Vapi Outbound Calling: How to Use Your Own SIP Trunk (and Why)

Vapi is a platform for building and deploying voice AI agents: it handles the speech-to-text, the language model, and the text-to-speech, plus the orchestration around a live phone call. To actually place a call on the public phone network, Vapi connects to a telephony provider.

You have two options:

• Vapi's bundled telephony, which is quick to start but typically resold and marked up.
• Your own SIP trunk (BYOC), where you connect a carrier directly to Vapi.

For anything beyond a prototype — especially outbound campaigns at volume — most teams move to their own trunk.

Four practical reasons, all of which matter more as call volume grows:

• Lower per-minute cost. Bundled minutes include a markup over wholesale termination. Connecting a carrier directly removes that layer. With per-second billing, the savings compound across thousands of short outbound calls.
• Better international coverage. Your own carrier can give you competitive A‑Z termination in regions a default provider serves poorly.
• Caller ID control. You decide the caller ID (CLI) presented on outbound calls, within each destination's operator rules.
• Routing control. You choose route quality (for example, CLI vs NCLI vs call-center routes) to match your traffic.

At a high level, Vapi uses two resources:

1. A byo-sip-trunk credential that tells Vapi how to reach your carrier — the SIP gateway address and your authentication.
2. A byo-phone-number resource that, for outbound calls, acts as the caller ID (from-number) the agent presents.

One important detail: Vapi recommends username/password authentication rather than IP-based auth for shared SIP gateways, because IP auth can misroute when many customers share the same servers. With a carrier like Bitcall, username/password is the clean path.

For the exact, copy-paste setup, see the step-by-step guide: Connect Vapi to Bitcall (Outbound SIP Trunk).

Outbound AI calling is sensitive to the telephony layer. Look for:

• G.711 support so there's no transcoding in the audio pipeline (lower latency, better transcription).
• Low-latency routing so RTP doesn't add avoidable delay to the agent's response time.
• Caller ID (CLI) control and, ideally, number validation (HLR) so you're not paying to dial dead or ported numbers.
• Per-second billing and no contracts, so cost tracks real usage.
• Elastic concurrency for campaign spikes.
• A‑Z coverage in the countries you actually call.

And a reliability note worth knowing in advance: AI calls that drop after ~30–60 seconds are usually an RTP inactivity timeout during the agent's "thinking" gaps — a configuration fix, not a carrier limit. We explain it here: AI voice calls dropping after 30 seconds.

Bitcall is a self-service telecom platform that provides the outbound side for your Vapi agent — A‑Z termination, routing, and caller ID:

• From $0.01/min with per-second billing to 195+ countries, with strong CLI/NCLI routes across Africa, the Middle East, Asia, and Europe.
• No contracts, no monthly minimums; prepaid top-ups by card, PayPal, or crypto.
• SIP-native with SRTP/TLS, plus HLR Lookup to clean your dialing lists.

Scope today: Bitcall powers outbound calling. Inbound numbers (DIDs) are on the roadmap, and we'll cover inbound Vapi setup when it goes live.

Ready to connect it? Follow Connect Vapi to Bitcall (Outbound SIP Trunk).

Does Vapi support bringing your own SIP trunk? Yes. Vapi supports a custom "BYO" SIP trunk, so you can route outbound calls through your own carrier instead of Vapi's bundled telephony.

Is a BYO SIP trunk cheaper than Vapi's built-in numbers? Usually, yes. Bundled minutes carry a markup over wholesale termination, and per-second billing reduces waste on short outbound calls. The savings grow with volume.

What codec should I use for Vapi outbound calls? G.711, in nearly all cases — it avoids transcoding in the audio pipeline, which keeps latency low and transcription accurate.

Why do my Vapi calls drop after 30 seconds? Almost always RTP keepalives aren't configured. During the agent's response gaps, the media stream goes quiet and a device times it out. Enable RTP keepalives and raise the RTP timeout to at least 60 seconds.