SIP trunking replaces the physical lines that used to connect a phone system to the outside world with channels carried over your internet or fiber circuit. The phone system, the handsets, and the way people work stay the same; only the connection to the phone network changes.
A PRI circuit provided 23 voice channels in a fixed block, so adding capacity meant adding another circuit. SIP channels are a software quantity — you buy the number of concurrent calls you need and adjust it without any physical change. Sizing should be based on concurrent calls at the busiest hour, which is nearly always far fewer than headcount.
A SIP-capable interface. Modern systems have one; older ones need a session border controller between the PBX and the trunks, which also handles security and protocol normalisation. Confirming compatibility before ordering avoids discovering it during a cutover.
Voice needs modest bandwidth and low jitter. Each concurrent call consumes roughly 85 to 100 kbps with common codecs including overhead, so capacity is rarely the constraint — prioritisation is. If voice competes with a large upload on an unmanaged connection, calls degrade, and no amount of extra bandwidth reliably fixes it.
Porting existing numbers is usually the critical path in a SIP cutover, and it moves on carrier timelines rather than yours: weeks, not days, for anything sizeable. Emergency calling is the other item that must be settled before turn-up, because current US rules require direct 911 dialling without a prefix and a dispatchable location accurate enough to find a caller inside a large building. Both are per-site configuration decisions.
For organizations still paying for PRI, SIP trunking is usually the clearest cost reduction available in telecom, with no change to how staff work. It also removes the block-of-23 constraint that made capacity planning awkward.
SIP Trunking is one of the specifications these comparisons turn on.
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