To mitigate the latency spikes and high costs associated with using CPaaS brokers like Twilio, this article details an architecture for building a direct SIP AI voice agent by terminating Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP) sessions natively within an application tier. By bypassing third-party gateways, developers can achieve sub-second latency and direct control over raw audio buffers. The proposed implementation uses Python's asynchronous networking capabilities to handle concurrent socket connections. The technical workflow involves a `DirectSIPServer` that intercepts SIP INVITE messages and responds with custom Session Description Protocol (SDP) parameters to dictate RTP streaming locations. An `RTPReceiver` class uses a non-blocking UDP socket to strip the standard 12-byte RTP header and convert the payload into 16-bit PCM audio arrays via NumPy for immediate inference. The processed audio is then passed to an asynchronous pipeline, and the resulting response is re-encapsulated into RTP packets using a custom `RTPSender` that manually constructs headers, including sequence numbers and timestamps, for transmission back to the carrier. To ensure production stability, the author emphasizes the necessity of managing NAT traversal, preventing event loop blocking during LLM inference, and implementing jitter buffers to prevent audio clipping.