SIP Trunking Architecture for Conversational Voice AI
A comprehensive engineering guide for connecting corporate PBX telephony, carrier SIP trunks, and WhatsApp voice calling directly to low-latency AI inference clusters.
Core Components of the Voice AI Telephony Stack
How Gyan VaniAi securely interfaces legacy telecom carriers with next-generation streaming AI models.
Carrier-Grade Session Border Controller (SBC)
Provides encrypted TLS signaling termination, topology hiding, rate limiting, and real-time packet inspection to protect internal voice inference clusters from DDoS attacks.
Full-Duplex SRTP to WebSocket Media Bridge
Translates inbound RTP UDP packets into binary PCM WebSocket streams in sub-10ms, forwarding audio frames directly to our streaming STT and VAD engines.
Dynamic Call Transfer & Human Agent Handoff
Executes SIP REFER or attended transfers to human rep extension lines when callers request complex assistance or escalate beyond AI authorization limits.
Regulatory Compliance & Legal Partitioning
Complies with Indian TRAI regulations governing VoIP and PSTN interconnection, ensuring proper caller ID (CLI) presentation and secure call detail recording (CDR).
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Frequently Asked Questions on SIP Architecture
Everything you need to know about our AI CRMs, integrations, and deployment timelines.
Yes. You can route specific extension ranges, toll-free DIDs, or overflow queues to Gyan VaniAi SIP endpoints while keeping your primary telecom provider and existing PBX infrastructure intact.
Our Kubernetes-orchestrated media bridge auto-scales across multi-zone cloud regions, supporting from 10 to 5,000+ simultaneous concurrent voice channels.
The SBC and media conversion bridge adds less than 12 milliseconds of overhead to the overall audio pipeline, preserving our sub-300ms total conversational turnaround.
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