AI-103: Developing AI Apps and Agents on Azure

By Microsoft · Question Mode

QuestionQ89

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You are developing a customer support web application called App1 in Microsoft Foundry that uses a GPT realtime model.

App1 must support:

  • Live, low-latency voice conversations powered by Azure OpenAI
  • Streaming audio input from users, along with playback of audio responses

You need to select a connection method that supports real-time audio streaming within the client application and achieves approximately 100 ms of latency.

Which connection method should you choose?

Explanation

WebRTC is purpose-built for peer-to-peer, real-time audio (and video) communication and is the connection method Microsoft recommends for client-side applications—such as browser-based or mobile apps—that use the Azure OpenAI GPT Realtime API to stream microphone audio to the model and play back generated speech with minimal delay, typically achieving round-trip latencies around 100 ms. WebSocket, while supported by the Realtime API, is intended for server-to-server integrations where strict low latency is not required, since it introduces higher end-to-end delay than WebRTC. SIP is intended for telephony/PSTN call integration rather than in-browser or in-app client streaming, and RTMP is not a supported protocol for the Realtime API at all.

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