Key takeaways
- Most SIP trunk providers are built for human call centres — voice AI has fundamentally different infrastructure requirements
- Bidirectional WebSocket streaming is the core requirement: the SIP trunk must stream live audio to your AI pipeline and accept synthesised TTS back in real time
- Media-layer processing (noise cancellation, voice isolation, PLC, FEC) at the SIP trunk level means your ASR receives clean audio — without any pipeline-side denoising
- For India: Mumbai PoP, 140-series numbers, and direct Jio/Airtel/VI interconnects are non-negotiable for sub-second response times and TRAI compliance
- Indian voice AI platforms — Sarvam AI, Bolna, Ringg, Rapida AI — connect via the same SIP + WebSocket infrastructure as global platforms
What makes a SIP trunk purpose-built for voice AI
Traditional SIP trunks deliver audio as a continuous RTP stream between two parties and handle billing. That’s enough for a human agent. AI voice agents have a different set of requirements that most SIP providers have never had to solve: Real-time bidirectional audio. The AI pipeline needs to receive the caller’s audio in 20ms frames and inject synthesised TTS speech back into the call — not at the end, but continuously throughout the conversation. Standard SIP trunks weren’t designed for this. It requires a bidirectional WebSocket channel that runs for the duration of the call. Media processing before the ASR. When a caller is in a noisy environment — a busy street, a factory floor, a vehicle — the audio reaching your ASR carries noise energy that degrades transcription accuracy. If the SIP trunk handles noise cancellation, voice isolation, and packet loss concealment at the media layer, your ASR receives clean audio regardless of environment. If it doesn’t, you’re either denoising in the pipeline (adds latency) or accepting worse transcription. Answering machine detection at the infrastructure level. When an AI agent calls a number that goes to voicemail, it shouldn’t start speaking its opening line. AMD at the SIP trunk level fires before the AI session starts — no LLM tokens consumed, no confused conversation where the agent addresses a voicemail recording as if it were a person. AMD needs to be a parameter on the call, not an afterthought. Latency architecture matched to AI inference. End-to-end turn latency for a natural-sounding conversation needs to stay under 1,000ms. The SIP trunk’s network contribution should be under 50ms one-way. For India-to-India calls, that means a Mumbai PoP — routing via Singapore adds 80–150ms, which is 8–15% of your total latency budget before ASR even starts. India regulatory infrastructure. 140-series numbers are mandatory for B2C outbound in India. TRAI’s Third Amendment (September 2026) adds declaration requirements for AI-initiated calls. Direct Jio, Airtel, and VI interconnects (not routed through international carriers) determine whether India-to-India audio quality is consistent.Provider comparison
Vobiz — built for voice AI
Vobiz is the only SIP trunk provider in India whose infrastructure was designed from the ground up for voice AI workloads. Every call that passes through the Vobiz media layer is processed through a full AI-optimised stack before audio reaches your ASR:- HD Voice (G.722, Opus up to 20 kHz) — wideband audio that ASR models are trained on, not the 8 kHz narrowband that PSTN was designed for
- Noise cancellation — stationary (HVAC, fan) and non-stationary (traffic, crowd) noise suppressed at the media server
- Voice isolation — primary speaker extracted from multi-speaker environments; crosstalk and background speech removed before ASR input
- Packet Loss Concealment (PLC) — lost RTP packets synthesised in real time; ASR receives a continuous stream even on degraded mobile networks
- Forward Error Correction (FEC) — Opus FEC embeds redundant data in adjacent packets; burst losses reconstructed without synthesis
- Voice Activity Detection (VAD) — silence suppressed, end-of-utterance signalled to the AI platform, interruptions detected
Exotel
India’s most established cloud telephony provider. Strong TRAI compliance history, INR billing, and direct carrier relationships with Jio, Airtel, and VI. Media layer is standard telephony infrastructure — not optimised for AI workloads. Limited bidirectional WebSocket streaming support means it’s better suited for traditional IVR with AI augmentation than fully AI-native agents. No native integrations with Retell AI, Vapi, or ElevenLabs. Best for: Existing Exotel customers adding AI features to traditional IVR; India-only deployments where TRAI compliance and established carrier relationships matter more than AI-native infrastructure.Plivo
Competitive global provider with India coverage routed via Singapore. Bidirectional streaming available; no India PoP means 80–150ms additional latency for India calls. Limited 140-series provisioning capability. Works with the main global AI platforms but adds avoidable latency for India-primary deployments. Best for: Global deployments where India is one of several markets; teams already using Plivo for SMS or international calling.Telnyx and Twilio
Both offer strong global coverage, well-documented AI platform integrations, and reliable infrastructure. Neither has an India media PoP — traffic routes internationally, adding significant latency for India-to-India calls. Neither supports 140-series number provisioning for TRAI-compliant outbound. Both are well-suited for US/EU-primary voice AI deployments. Best for: Teams serving US/EU markets; not recommended as a primary SIP trunk for India voice AI deployments.Indian voice AI platforms — SIP trunk requirements
The Indian voice AI ecosystem has developed its own set of platforms optimised for Indian languages, accents, and regulatory context. Each connects to the telephony layer via SIP or WebSocket.Sarvam AI
India’s leading multilingual ASR + TTS provider — covering Hindi, Tamil, Telugu, Kannada, Bengali, Marathi, and more. Sarvam AI connects via WebSocket audio streaming: the telephony layer sends audio frames, Sarvam transcribes and synthesises, the response flows back. Vobiz’s bidirectional WebSocket streaming is fully compatible. The Mumbai PoP matters significantly for Sarvam pipelines — sub-50ms network latency keeps the full Sarvam ASR + custom LLM + Sarvam TTS loop under 1 second. Sarvam AI + Vobiz integration →Bolna
Open-source voice AI platform built for Indian deployments. Bolna orchestrates the ASR + LLM + TTS pipeline and connects to the PSTN via SIP trunking. Vobiz provides the SIP trunk and phone numbers. Bolna’s architecture supports Indian language models natively and handles conversation state across multi-turn interactions.Ringg
AI calling platform for Indian businesses — outbound campaigns, customer service, appointment reminders. Ringg’s AI agents run on top of PSTN telephony. SIP trunking provides the carrier connectivity and phone number provisioning. For 140-series outbound campaigns on Ringg, the SIP trunk provider must support TRAI-compliant caller ID.Rapida AI
AI voice platform with India focus. Connects to telephony via SIP trunking. Vobiz provides a native integration for Rapida AI agents to place and receive calls on Indian numbers. Rapida AI + Vobiz integration →Dograh
Open-source voice agent platform that supports multiple SIP trunk providers. Dograh documentation explicitly lists Vobiz as the recommended option for India-based deployments: “If callers are in India, look at Exotel, Vobiz or a carrier through the India region of Managed SIP.” Dograh + Vobiz integration →Setup by platform
Global platforms:- Retell AI + Vobiz — no-code dashboard setup, 5 minutes to first call
- Vapi + Vobiz — import SIP credentials, inbound and outbound
- ElevenLabs + Vobiz — ultra-realistic voices on a real phone line
- LiveKit + Vobiz — real-time media pipeline via SIP
- Pipecat + Vobiz — custom ASR + LLM + TTS pipelines
- OpenAI Realtime + Vobiz — GPT-4o voice on PSTN
- Sarvam AI + Vobiz — 10 Indian languages, full ASR + TTS
- Rapida AI + Vobiz — India-focused AI calling
- Dograh + Vobiz — open-source voice agents on Indian numbers
What to verify before choosing a SIP trunk for voice AI
- Media server is in Mumbai or Chennai — not Singapore or US
- Bidirectional WebSocket audio streaming is natively supported
- Media layer includes noise cancellation and voice isolation (not just standard RTP delivery)
- AMD fires before AI session starts, not post-call
- 140-series caller IDs are available for TRAI-compliant outbound
- Direct Jio, Airtel, and VI interconnects (verify with a test call, not a spec sheet)
- Wideband audio (16 kHz L16 or G.722) supported on WebSocket stream
- API-first: programmatic call creation, number provisioning, trunk management
Related
- SIP trunking explained — how SIP trunks work, sizing, authentication
- SIP vs WebSockets for AI agents — when to use each
- Audio quality infrastructure — Vobiz media layer: HD Voice, VAD, noise cancellation, voice isolation, PLC, FEC
- AI voice agent architecture — full stack guide
- India calling compliance — 140-series, NDNC, TRAI
- Sarvam AI integration — 10 Indian language ASR + TTS on Vobiz