Why SIP Trunking
Initially, the ubiquitous Internet was used only for data communications between nodes on the network using the Internet Protocol or IP, the communications protocol for relaying datagrams across network boundaries. With advances in digital communications, analog calls can now be converted into packets of data that can travel like any other type of data, such as e-mail, over the public Internet and/or any private IP network, resulting in voice over IP (VoIP).
This VoIP communications is also known as Internet or IP Telephony. As a result, Internet today provides both data and voice communications, including video. Many service providers today have combined all these services into a single package called Unified Communications (UC). Now organizations small, medium or large, that plan to switch from traditional Public Switched Telephone Network (PSTN) to VoIP must replace conventional private branch exchange (PBX) with IP enabled PBX or IP-PBX. For that, they must deploy SIP Trunking.
For those who are not familiar with SIP, known as Session Initiation Protocol, it is an application layer control or signaling protocol used for creating, modifying, and terminating media sessions with one or more participants over IP networks. These media sessions typically are VoIP calls. But, it can also include instant messaging, presence, and video conferencing.
SIP is similar to other widely used protocols such as HTTP and SMTP in that it provides a flexible, open standard that can be leveraged efficiently with a variety of different UC communications systems and technologies.
Moving on to SIP trunking, it is a VoIP and streaming media service based on the SIP protocol by which Internet telephony service providers (ITSPs) deliver telephone services and UC communications to customers equipped with SIP-based private branch exchange (IP-PBX) and UC facilities. It offers a modern alternative to traditional TDM trunks. According to marketing experts, SIP Trunking will eventually supplant traditional circuits to become the leading mechanism for voice traffic between an enterprise and the PSTN. As such, it will eliminate the need for the fixed PSTN lines, such as PRI (primary rate interface), BRI (basic rate interface) and/or POTS (plain old telephone system) and send all voice traffic over the existing data network to the service provider who then routes it to the PSTN.
Benefits of SIP trunking, by comparison, SIP Trunking offers numerous benefits over traditional circuits. Some of the notable advantages are convergence, centralized access, cost reduction, disaster recovery-business continuity, quality of service, geographic number portability and future growth. Besides enabling a business to combine voice, internet and networking services, SIP trunking also allows main and branch offices to share hardware and access at the primary site.
By eliminating the need to maintain traditional circuits at each site, the cost savings are tremendous. In addition, it leverages the already existing data network for voice.
Furthermore, SIP trunking gives you the ability to easily redirect individual direct inward dial (DID) numbers from one SIP trunk to another. In other words, inbound phone numbers are not confined to any single location or local exchange carrier (LEC). Hence, when an office relocates to another location with a different LEC, those numbers can be easily ported to the new network.
And finally, in terms of the future growth, SIP trunking gives you the much needed flexibility and scalability.