Technical Reference

Voice AI & Telephony Glossary.

An authoritative encyclopedia of technical specifications, protocols, and definitions for conversational voice AI, WebRTC, SIP trunking, and neural telephony.

Technical specifications
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01Protocols02Codecs03Standards
Showing 16 verified definitions
Telephony & SIP

SIP Trunking (Session Initiation Protocol)

A digital telephony protocol connecting an enterprise PBX or voice AI system directly to the public switched telephone network (PSTN) over the internet.

SIP Trunking eliminates the need for traditional physical analog telephone lines by streaming voice calls as digital data packets using VoIP protocols. NestVoice AI integrates natively with carrier SIP trunks from Twilio, Telnyx, Plivo, FreeSWITCH, and Asterisk via standard SIP INVITE and BYE methods.

Key Specifications
RFC 3261 CompliantDirect SIP URI RoutingG.711 & Opus Codec Negotiation
Related:Session Border Controller (SBC),Warm Agent Handoff (SIP REFER)
Audio & Streaming

WebRTC Full-Duplex Audio

A real-time open-source communications framework allowing continuous, bidirectional audio streaming between web browsers, edge servers, and AI voice engines.

Unlike sequential HTTP-based audio requests, WebRTC provides true full-duplex communication over UDP with low packet overhead. NestVoice AI utilizes LiveKit SFU infrastructure to stream Opus 48kHz high-definition audio with dynamically adapted jitter buffers, keeping total round-trip audio delivery under 100 milliseconds.

Key Specifications
Opus 48kHz Wideband CodecSub-100ms Media TransportSRTP Media Encryption
Related:Voice Activity Detection (VAD),Time-to-First-Audio (TTFA)
Audio & Streaming

Voice Activity Detection (VAD) & Barge-in

Edge algorithms that determine the exact millisecond human speech begins or ends, enabling conversational AI agents to pause speaking when interrupted.

Barge-in allows callers to interrupt an AI agent naturally mid-sentence. When the caller speaks, the edge VAD (using deep neural networks such as Silero VAD) detects speech within 20 milliseconds, immediately stops audio synthesis playback, and resets the LLM generation state to listen to the new utterance.

Key Specifications
< 20ms Frame Processing< 80ms Total Barge-In ResponseAcoustic Echo Cancellation (AEC)
Related:WebRTC Full-Duplex Audio,Conversational Turn-Taking Latency
AI & Models

Automatic Speech Recognition (ASR)

The speech-to-text process that transcribes incoming caller acoustic audio streams into clean text tokens in real time.

Streaming ASR models emit intermediate word hypotheses every 30 to 50 milliseconds before a caller finishes their sentence. This speculative pre-transcription allows the downstream LLM orchestrator to begin reasoning early, drastically cutting turn-around time. NestVoice AI supports Deepgram Nova-2, Whisper Large-v3 Turbo, and Sarvam AI.

Key Specifications
Word-level TimestampsConfidence ScoringCustom Acoustic Vocabulary Boosting
Related:Text-to-Speech (TTS),Conversational Turn-Taking Latency
AI & Models

Text-to-Speech (TTS) Neural Synthesis

Neural models that generate human-sounding spoken voice audio from textual LLM responses with natural inflection, breathing, and prosody.

State-of-the-art streaming TTS models (such as Cartesia Sonic, ElevenLabs Flash, and Sarvam Bulbul) synthesize audio chunks progressively as text tokens stream out of the LLM. Rather than waiting for a full sentence to generate, audio starts streaming to the caller in under 120 milliseconds.

Key Specifications
Time-to-First-Audio: 90–140msContextual Prosody AdaptationMulti-voice Cloning Support
Related:Automatic Speech Recognition (ASR),Time-to-First-Audio (TTFA)
Audio & Streaming

Conversational Turn-Taking Latency

The total elapsed time from when a human speaker stops talking until the AI voice agent begins audible response playback.

Also known as glass-to-glass latency, it includes speech endpointing, ASR finalization, LLM first-token generation, TTS first-chunk synthesis, and audio network transit. NestVoice AI delivers conversational turn-taking latency below 600 milliseconds (p95 at 580ms), matching the cadence of natural human dialogue.

Key Specifications
p95 Target: < 600msHuman Average Cadence: ~300–700msSpeculative Pipeline Overlap
Related:Voice Activity Detection (VAD),Time-to-First-Audio (TTFA)
AI & Models

Code-Switching (Multilingual NLP)

The linguistic ability of a voice AI model to switch between multiple languages or dialects within a single continuous conversation.

In global enterprise deployments, callers frequently alternate between languages (such as English and Hindi in 'Hinglish', or Spanish and English in 'Spanglish'). NestVoice AI's multi-lingual neural pipeline recognizes vocabulary shifts in real time without requiring the caller to restart or select a language option.

Key Specifications
60+ Supported LanguagesDynamic Mid-Call Language AdaptationNative Regional Accents
Related:Automatic Speech Recognition (ASR),Text-to-Speech (TTS)
Telephony & SIP

Warm Agent Handoff (SIP REFER RFC 3515)

A telephony transfer protocol that bridges an ongoing AI call to a live human representative while transferring real-time transcripts and caller history.

When a caller requests a person or an escalation threshold is reached, NestVoice AI sends a SIP REFER request to the carrier SBC. The live representative receives the incoming call along with a real-time summary of the conversation, caller identity, and collected CRM data so the customer never has to repeat themselves.

Key Specifications
RFC 3515 SIP REFER ProtocolSIP User-to-User Info (UUI) HeadersReal-Time Webhook Synopsis
Related:SIP Trunking,Session Border Controller (SBC)
Telephony & SIP

STIR/SHAKEN Caller ID Attestation

A suite of cryptographic telecom protocols designed to authenticate caller ID numbers and prevent outbound calls from being labeled as spam.

STIR (Secure Telephone Identity Revisited) and SHAKEN (Signature-based Handling of Asserted information using toKENs) use digital certificates to verify that the calling party has legitimate ownership of the displayed telephone number. NestVoice AI supports full A-Level Attestation with verified carrier trunks.

Key Specifications
A-Attestation (Full Ownership Verified)FCC & Telecom Regulatory ComplianceSpam Label Prevention
Related:SIP Trunking,Session Border Controller (SBC)
Security & Compliance

Bring Your Own Key (BYOK)

An enterprise procurement architecture allowing organizations to connect their direct API credentials for AI and telephony providers with zero markup.

Under BYOK, customers connect their existing OpenAI, Anthropic, Cartesia, Deepgram, Twilio, or Telnyx accounts directly into NestVoice AI. The platform charges a flat orchestration fee rather than marking up third-party tokens or carrier minutes by 200–400%.

Key Specifications
$0.00 Token MarkupEncrypted Vault Credential StorageWholesale Provider Billing
Related:Zero Data Retention (ZDR),SIP Trunking
Security & Compliance

Zero Data Retention (ZDR)

A strict privacy configuration where call audio recordings and transcripts are processed exclusively in volatile RAM and immediately wiped.

For healthcare providers, financial institutions, and GDPR-compliant organizations, ZDR guarantees that customer audio streams and text transcriptions are never written to persistent disk storage. Once the call terminates and webhook payloads are dispatched, all memory buffers are cryptographically zeroed.

Key Specifications
In-Memory Ephemeral ProcessingHIPAA PHI ProtectionGDPR Article 17 Compliant
Related:Bring Your Own Key (BYOK),WebRTC Full-Duplex Audio
Telephony & SIP

Session Border Controller (SBC)

A specialized network security appliance governing how phone calls are initiated, routed, and secured across IP telephony boundaries.

An SBC sits between an enterprise internal private PBX and the external telecom provider or cloud AI gateway. It manages NAT traversal, enforces TLS and SRTP media encryption, shields internal SIP infrastructure from DDoS attacks, and normalizes SIP signaling dialects.

Key Specifications
SIP Signaling FirewallNAT Traversal & Topology HidingTLS & SRTP Media Transcoding
Related:SIP Trunking,Warm Agent Handoff (SIP REFER)
Security & Compliance

Dual-Tone Multi-Frequency (DTMF) Masking

The secure handling and audio suppression of keypad tones to protect credit card numbers, PINs, and sensitive financial inputs during calls.

When a caller enters sensitive payment card data using their phone keypad, DTMF tones (RFC 4733 / RFC 2833) are transmitted out-of-band via SIP signaling rather than audible audio. NestVoice AI masks these tones from recordings and logs, ensuring full PCI-DSS compliance.

Key Specifications
RFC 4733 / RFC 2833 CompliantPCI-DSS Level 1 ComplianceAudio Tone Muting
Related:Zero Data Retention (ZDR),SIP Trunking
Audio & Streaming

Time-to-First-Audio (TTFA)

The duration between the moment an LLM starts producing response tokens and the first audible sound wave rendered to the listener.

TTFA is a critical benchmark for conversational naturalness. By streaming partial sentence chunks directly to low-latency neural TTS models without waiting for punctuation, NestVoice AI minimizes TTFA to 90–140 milliseconds.

Key Specifications
Target: 90–140msChunked Audio StreamingToken Speculation
Related:Text-to-Speech (TTS),Conversational Turn-Taking Latency
Telephony & SIP

Answering Machine Detection (AMD)

Real-time acoustic analysis determining whether an outbound call was answered by a live human or an automated voicemail inbox.

In outbound calling campaigns, reaching voicemail requires a different behavioral flow than speaking with a live human. NestVoice AI analyzes speech rhythm, greeting duration, and audio frequency within the first 1.5 seconds to distinguish voicemail greetings from live human answers with >98% accuracy.

Key Specifications
> 98% Detection Accuracy< 1.5 Second Decision WindowAutomated Voicemail Drop Option
Related:Voice Activity Detection (VAD),SIP Trunking
AI & Models

Voice RAG (Retrieval-Augmented Generation)

Real-time semantic retrieval of enterprise knowledge base articles during a live voice call to ground AI answers in verified facts.

During a phone conversation, the agent identifies caller questions, generates high-speed vector query embeddings, retrieves top-ranked enterprise documentation chunks within 25 milliseconds, and injects them into the LLM context window to provide accurate, hallucination-free answers.

Key Specifications
Sub-25ms Vector Database SearchHallucination MitigationDynamic Context Injection
Related:Automatic Speech Recognition (ASR),Code-Switching (Multilingual NLP)

Looking for Machine-Readable Technical Specifications?

All architectural parameters, audio pipeline budgets, and benchmark tables are accessible via standardized endpoints.

/llms.txt (Curated Overview)•/llms-full.txt (Full Technical Specs)