SIP trunks explained

A SIP trunk is a virtual phone line service that connects your phone system to the public telephone network over an IP connection. It replaces bundles of copper lines or PRI circuits, lets you add capacity without new wiring, and can carry calls for several sites from one place. It works with on-premises phone systems and with many hosted systems that bring their own carrier connection.

What a SIP trunk does

Your phone system handles internal calls, extensions and features. To reach outside numbers, it needs a connection to a carrier. Traditionally that was analog lines (one call per line) or a PRI circuit (a fixed number of channels over a dedicated line). A SIP trunk does the same job using the SIP protocol over a data connection, which can be your internet link or a dedicated private circuit from the carrier.

A SIP trunk service usually includes:

• A number of concurrent call paths, sometimes called channels or sessions
• Direct inward dial (DID) numbers, so each user or department can have its own number
• Calling rates or plans for long distance and international calls
• 9-1-1 service tied to registered addresses

Sizing a SIP trunk

Call paths

You need enough call paths for your busiest period, not one per employee. Look at call records from your current system for the peak number of simultaneous calls, then add a margin for growth and unusual days. Remember inbound and outbound calls share the same paths, as do fax and other devices routed through the trunk.

Bandwidth

Each call uses bandwidth in both directions. An uncompressed G.711 call needs roughly 80 to 100 kbps each way including network overhead. Multiply by the peak concurrent calls and make sure the link has room for that on top of everything else. On a shared internet link, quality of service and traffic shaping at the edge protect calls from other traffic; see QoS for voice and video.

A worked sizing example

Suppose call records from the current system show that on the busiest days, the most simultaneous outside calls at any moment was 18. A reasonable trunk size might be about 24 call paths, which covers that peak with room for growth and a busier than usual day. If every call uses G.711 at roughly 100 kbps each way including overhead, 24 calls need about 2.4 Mbps in each direction reserved for voice. That figure goes into the QoS design at the internet edge, and the rest of the link must still carry everyone's other traffic at the busy hour. If the site uses a fax machine or alarm communicator through the trunk, count those as calls too.

Codecs and compatibility

The phone system and the carrier must agree on codecs, number formats, fax handling and how features such as caller ID and call transfer are passed. Carriers usually publish a list of tested phone systems and configuration guides. Using a tested combination saves a lot of troubleshooting.

SIP trunk vs traditional lines

Security and the session border controller

Exposing a phone system to the internet invites attack. Toll fraud, where criminals use a compromised phone system to make expensive international calls, is a real and costly risk. Common protections include:

• A session border controller (SBC), or a firewall with proper SIP handling, between the phone system and the carrier
• Restricting SIP traffic to the carrier's known addresses
• Strong passwords and limited access on phone system administration and voicemail
• Blocking or requiring authorization for international and premium-rate calling
• Spending limits and fraud alerts from the carrier
• Regular updates to phone system software

A cutover checklist

Moving to a SIP trunk is usually done on a planned date, once numbers have been ported. A practical checklist for the day, with results recorded:

• Trunk registered and passing test calls before the port date, using temporary numbers
• Inbound calls to the main number, a sample of DIDs and every hunt group reach the right place
• Outbound local, long distance and toll-free calls connect, with the correct caller ID shown
• 9-1-1 address records confirmed with the carrier for every site, and a test call arranged following the carrier's and local procedure
• Call transfer, conference and voicemail work across the trunk
• Fax, alarm and other special devices tested, or moved to a suitable alternative
• Failover forwarding tested by simulating a site outage
• International and premium-rate restrictions confirmed in place
• Old lines kept active until the port completes, then cancelled in writing

Keep the carrier's support contact and the porting order details at hand on the day. Porting problems are much easier to fix while the carriers involved are still working on the order.

9-1-1 and number porting

Because a SIP trunk is not tied to a physical location, the carrier relies on the addresses you register. Every site and number must have the correct civic address, and multi-site systems must send emergency calls out with the right location for the caller. Confirm how your carrier handles 9-1-1 and the current CRTC requirements. When moving numbers from an existing carrier, porting takes coordination and lead time; plan the cutover date around it. See moving from landlines to VoIP for the full transition.

How ehZee can help

Through our VoIP design and build service we work out what capacity you actually need, check the network, configure the connection to the carrier with security and failover in mind, and test calls before and after cutover.

Common questions