The SIP-AI Gateway is a flexible, high-performance solution that bridges traditional SIP systems with modern artificial intelligence capabilities. 
Whether you need real-time speech recognition, conversational AI agents, sentiment analysis, real-time translation, automated transcription, or custom AI-driven voice interactions, our gateway can make it possible with no changes required in your existing infrastructure. 

We have already successfully integrated advanced AI features into our JVoIP library and are extending these capabilities to our core VoIP Server and SIP SBC. The SIP-AI Gateway builds on this proven foundation.  
With the SIP-AI gateway you can bridge legacy SIP infrastructures to modern AI services unlocking the power of AI in your SIP infrastructure, making it a perfect solution for companies looking to modernize legacy PBX systems with AI. 
Add intelligence to inbound/outbound calls - detect intent, respond naturally, transcribe conversations, analyze sentiment, and more.

• VoIP Protocols: SIP and WebRTC (trunks and endpoints/extensions)
• Transport protocols: UDP, TCP, TLS (SIPS), WS/WSS (Secure Websocket), RTP/SRTP, DTLS
• Codecs: raw PCM, G.711, G.729, GSM, OPUS, Speex and more. Automatic transcoding when necessary 
• Integration possible with most AI platforms via websocket (OpenAI Realtime API, Gemini Live, Deepgram Voice Agent, Google Cloud, Azure GPT Realtime, Anthropic, local LLMs, etc.)

With the SIP-AI gateway, we can implement any of the followings:

• Real-time speech-to-speech integration (STS)
• Live transcription and summarization of calls
• Sentiment analysis and call quality monitoring
• Custom business logic triggered by AI insights
• Custom call flow logic (DTMF, announcements, call transfer)
• Custom AI endpoint extensions can be implemented with the JVoIP library.
• Intelligent IVR with natural language understanding
• Conversational voice bots and virtual agents
• Secure, low-latency media and signaling processing
• Scalable from small deployments to carrier-grade environments
• And many more. Contact us with your idea or requirements.

• Seamless SIP integration: Acts as a powerful transparent intermediary between your PBX / SIP server (Asterisk, Avaya, Cisco, FreeSWITCH, etc) and SIP/WebRTC endpoints / trunks. No changes are required in your existing setup and configurations. 
• On-premise, private, secure: The gateway runs on your infrastructure, the audio data stays on your network, you control what is sent to external AI services.
• No lock-in: We provide a perpetual license and you choose your AI provider.
• Custom development: We will adapt to your existing infrastructure (any SIP/PBX, API, backend/webservice).
• Highly customizable: We tailor the solution to your exact needs.

Send an email to info@mizu-voip.com with your idea or requirements. Our team will find out the best possible solution and will deliver a tailored SIP-AI gateway to match your exact needs. 

If possible, please let us known the followings:

• What are your main requirements, use-case, business-logic?
• What PBX / SIP server(s) are you using?
• Which AI service do you wish to use (if you have any preference)?
• What should the AI do (answer, translate, transfer, analyze)? 
• What is your call volume (maximum number of concurrent calls)?

We will respond asap with a proposal, timeline and cost estimate. Once built, we can also deploy it for you if you wish or will send easy to follow step-by-step instructions.