presence – all riding over the SIP infrastructure. “Then they’ve opened up this entire UC world to their employees,” he says. “That’s how we’ve seen this progress.”