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SIP TRUNKING

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SIP (Session Initiation Protocol) Trunking is a service offered by many ITSPs (Internet Telephony Service Providers). It connects a company’s IP-PBX to the existing PSTN (Publicly Switched Telephone Network) infrastructure, either via the internet or a private MPLS network, and uses bandwidth from the IP connection to provide local and long distance voice service. In effect, the Internet replaces the conventional telephone trunk. By allowing businesses to send voice and data traffic, both internally and externally, over the internet instead of the regular telephone network, it circumvents traditional VoIP applications, which have to manipulate internal IP communications before they can travel – as voice communications – over phone lines.

 
Three components must be present to successfully deploy SIP trunks: a PBX with a SIP-enabled trunk side, a gateway that serves as the interface between the PBX and the ITSP, and an Internet telephony or SIP trunking service provider.

 
The SIP’s “direct connection” to the internet translates into substantial cost savings over traditional telephone services by reducing the number of physical trunks to the phone company. It also provides for increased efficiency on your Wide Area Network connections when a service known as “Trunk Pooling” or “Direct Trunk Overflow (DTO)”, provided by some carriers, enables all of your locations across the enterprise to share a determined number of SIP trunks. Trunk Pooling (or DTO) eliminates the situation where one location has lines busy and another site has lines available. Vis-a-vis ‘the cloud’, calls can get redirected to other locations, which not only ensures efficiency but also business continuity by guaranteeing continuous uptime in a disaster recovery scenario.

 
Much like a PRI/T1 solution, the SIP protocol makes transmitting Caller ID and other call-related information easy. In the same manner, there is no need to limit the telephone number to 10 digits. This makes it possible to assign Direct Inward Dialing (DID) numbers for every extension, so that the auto attendant in the PBX plays a less important role. Conversely, instead of having a PRI/T1 for voice only, which may never be fully utilized or may outgrow its capacity, you could have an IP connection to the SIP/ITSP carrier for internet, MPLS, Local, Long Distance and Toll Free service, all of which can be seamlessly accommodated through incremental increases in bandwidth. This ability to have a set of unified communication tools converging on one high-speed IP access line makes SIP trunking a much more flexible and scalable alternative.

 
It’s important to note, however, that while SIP Trunking, in lieu of installing and maintaining a PRI system at each location, is a more cost-effective solution, it almost always comes at the expense of the reliability of a traditional PSTN connection. Although issues from analog lines like connect and disconnect detection are simple, the challenge with SIP connections over the public internet is call quality. Because SIP connections are digital, if the traffic runs on the same connection with other traffic like e-mail or web, voice and even signalling packets may be dropped and the voice stream can get interrupted, resulting in “choppy” conversations.

 
Businesses must, therefore, understand the potential drawbacks when they want to use SIP connections as their primary PSTN termination. Also, the general stability of the internet connectivity itself becomes a critical issue. Because of this, many companies split voice and data up into two separate internet connections to solve this problem, so that the resource conflict on the internet access side is avoided.