Muhammad Bilal T.
VoIP | AI Voicebots | Freeswitch | Asterisk |SIP|WebRTC| ASR/TTS/LLM
I Build robust VoIP systems and intelligent AI voicebots. I work with Asterisk, FreeSWITCH, SIP, WebRTC, ASR, TTS, and LLM-driven conversational logic. My goal is to deliver scalable voice platforms that are secure, cost-efficient, and designed to solve real business problems. What I Can Do for You - Design and deploy VoIP infrastructure (Asterisk / FreeSWITCH) - SIP trunk setup, routing, and troubleshooting - WebRTC browser-based calling integration - IVR / automated call flows and menus - AI voicebots with ASR, TTS, and LLM-driven conversational logic - Speech-to-text (Whisper or other open-source / cloud models) - Intelligent call routing, intent detection, and conversation management - Contact center solutions: queues, failover, call logging, and reporting Why Work With Me - I combine deep VoIP expertise with AI voicebot development - I focus on reliability, scalability, and performance - I deliver solutions that save time, reduce costs, and automate repetitive tasks - I communicate clearly and provide regular updates - I document solutions for easy maintenance and hand-off Skills VoIP, Asterisk, FreeSWITCH, SIP, WebRTC, ASR, TTS, Whisper, LLM Integration, IVR / Voice Flow Design, AI Voicebots, Contact Center Architecture, Linux Administration, SIP Trunking, Debugging & Monitoring. Sample Services / Projects: AI Voicebot for Customer Service -Build a voicebot that understands customer requests via ASR, interprets intent with LLM logic, and responds using TTS. Can route callers to human agents or automate business workflows. VoIP Server Deployment -Deploy Asterisk or FreeSWITCH, configure SIP trunks, extensions, failover, and test call quality. Deliver a stable, documented, and scalable setup. IVR & Call Flow Automation -Create IVR menus, routing logic, fallback paths, and integrate with AI voicebots or contact center systems. WebRTC Softphone / Browser Calling -ntegrate WebRTC calling into your application or website with signaling, TURN/STUN setup, and media configuration. VoIP Monitoring & Optimization -Implement call analytics, monitor RTP quality, and optimize call flow for reliability, scalability, and cost efficiency.
Earnings and projects
Last 6 months ÷ 6
Across 1 contracts
Across 7 contracts