SIP - Session Initiation Protocol
What is SIP?
Imagine a communications environment where a central directory server not only knows how to reach an individual’s work phone, cell phone, and pager, but also their instant messaging (IM) program, e-mail, and PDA. Not only that, but also imagine that the central directory server also knows a party’s communication preferences and capabilities, and can intelligently alert a called party when someone is trying to reach them. Finally, imagine that phone calls to an unavailable person can be intelligently rerouted to another person or group depending upon a number of interrelated factors such as time of day, whether the called person is scheduled to be in a meeting, or whether one or more of their modes of communication is unreachable.
These capabilities aren’t some dream of a far-off utopian future, but are available today thanks to a remarkable advance in communications: Session Initiation Protocol (SIP).
SIP is the glue — and the intelligence — that makes these advanced communications capabilities possible.
Presenting Presence Services
SIP introduces a new model for communications through its support of presence. Presence enables you to locate a user and determine his willingness and ability to participate in a session, even before you initiate communications. This information, reflected across multiple devices such as IP phones, cell phones, and instant messaging clients, makes communication simple and efficient by helping you to reach the right person at the right time, on the right device.
Transforming trunking with SIP
SIP trunks enable enterprises to carry their voice data over a pure IP connection to carrier clouds, rather than through separate circuits as has been the practice for decades. An Enterprise SIP proxy peers with a Carrier SIP proxy, with the appropriate federations and security protections established between them. The IP circuit continues to carry e-mail, Internet, and other corporate traffic as it does today, and voice is simply layered on top of the circuit as another IP application. SIP sets up and tears down voice calls to and from the enterprise over this IP circuit.
On-net calls traverse the carrier’s VoIP backbone (which is typically dedicated to voice so that voice quality can be guaranteed). Off-net calls ride the carrier IP network until the last mile, where a gateway converts VoIP to TDM for calls to PSTN
- PSTN origination/termination
- Many SIP service providers support origination/termination services directly to the PSTN from their SIP networks. This practice enables the enterprise to reduce the monthly recurring costs associated with multiple TDM circuits by deploying a single IP pipe to the service provider network.
- Number Mobility
- These features take advantage of the fact that SIP is geographically agnostic. They allow calls destined to a local or national number to be ported to any number automatically rerouted over the service provider SIP network to another enterprise location. For enterprises, this system offers great flexibility in providing a local presence in all their markets while routing calls to a centralized call center for more efficient service.
- Cost savings
- For enterprises, SIP networking means reducing the monthly recurring cost of separate PSTN and data circuits to the premises. When you remove voice circuits, you reduce the number of TDM T1 interfaces on the IP PBX, because hundreds of VoIP calls can come from the same hardware footprint as a single T1 interface. Service providers may also offer reduced toll charges to customers when SIP is used as the interface to the PSTN.
