OpenAI’s Realtime API documentation explicitly describes connecting phone calls through SIP or a server-side audio bridge, with a telephony provider supplying the phone connection. Google’s Gemini Live API documentation describes WebSocket-based live audio and lists a partner integration for inbound and outbound calls, but the reviewed pages do not document a Google-issued number or direct SIP endpoint. The third model implied by the original title is not identified in the available material, so an exclusive three-model comparison cannot be verified.
Which AI voice models can connect to a phone line?
Of the two platforms whose official documentation can be identified here, OpenAI documents a direct SIP route into its Realtime API and an alternative audio-bridge architecture. Google documents Gemini Live API access over WebSockets and lists telephony integrations, including Voximplant. Those are different levels of evidence: OpenAI’s guide explains a phone-call connection path, while Google’s overview points to an integration rather than describing a Google-native phone endpoint.
“Native path” needs qualification. In both architectures, a model API is not itself a telephone carrier: a telephony provider or integration supplies the phone number and connection to the public telephone network. The distinction is whether the model vendor documents how call audio reaches its API, and which party handles the signaling and call setup.
| Platform | Documented route | What the documentation establishes |
|---|---|---|
| OpenAI Realtime API | SIP or a server-side audio bridge | OpenAI’s telephony guide explains inbound and outbound calling and says a SIP trunking provider is required to connect a number. OpenAI Realtime SIP guide. |
| Google Gemini Live API | Stateful WebSocket sessions; telephony partner integrations are listed | Google documents live audio over WebSockets and describes Voximplant as connecting inbound and outbound calls. The reviewed pages do not show a Google-issued number or direct SIP endpoint. Gemini Live API overview. |
| Third model implied by the title | Not established | The model is not named in the available documentation, so a comparison or ranking across all three cannot be verified. |
How OpenAI’s documented phone-call path works
OpenAI’s Realtime telephony guide gives two options: connect a SIP trunk to the API, or bridge call audio through a server-side application. In the SIP arrangement, the telephony provider exchanges call audio with OpenAI; the application remains responsible for webhooks, session configuration, call decisions and business logic. OpenAI states: “If you want to connect a phone number to the Realtime API, use a SIP trunking provider (e.g., Twilio).” The guide also covers accepting inbound calls and setting up outbound calls, subject to provider and organization configuration.
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The phone number, trunk and telephone-network connection therefore come from the telephony provider, not from the voice model. OpenAI’s broader Live API guide also lists telephony/SIP alongside WebRTC and WebSockets as connection choices, and points to integrations such as Twilio, Telnyx and Daily/Pipecat. OpenAI Live API guide.
How Gemini Live connects to live audio and calls
Gemini Live is documented as a real-time, bidirectional API accessed through stateful WebSocket sessions. Google’s SDK guide shows connecting to a Live session and sending and receiving audio; its API reference describes sessions exchanging audio, text, video and function-call requests. These are model-session transports, not evidence by themselves of a phone number or public telephone-network endpoint.
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- Google’s SDK guide demonstrates Live sessions and audio exchange.
- The Live API reference documents the WebSocket session and supported exchange types.
- The overview lists integrations and describes Voximplant connecting inbound and outbound calls to Live API.
That partner listing makes a phone-call integration a documented option, but it does not establish that Google provides the phone number or a direct SIP endpoint. Google’s model catalog identifies Gemini 3.8 Live as the default option for most low-latency voice-agent experiences; model names and lifecycle status can change, so check the catalog when choosing a deployment model. Google Gemini model catalog.
What “connect to a phone line” means in practice
A working phone-based voice agent spans several components. Do not treat live audio support, a telephony partner listing and a carrier connection as interchangeable:
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- Phone number and public network: supplied by a carrier or telephony provider.
- Call signaling and routing: handled through the provider’s trunk or integration.
- Audio transport to the model: SIP, an application-managed bridge or a partner-supported route, depending on the documented architecture.
- Model session and behavior: configured by the application, including session settings, tool/function handling and business logic.
OpenAI’s separate consumer ChatGPT voice feature should not be confused with its public API telephony route. The Help Center describes voice in ChatGPT app experiences, not a public phone-number integration. ChatGPT Voice Mode FAQ.
How to choose and verify a route
- Identify the exact model and API. Confirm whether the intended product is an API for developers or a consumer app voice feature.
- Choose the telephone provider. Verify that it supplies numbers and supports the connection type required for your region and call direction.
- Match the provider route to the model documentation. For OpenAI, review the SIP guide or implement the server-side bridge. For Gemini Live, review the WebSocket API documentation and the specific partner integration’s setup requirements.
- Check inbound and outbound support separately. OpenAI’s guide covers both, subject to provider and organization configuration; Google’s overview describes the Voximplant integration for both directions. Provider and account prerequisites still apply.
- Keep the responsibilities explicit. Confirm who owns call signaling, audio transport, session control, webhooks and call decisions before implementation.
Can I connect a voice AI model to a phone number?
Yes, where the model API’s documented route is paired with a telephony provider or integration. OpenAI explicitly describes a SIP-trunk route and an audio bridge. Google documents WebSocket-based Live API access and lists a partner call integration. Neither description means the model provider is automatically your phone carrier.
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- SMART CALL BLOCKER & CALLER ID ANNOUNCE: Say goodbye to unwanted calls. Robocalls on your landline are automatically blocked from ever ringing through - even the first time. You can also permanently blacklist any number you want with one touch on the dedicated key on the handset. The call block directory can store up to 1,000 name and number entries. Plus, the handset announces the name of the caller, so you can decide on answer the call or block it - screening call is never easier
- LARGE 2-INCH SCREEN, BIG TEXT, LIGHTED KEY PAD: High-contrast text on the extra-large 2 inch screen makes it easy to read incoming caller ID or call history records. Plus, the enlarged font and extra-large and lighted handset keypad allows for easy dialing in low-light conditions. This feature is especially helpful for those who are visually impaired
- HANDSET SPEAKERPHONE, AUDIO ASSIST, INTERCOM: This cordless system has built-in a full-duplex speakerphone on handset allowing both ends to speak - and be heard - at the same time for conversations that are more true to life. Also designed with useful features like Audio Assist, handset intercom to help your daily communications enjoyable
The original three-model comparison remains incomplete because the third model is not identified. Without that identity and its current official voice/telephony documentation, the claim that only one of the three documents a native path cannot be confirmed.
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