SIP trunk explained: channels, codecs and cost

A SIP trunk is a virtual phone line that connects your phone system to the public telephone network over an internet connection. Instead of physical analog or ISDN lines, your PBX or softswitch sends and receives calls as SIP traffic through a provider, which hands them over to the rest of the phone network. You pay for the capacity and the calls you use, keep your phone system, and can add numbers and channels without new cabling.

How a SIP trunk works

Three parties take part in every call:

1. 1Your phone system: an on-site PBX, a cloud PBX you run yourself, or a softswitch if you are an operator. It knows your users, extensions, IVR menus and queues.
2. 2The SIP trunk provider: it accepts your calls, routes them to their destination and sends you the calls made to your numbers.
3. 3The public phone network: landlines and mobiles everywhere, reached through the provider's interconnections with other operators.

Each call has two parts. The signaling, in SIP messages, sets the call up and ends it: your PBX sends an INVITE for the number dialed, the provider answers with 100 Trying, 180 Ringing and 200 OK when the person picks up. The voice itself travels separately as RTP packets, encoded with a codec both sides agreed on. For incoming calls, the same happens in the other direction: the provider sends the INVITE for one of your numbers (DIDs) to your PBX, which rings the right person or queue.

SIP trunk, ISDN line or cloud PBX?

ISDN and analog lines are reaching their end. In France, new analog lines stopped being sold in 2018 and new ISDN lines in 2019, and the copper network closes in stages until 2030; other European operators have already closed their ISDN networks. A SIP trunk is the natural replacement when you want to keep the PBX you already own. If you would rather not run a phone system at all, a cloud PBX is the alternative.

How many channels do you need?

A channel is one simultaneous call. A trunk with 10 channels can carry 10 calls at the same moment, inbound and outbound together; the 11th call gets a busy signal or waits, depending on the setup. The number of users does not matter, only the number of calls at the busiest moment.

To size a trunk:

• Office use: look at the busiest hour in your call logs or call detail records and count the peak of simultaneous calls. Without data, many offices start with one channel for every three to four users and adjust.
• Call centers: count one channel per agent who can be on a call at the same time, plus the inbound calls waiting in a queue or an IVR, which also hold a channel. Predictive campaigns need more: the dialer places more calls than there are agents.
• Growth and peaks: add a margin for seasonal peaks, mailings or campaigns, rather than sizing for an average day.

Example: a 25-agent outbound team that peaks at 25 calls in progress, with up to 10 callers waiting in the inbound queue at the same time, needs about 35 channels.

Bandwidth and codecs

Each call uses bandwidth in both directions, depending on the codec:

Multiply by your channels to size the connection: 35 simultaneous calls in G.711 need about 3 Mbit/s up and 3 Mbit/s down, plus headroom for everything else on the link. Bandwidth alone is not enough, though. Voice is sensitive to latency, jitter and packet loss, so give voice traffic priority on your router (QoS) and avoid running a call center on a saturated or unstable connection.

Both ends must share at least one codec. When they do not, the platform has to transcode between them, which adds delay. Keypad tones (DTMF) also deserve a check: compressed codecs can distort them, so DTMF is best sent as separate events (RFC 4733).

Connecting your PBX: IP authentication or registration

Providers accept your traffic in one of two ways:

• IP authentication: the provider allows your PBX by its public IP address. Nothing secret is stored on the device, and it suits operators and sites with a fixed IP.
• Registration: your PBX logs in with a SIP username and password and renews its registration regularly. It works behind a changing IP address, at the cost of credentials that must be protected.

A few settings cause most first-day problems: the SIP port (usually 5060, or 5061 for TLS), the range of UDP ports used for audio, and NAT on the office router. One-way audio almost always comes from NAT or a firewall blocking the RTP ports. Larger setups put a session border controller (SBC) between the PBX and the trunk to handle security, NAT and codec conversion in one place.

Security: protect the trunk against toll fraud

A compromised PBX can place thousands of expensive international calls in a night, and the bill goes to the trunk's owner. The basics:

• restrict access to your PBX to the provider's IP addresses, and never expose its admin interface to the internet;
• use long, unique SIP passwords and change default accounts;
• block the destinations you never call and set spending or channel limits;
• keep the PBX software up to date and watch for unusual calls at night or at weekends.

Phone numbers on a SIP trunk

Your numbers (DIDs) are routed to the trunk, and your PBX decides who answers. A trunk can carry numbers from several countries, so a team working from one place can answer customers on local numbers in each country. Regulators usually require a link with the country: in France, Belgium and the other countries OVOKY serves, numbers go to companies with an address in the country or a contract with a client based there.

To keep existing numbers, you have two options. Porting moves the number itself to the new provider, when both operators support it. Forwarding keeps the old line with its current provider and sends its calls to a new number on the trunk, which is quicker to set up and avoids a cut-over date.

What drives the cost of a SIP trunk

Prices vary a lot between providers and offers, but the same items make up the bill:

• Channels: the number of simultaneous calls, often a monthly fee per channel, or included in a bundle.
• Calls: outbound minutes by destination, billed per minute or per second; landline, mobile and international rates differ. Incoming calls on standard numbers are usually free for the receiver.
• Numbers: a monthly fee per number, which depends on the country and the number type.
• Setup and support: some providers charge for activation, a dedicated link or premium support.

Compare offers on your real traffic, not on the headline price: how many channels at the peak, how many minutes to which destinations, and how many numbers in which countries. Billing increments matter for short calls; per-second billing avoids paying a full minute for a 10-second call.

A checklist for choosing a provider

• Authentication method that suits your equipment (IP or registration).
• Codecs your PBX supports, and how DTMF is sent.
• Channels you can add or remove quickly when traffic changes.
• Numbers in the countries you need, and the documents required to get them.
• Failover: what happens to incoming calls when your PBX or internet link is down.
• Fraud protection: destination blocking, limits, alerts.
• Clear billing: increments, rates by destination, fees for numbers and channels.
• For operators: the option to resell trunks under your own brand.

OVOKY's SIP trunk

OVOKY's SIP trunk is built for VoIP operators and resellers, and for businesses connecting a PBX:

• connection by IP authentication or by registration with a SIP username and password;
• G.711, G.729 and Opus;
• channels sized on request, from a few simultaneous calls to high volumes;
• calls to destinations worldwide, and new phone numbers in France, Belgium, Switzerland, Spain and Canada, routed straight to your trunk;
• failover routing: incoming calls follow a backup route when your PBX or link is down;
• white-label resale of trunks through the hosted softswitch.

Number porting is not offered: ask your current provider to forward your existing lines to new OVOKY numbers. Trunks are quoted on the channels, destinations and numbers you need.