Session Initiation Protocol (SIP) is a cornerstone of modern voice communication, enabling calls over the internet (VoIP). By integrating SIP with MediaSFU, you bridge traditional telephony (PSTN) with our powerful real-time media processing capabilities. This allows your applications to make and receive phone calls, interact with callers using automated systems, and seamlessly blend voice interactions with other digital communication channels.

Key Benefits:

• PSTN Connectivity: Connect your application to the Public Switched Telephone Network, allowing users to dial in from and dial out to standard phone numbers worldwide.
• Automated Voice Agents (AI & IVR): Leverage MediaSFU's AI pipeline (Speech-to-Text, Large Language Models, Text-to-Speech) to build intelligent voice bots. Implement Interactive Voice Response (IVR) systems for self-service or call routing.
• Dynamic Audio Playback: Play pre-recorded audio files or dynamically generate Text-to-Speech messages to callers, using templates for personalized communication.
• Advanced Call Control: Manage call recording, implement automated callback queues, and integrate with your backend systems via webhooks for real-time call events and control.
• Unified Communications: Blend SIP calls with WebRTC sessions for comprehensive communication solutions, bridging traditional telephony with modern web-based interactions.

This guide will walk you through configuring SIP trunks by setting up your SIP provider details within MediaSFU, managing contact numbers (DIDs), and integrating advanced features like AI agents and custom audio prompts.