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Glossary

SIP trunk

A SIP trunk is a phone line from a carrier that runs over an IP network using SIP instead of a copper phone line or a T1 line. Calls travel over it between a PBX or contact center platform and the carrier. Numbers and concurrent channels are added without new hardware.

Why it mattersYou add numbers and lines by asking your carrier – no new hardware.

Also called: SIP trunking, SIP line, IP trunk

How it works

How it works

The carrier gives you a SIP login and password or accepts calls from your IP address. The contact center's PBX or SIP proxy connects to the carrier and sets up calls through it over SIP, while the voice travels as separate RTP streams.

Trunk capacity is counted in concurrent channels: one channel carries one call. A 30-channel trunk handles 30 calls at once, inbound and outbound combined. The 31st caller hears a busy tone.

  • Signaling: SIP over UDP, TCP, or TLS
  • Voice: RTP or encrypted SRTP, with G.711, G.729, or Opus codecs
  • The carrier routes your numbers to the trunk, and your phone system distributes the calls from there

Example

How many channels you need

A channel is busy for as long as the call is on the line: while the customer talks to an agent and while they wait in the queue or listen to the IVR menu. So you need more channels than agents.

For example, at the peak hour there are 20 agents on the line, plus up to 10 calls waiting in the queue and the IVR at the same time. The trunk needs at least 20 + 10 = 30 channels. With an auto-dialer the system dials more numbers than there are free agents, so add headroom for that too.

The Erlang calculator shows how many agents you need at the peak from call volume and talk time.

Use

Where it is used

Contact centers use SIP trunks for inbound lines, toll-free numbers, and outbound campaigns. You can connect trunks from several carriers at once: if one fails, calls go through another, and calls to different destinations go through whichever carrier is cheaper for them.

  • Inbound lines and hotlines
  • Outbound campaigns through an auto-dialer
  • Backup: failover between two or more carriers

Practice

What to check before connecting

  • Concurrent channels at the peak hour, not the daily average
  • Which codecs the carrier supports and whether voice will need transcoding
  • Connection method: login and password or IP address
  • Encryption: TLS for signaling, SRTP for voice
  • Which number the customer will see on outbound calls

In CONTACT.WORK

How it works in CONTACT.WORK

The platform connects carrier SIP trunks through a Kamailio SIP proxy to Asterisk or FreeSWITCH. Inbound calls, IVR menus, and the auto dialer in all three modes share the same trunks, both in the cloud and on the customer's servers. You can keep your carrier and your numbers.

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