SIP Telephony & Voice AI Integration

Implement integration of SIP telephony with voice AI agent for handling incoming/outgoing calls through Asterisk or alternative

Initial Data
Telecommunication operator: TBD

Telephony server: Asterisk (Linux VPS/server, dedicated IP)

Task: automatic call handling through AI agent in real time.

Main Tasks
1. Connecting SIP Trunk
Configure Asterisk or similar

Check routing of outgoing calls.

Self-assembly (STT + LLM + TTS)

Use a fast engine for STT (for example, Deepgram, Google Speech-to-Text, AssemblyAI).

Dialogue logic on GPT (OpenAI)

Speech synthesis via Deepgram TTS, Google TTS, Microsoft Azure TTS.

Ensure audio streaming to minimize latency (practically in real time).

3. Functional Requirements
Automatic response to incoming calls by the voice agent.

Scenario behavior capability (for example, transferring the call to a live operator).

Logging: text transcription summary

Scalability: at least 10 simultaneous calls, with the possibility of expansion.

call transferring: transfer calls to human agent when complete.

4. Quality and Reliability Requirements
Minimal delays in dialogue (natural conversation).

High quality of speech recognition and synthesis.

Stability of operation even under peak load.