SIP trunking sounds like something one does on a vacation in warm climes with at least one fruity drink in hand. In actuality it’s more akin to those employees at the airport do who wave big light sticks at airplanes. (Their job title is actually Aircraft Marshall or Signaler.)
Before going too far, a roadmap should help shed some light on what’s covered below. Read on for answers to questions like: What are SIP and SIP trunking? How does SIP trunking differ from PSTN calling? What are the advantages of SIP trunking?
What is SIP?
Before discussing what SIP trunking, it’s necessary to flesh out what SIP is. SIP stands for Session Initiation Protocol. In essence, SIP is a method signaling and managing communications.
As described by TechTarget, Session Initiation Protocol (SIP) trunking is a service offered by a communications service provider that uses the protocol to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN).
Every group of friends has that one person who coordinates everything and is a master at getting everyone together. SIP is like that person, but instead of dealing with people it deals with IP-based applications, most notably voice, video, chat, and other multimedia communications.
SIP works by sending packets of information between SIP-enabled devices. There are two types of packets – signal and media. The signal packets establish and end a connection between two devices, which then allows for the exchange of media packets.
Of course, communications are dependent on both parties knowing where the other one is and an ability to understand what’s being said. SIP dictates both of these things by keeping track of device IP addresses, and laying down the rules for which codecs are appropriate for different session types so that all devices involved can process the media properly.
What is SIP trunking?
Whereas plain old SIP is like the aircraft marshal working one-on-one with a pilot on the tarmac, introducing the PBX makes SIP trunking more like the air traffic controller in the tower pushing tin between multiple planes at the same time.
SIP trunking is a VoIP-based media streaming service offered by Internet Telephony Service Providers (ITSP) that provides voice and unified communications to companies with a SIP-enabled private branch exchange (PBX). It basically puts all of a company’s IP-based communications tools in the same toolbox and lets anyone with access to that toolbox use them with each other.
How does SIP trunking work?
Like many things in telephony, the basic concept is pretty easy to grasp, but the actual inner-workings of SIP trunking can be extremely complicated. Staying true to form, this space will cover basic SIP trunking functionality, but suffice it to say this just scratches the surface of what SIP is capable of doing.
The difference between yesterday…
Suppose a company has an office in Los Angeles and another location in Philadelphia. The two offices are connected by a Wide Area Network (WAN) for their IP needs. Each office also has a dedicated PBX for telephone calls that connects to the rest of the PSTN along a Primary Rate Interface (PRI) trunk.
If a customer calls the company’s 800-number it rings the LA office. But say all the lines in LA are busy. What happens to the call? Most likely it gets a busy signal or sits in a call queue until it’s picked up. Which one depends on how many PRI trunks the company leased for its LA office, as each trunk only has 23 channels, or in layman’s terms, is capable of handling 23 concurrent calls. Regardless, whether it’s a busy signal or waiting on hold neither option is ideal for the customer.
If the company needs more channels (i.e. concurrent calls) they need to lease additional PRI trunks, potentially a very costly operating expense. These lines connect directly to a specific PBX so if a call needs to be re-routed it’s up to a person to physically transfer the call.
In a SIP trunking world those PRI trunks are eliminated. Instead, the PSTN connects directly to a router with Multiprotocol Label Switching (MPLS) capabilities that resides on the company’s WAN. As “multiprotocol” suggests, MPLS can handle a variety of different protocols, including SIP.
Also, remember that 23 concurrent call threshold a PRI trunk has? Well SIP trunking doesn’t abide by a fixed number of channels. Instead, the number of concurrent calls depends on available bandwidth. Using compression can make SIP trunking even more bandwidth efficient, too.
Now, replay that same call scenario with SIP trunking. The call moves from the PSTN directly to the company’s MPLS on its WAN network. Using SIP, the router sees that all of the lines in LA are down due to rolling blackouts, but finds an available one in Philadelphia and automatically re-directs the call there. The caller connects with someone right away, creating a better customer experience.
Benefits of SIP trunking
One of the most notable benefits to utilizing SIP trunking is that it typically corresponds to a sizable cost reduction. Eliminating the cost of leased PRI lines is much greater than paying for an uptick in bandwidth. Further, as SIP trunking service fees continue to decline, companies should sustain cost savings in the long-term. It’s not unheard of for a company to halve their telephony costs by switching to SIP trunking.
The bandwidth-to-available-channels relationship means that SIP trunking scales easily, unlike legacy systems. Also, given the fact that SIP trunking consolidates a company’s voice technology at the enterprise level, the need to allocate resources by location exits stage left too. This makes it easier to compensate for call spikes, whether anticipated or not.
As the example above also illustrates, SIP trunking has built in disaster relief capabilities because it can detect which lines are available throughout the entire network and dynamically route calls to available lines.
Because SIP trunking combines voice and data on the same lines and plays nice with unified communications, which typically includes voice, video, instant messaging, and even applications for web conferencing and real-time collaboration, companies can get a lot more use out of the technology than just cheaper phone calls.
The IP-based nature of SIP trunking makes collaboration across multiple locations or with mobile devices easier. This doesn’t have to be limited to two people either, with SIP and unified communications multiple parties in separate locations can talk, web conference, or share screens in real-time. This creates a more flexible, efficient workforce.