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

Retell builds and runs the voice agent — speech-to-text, the language model, text-to-speech, and the live-call orchestration. To reach the public phone network, it connects to a telephony provider. Retell supports two integration paths:

• Elastic SIP trunking — the recommended method, where you connect your own SIP trunk and point Retell at it.
• Dial to SIP URI — an advanced method for custom setups, where your own system dials Retell's SIP URI.

For most outbound use cases, elastic SIP trunking with your own carrier is the right path.

• Lower per-minute cost. Bundled minutes carry a markup. A direct carrier plus per-second billing reduces waste across high volumes of short outbound calls.
• Wider international reach. Your own carrier can offer competitive A‑Z termination where a default provider's coverage is thin.
• Caller-ID control. You set the caller ID (CLI) presented on outbound calls, within each destination's operator rules.
• Routing control. You choose route quality to match campaign traffic.

In Retell's dashboard you connect a number via SIP trunking and provide your trunk's termination URI (your SIP server) plus authentication. Two practical notes:

• Retell supports G.711 (PCMU/PCMA) and G.722, with SRTP when the transport is set to TLS — so you get a transcoding-free, encryptable path.
• Retell publishes specific SBC IP blocks, which means you can authenticate with either SIP credentials or IP whitelisting on the carrier side.

For the exact, copy-paste steps, see: Connect Retell to Bitcall (Outbound SIP Trunk).

• G.711 support for a transcoding-free audio path (lower latency, better transcription).
• Low-latency routing so the telephony layer doesn't add avoidable delay.
• Caller-ID (CLI) control and HLR number validation so you don't pay to dial dead numbers.
• Per-second billing and no contracts.
• Elastic concurrency for campaign spikes and A‑Z coverage where you call.

And one thing to expect: AI calls that drop after ~30–60 seconds are almost always an RTP inactivity timeout during the agent's thinking gaps — a configuration fix, not a carrier limit. See AI voice calls dropping after 30 seconds.

Bitcall provides the outbound layer for your Retell 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 dialing lists.

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

Ready? Follow Connect Retell to Bitcall (Outbound SIP Trunk).

Does Retell support bringing your own SIP trunk? Yes. Retell supports elastic SIP trunking (recommended) and a dial-to-SIP-URI method, so you can route calls through your own carrier instead of bundled telephony.

Does Retell need two SIP trunks? Retell typically uses one trunk for inbound and one for outbound. Since Bitcall is outbound-only today, you configure the outbound trunk now.

What codec should I use for Retell outbound calls? G.711 (PCMU/PCMA). It avoids transcoding in the audio pipeline, keeping latency low and transcription accurate. Retell also supports G.722.

Why do my Retell calls drop after 30 seconds? Almost always RTP keepalives aren't configured. Enable keepalives/comfort-noise and raise the RTP timeout to at least 60 seconds.