WhatsApp Voice AI services with Kamailio and Asterisk

Telecom platforms are increasingly expected to support real-time AI interactions, yet most implementations rely on CPaaS abstractions that hide the underlying call mechanics.

This session presents a practical implementation of a WhatsApp voice integration built on SIP, using Kamailio as the core.

We start with the signaling and media layer:

• Handling WhatsApp voice calls via Meta’s gateways
• Managing RTP streams and media flow
• Implementing routing logic, authentication, and CDR generation in Kamailio

On top of this, we introduce an open source AI voice service integrated as a SIP endpoint:

• Real-time RTP stream capture
• Streaming audio to STT services
• Processing with an LLM
• Returning synthesized speech (TTS) into the live call

We will discuss different service examples and also present learnings from real-world usage of the service.