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INCOMING CALL
SIP Endpoint / PBX
Send a SIP call to the gateway.
Connect SIP calls to OpenAI, Gemini, ElevenLabs and Azure Voice Live.
Full-duplex audio, barge-in, SDES-SRTP and native AI configuration in a standalone Linux gateway. No FreeSWITCH required.
Send a SIP call to the gateway.
Handle call signaling, media and profile selection.
Stream audio in both directions for a live conversation.
SIP signaling, profile routing and realtime media — in one Linux gateway.
Accept incoming SIP calls from phones, PBXs or other SIP endpoints. Negotiate PCMU/PCMA audio and establish RTP or SDES-SRTP media sessions.
Route each call to a named AI profile using the SIP Request-URI. Connect to OpenAI, Gemini, ElevenLabs or Azure Voice Live with provider-native settings.
Stream audio bidirectionally with automatic codec conversion and resampling. Enable natural conversations with voice activity detection, response interruptions and barge-in.
Full-duplex SIP media, native AI provider options and automatic audio conversion — without FreeSWITCH.
Built-in SIP server for handling incoming calls.
G.711 μ-law (PCMU) and A-law (PCMA) media for compatible SIP calls.
OpenAI, Gemini, ElevenLabs and Azure Voice Live integrations.
Stream caller audio to AI and return generated speech to the caller.
Choose the configured AI profile from the SIP destination.
No FreeSWITCH installation or additional PBX is required.
Allow callers to interrupt AI speech for more natural conversations, using provider-supported interruption controls.
Convert and resample audio between 8 kHz SIP G.711 media and provider-specific input and output formats.
Use provider-native settings for voices, VAD, instructions, audio and supported tool or function declarations.
Assign different AI configurations to different SIP destinations.
The SIP Request-URI selects the matching named profile.
Use OpenAI, Gemini, ElevenLabs or Azure Voice Live without changing how callers reach the gateway.
Use provider-native JSON settings for models, voices, VAD, interruptions, audio formats and supported tool or function declarations — without a proprietary configuration format.
Create profiles for sales, support or other workflows and select them through the SIP Request-URI.
INVITE sip:sales@gateway.example SIP/2.0
Request-URI user: sales
Selected AI profile: sales
INVITE sip:support@gateway.example SIP/2.0
Request-URI user: support
Selected AI profile: supportRun sip_realtime_ai as a standalone application. Send compatible SIP calls directly to the gateway, or route them from your existing SIP infrastructure.
The gateway handles SIP signaling, RTP audio, provider sessions and bidirectional audio conversion in one service.
Built-in SIP server for incoming calls.
G.711 μ-law and A-law audio handling.
Encrypted RTP media with AES_CM_128_HMAC_SHA1_80 when negotiated by compatible SIP endpoints.
Run a standalone SIP-to-AI gateway with multiple realtime AI providers.
Explore SIP configuration, RTP and SRTP media, AI provider profiles and gateway deployment.
Read the Documentation