Voice over IP for business telephony is old news. But when business enterprises like Cisco connected to the outside world, they still used old-world technology. In the past two years Cisco IT has migrated its big connections to the outside world to Session Initiation Protocol (SIP). This move has saved us millions per year, made our contact center service better, and enabled global collaboration without breaking our budget. It has also simplified our internal voice architecture.
Best of all, it has positioned Cisco to build a B2B voice / video network to enable easier partnerships and better B2B collaboration.
Here’s Rich Gore from Cisco IT, to give a quick and simple overview of SIP, and how Cisco IT is using it to build new services, simplify architectures, and save money.
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The way that enterprises connect to the outside world is changing. The transition to voice over IP (VoIP) that began with enterprise networks a decade ago, is now in full force in service provider networks. In a report issued on Monday, Infonetics Research reported that Cisco, the global market leader for unified communications and collaboration, is now the new market leader in global enterprise session border control (SBC) solutions for the first half of 2012, providing secure IP connectivity from the enterprise edge to the service provider session initiation protocol (SIP) trunking service.
Why is this so important? Service providers are now offering SIP trunking services instead of legacy dial tone (also known as time-division-multiplexing or TDM) to connect to enterprises of all sizes, including small businesses. In fact, according to their 2012 VoIP and UC Services Report, Infonetics forecasts SIP trunks to grow over 66 percent in 2012 alone. Customers are quickly embracing the new technology, which offers substantial cost savings and the promise of extending real-time rich-media collaboration applications beyond the enterprise to customers, partners and suppliers.
To begin realizing the benefits of SIP trunking, businesses need to deploy a session border controller in order to efficiently and securely connect to service providers while preserving voice quality and features. Session border controllers connect IP networks and provide session control, security, demarcation for better troubleshooting and interworking to help overcome differences in the deployment of the SIP standard (such as CODEC or signaling).
Cisco reinvents the collaboration edge
Cisco’s session border controller, called, Cisco Unified Border Element (CUBE) is a software license add-on to the widely deployed Cisco Integrated Services Routers (ISRs) and Aggregation Services Routers (ASRs). CUBE provides significant benefits over competitors’ stand-alone session border controller offerings. For example, CUBE enables customers to transition more smoothly to SIP trunking while reducing costs and operational complexity, often requiring no new hardware to be purchased or deployed. As a result, CUBE has been adopted by over 5,000 customers in 160 countries.
In their report, Infonetics credited Cisco’s differentiated model for delivering SIP trunking service, stating: “This is a natural extension of Cisco’s dominant market position in the router market—the majority of organizations have Cisco routers already installed and deployed at the important network border points.”
Like many large enterprises, Cisco makes a lot of phone calls. Cisco previously used a lot of TDM trunks from multiple carriers to carry thousands of voice calls from our North American Cisco offices to the PSTN. The problem is, we had over 100 TDM trunks we were paying for every month, to carry our voice calls for these sites. Four years ago we started looking around for a more cost-effective and manageable way to support all these calls. After a good deal of searching, screening vendors and testing, we finally found it, using Session Initiation Protocol (SIP) trunking technology.
For the Cisco campuses in San Jose and Research Triangle Park (RTP), we will replace over a hundred PRI (23 channel) TDM trunks, used for long-distance voice calls for all of our North American sites, with SIP trunks. The new San Jose link is a 250 Mbps SIP trunk carved out of a 10 Gigabit Ethernet WAN access line, while the RTP link is a 20 Mbps SIP trunk carved out of a 45 Mbps DS3 WAN access line. Together, these SIP trunks give us the capacity to carry over 2400 concurrent calls and a total voice call volume of 2 million minutes per month. Read More »