SIP Trunks for a New Call Center: Channels, CPS, DIDs and Failover
A SIP trunk connects your dialer or PBX to the public telephone network over the internet. For a call center the questions that matter are capacity (channels and calls per second), route quality for your destinations, caller-ID rules, billing increments and what happens when a route fails.
Outbound termination and inbound DIDs
- Outbound (termination): your dialer sends calls to the carrier, which delivers them to phone numbers in the destination country. Billed per minute by destination prefix.
- Inbound (origination): customers call a DID (phone number) that the carrier routes to your PBX or IVR. Usually billed as a monthly number rental plus inbound minutes.
- Many call centers use different providers for the two; you do not need to buy both from the same carrier.
Channels vs. CPS
Channels (concurrent calls) is how many calls can be in progress at once. CPS (calls per second) is how many new call attempts can start each second. They are independent limits: a predictive dialer can hit a CPS limit at campaign start while using few channels, or run out of channels during long calls at low CPS.
SIP channels and bandwidth calculator (requires JavaScript). Formulas used: peak outbound channels = agents × concurrent-calls-per-agent factor; inbound channels are added separately; channels with headroom = ceil(total × (1 + headroom%)); per-call bandwidth per direction = (codec payload bytes per packet + 40 bytes IP/UDP/RTP + 18 bytes Ethernet) × 8 × packets per second; total bandwidth = channels with headroom × per-call bandwidth. CPS is estimated separately from call attempts per minute ÷ 60.
Routing, IVR and Press 1
- Ordinary outbound termination carries a live call between your agent and the called person.
- Press 1 / IVR campaigns play a recorded message and connect the person to an agent only if they press a key. They depend on reliable DTMF detection and are subject to stricter robocall and consent rules in many countries — see Press 1 routes.
- Inbound IVR menus route callers to queues; keep them short and include required recording notices.
Caller ID (CLI)
Present a caller ID you own or are authorised to use, in the format the route expects. Using numbers you are not authorised to present (spoofing) is unlawful in many jurisdictions and is prohibited by CallCenterBoost's Acceptable Use Policy. Whether a route reliably passes caller ID through to the destination varies — see CLI and NCLI routes.
Testing a route
Test before committing traffic: connectivity, answer rates, post-dial delay, audio quality, DTMF and caller-ID presentation on your own permitted numbers. The route-testing guide gives a three-stage plan.
Failover
- Configure a second carrier with lower priority so calls overflow when the primary rejects them or times out.
- Send a small share of traffic over the backup route regularly so you know it works.
- Check that caller ID, codec and DTMF settings work on both routes.
- Compare billing increments on both routes — a backup with 60/60 billing can cost much more for short calls.
Estimate calling costs with published rates
Calling cost estimator (requires JavaScript): enter a prefix, average duration and number of calls; it uses the currently published rate and its exact billing increment.
How CallCenterBoost provisions SIP
Nothing in this flow is instant or automatic, and no route, capacity or quality figure is guaranteed in advance.
- Search published rates for your destinations.
- Request a route test with destinations, expected minutes, concurrency, CPS and your dialer.
- The team reviews the request manually and confirms availability and test terms.
- Authentication is set up with SIP credentials or your approved fixed IP.
- Concurrency and CPS are agreed for your service.
- After testing, the account is funded using a currently accepted payment method and traffic is ramped gradually.
Frequently asked questions
How many SIP channels do 10 agents need?
About 10–15 for inbound or manual dialing; 20–40 at peak for a predictive dialer. Use the calculator, then confirm in a route test.
What CPS does a small call center need?
Often only a few calls per second — but predictive campaigns can burst at the start of a shift. Estimate from your peak call attempts per minute and agree the limit with your carrier.
Which codec should I use?
G.711 is the safest default for PSTN interconnection and audio quality; it uses more bandwidth than compressed codecs. See [VoIP codecs compared](/help/voip-codecs-g711-g729-opus).
Next steps
Next guide: Hiring, Training and Managing Call Center Agents. Get a Setup Quote · View VoIP Rates · Request Route Test
Related pages
- Start a call center: overview
- Startup cost calculator
- Live VoIP rates
- Request a route test
- Contact sales
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