This article is meant to provide a very basic black and white overview of the differences between PRI Circuits and SIP Trunking. This is by no means a thorough explanation, as there are many gray areas around both. For example, SIP can also be used to transmit video (not just voice communication). As such, if there is anything you’re unclear about, please email us – and we will personally answer your question!
PRI and SIP Trunking are two different ways of connecting your business to the PSTN (Public Switched Telephone Network). Typically both methods, PRI trunk and SIP Trunking, require a piece of an equipment in your office called a PBX (Private Branch Exchange). Most PBX and IP PBXs can support both PRI and SIP Trunking connections, either directly or with an adapter.
A PRI is a single line (typically a T1 connection in North America) with 23 voice channels (and a single data/control channel), that allows your business to hold 23 calls simultaneously. This is not always equivalent to the number of phone numbers a business has. Your business may have 100 phone numbers that are directed to come across a single PRI, however, you will only be allowed to hold 23 phone conversations at once.
In short, SIP Trunking is a method of sending your voice connection over an existing data line, and therefore is commonly referred to as VoIP (Voice over Internet Protocol).
Your call goes out over the SIP Trunk, is treated exactly like every other piece of data on that connection and is not given priority over webpages, emails, or instant messages.
Once it arrives at your carriers network over the SIP Trunk, your provider’s equipment knows where to direct that call; this may either switch it over to the PSTN (so that someone with a regular phone can receive the call) or perhaps send it to a PBX at another location. A single SIP Trunk is typically limited only by the amount of bandwidth on your data connection, or in some cases, limited by the number of “call paths” you are purchasing from your SIP Trunk provider.
This is typically as easy as calling your provider and purchasing more SIP Trunk virtual connections, or even just increasing your data connection’s bandwidth. (Dependent again on how your carrier is selling the SIP Trunk to you).
Since the PSTN’s main purpose is to handle voice connections, there is always a guaranteed bandwidth capacity and availability for phone calls. Voice calls through the PSTN will always take priority over any other type of potential traffic. Likewise, PRI Circuits always have a guaranteed amount of bandwidth set aside for each call that can never be used for anything else. SIP Trunking, however, cannot always guarantee this QoS. Since each voice connection is sent as a packet like any other piece of data, it is subject to lag, delay, high pings, or packet loss. These are not usually issues for websites or emails, since the server will continue sending the same packet repeatedly until your computer receives it. However, with the real time transmission of Voice, these packets can typically only be sent once. If the packet doesn’t make it, it may sound a lot like when someone is on their cell phone with poor coverage. Currently the only way to provide a QoS for a SIP Trunk is to purchase an MPLS, and the circuit from the same provider offering the SIP Trunk. Since you’re using the same carrier for all of the services, they can prioritize the voice packets for you.
PRI’s are becoming less relevant and compatible with newer technologies. Although PRI’s were once very reliable, SIP will allow flexibility for your business of any size and will save your business money in the long run.
