Everything worth knowing, nothing that isn't.
How it works, end to end
A SIP trunk is a virtual connection that sets up calls with SIP signaling and streams the audio as RTP media. Your PBX, softswitch, or application registers to the trunk; when a call is placed, SIP negotiates the session (via an SDP offer/answer) and RTP carries the voice. A voice API wraps this so an HTTPS request starts a call and webhooks drive the flow as it progresses. See /answers/what-is-sip, /answers/what-is-rtp, and /products/voice.
SIP trunking vs a programmable voice API
A SIP trunk is raw connectivity — you bring your own PBX or softswitch and speak SIP directly, ideal for contact centres and carriers moving high call volumes. A programmable voice API abstracts SIP entirely behind REST and webhooks, ideal for developers adding click-to-call, IVR, or voice OTP without running telephony infrastructure. Most platforms, BroadNet included, offer both. See /answers/programmable-voice and /products/voice.
The building blocks
DID numbers for inbound calls; PSTN and mobile termination for outbound; concurrent-channel capacity (how many calls run at once) and a calls-per-second cap that pace bursts; a Session Border Controller policing the SIP edge; and codecs (G.711, Opus) encoding the audio. Each is provisioned per account and composed into a call flow. See /answers/what-is-did, /answers/what-is-sbc, and /answers/what-is-voice-codec.
Benefits over legacy PRI lines
No physical circuits to install, capacity that scales by adding sessions rather than cabling, one interconnect for 200+ countries, pay-per-use billing, and call flows changed in code instead of reprogramming a PBX. SIP trunking also consolidates inbound and outbound onto the same IP path, cutting cost and simplifying failover. See /products/voice and /tools/erlang-calculator.
Limitations and what to plan for
Voice over IP is sensitive to network quality — jitter, packet loss, and latency degrade call audio, so trunks need QoS and monitoring. NAT and firewall traversal must be handled at the SIP edge, capacity is bounded by concurrent channels and calls per second, and outbound dialing is constrained by caller-ID (CLI/STIR-SHAKEN) and Do-Not-Call rules. See /answers/voip-quality, /answers/what-is-nat-traversal, and /answers/what-is-cps.
Is a SIP trunk the same as a voice API?
No. A SIP trunk is the connectivity layer — it carries calls between your telephony system and the network using SIP and RTP. A voice API is a developer control layer on top that lets code place and manage calls without speaking SIP at all. You can use a SIP trunk without an API, but a modern programmable voice service gives you both. BroadNet delivers voice connectivity across 1,000+ operator connections at 99.99% uptime. See /products/voice.