What is SIP Trunk?
Also called SIP trunking, IP trunk.
A SIP trunk is a virtual connection that carries phone calls between an organization's phone system and a voice service provider over an IP network. Session Initiation Protocol handles call setup and teardown while the audio itself travels as RTP media streams. It replaces the physical circuits that older systems used to reach the public telephone network.
SIP signaling negotiates who is calling whom, which codecs both sides support, and where media should flow, using request methods such as INVITE, ACK, and BYE. The negotiated audio then travels separately over RTP, often secured as SRTP with signaling protected by TLS. Because signaling and media take different paths, a call can fail in ways where one direction works and the other does not.
Capacity on a trunk is expressed in concurrent call paths rather than physical lines, and it can typically be adjusted without new cabling. Providers authenticate a trunk either by registration with credentials or by allowing traffic from specified IP addresses. Session border controllers commonly sit at the edge to handle authentication, topology hiding, protocol normalization between vendors, and rate limiting against toll fraud.
Voice quality on a SIP trunk depends on the underlying network. Jitter, packet loss, and one-way delay are the metrics that matter, and quality of service marking or a dedicated circuit is often used to keep voice traffic ahead of bulk data. Codec choice trades bandwidth against fidelity, with wideband codecs giving noticeably better speech clarity where bandwidth allows.
Regulatory obligations follow the trunk. Emergency calling requires an accurate registered service address per number or endpoint, number portability governs how existing numbers move between providers, and caller identity attestation frameworks apply in several countries. These requirements vary by jurisdiction and change over time, so they are configured per market rather than globally.
Key points
- SIP handles signaling; RTP carries the audio separately
- Capacity is measured in concurrent call paths
- Authenticated by registration or by permitted IP addresses
- Session border controllers normalize protocols and limit fraud
- Emergency address and number portability rules apply per market
In practice
A company connects its cloud phone platform to a carrier using a SIP trunk provisioned for sixty concurrent paths. Outbound calls leave through a session border controller that presents the main business number and enforces a per-minute rate cap. When a voice agent answers an inbound call, the signaling arrives as a SIP INVITE, the audio flows as SRTP, and the platform logs jitter and packet loss for each leg.