How a call flows
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Wetel’s telephony capability lets an AI agent talk to people over a real phone call. This page explains the architecture for integrators.
Current status: Telephony is a test-environment capability available on request. It is not in production.
Status at a glance
Section titled “Status at a glance”| Capability | Status |
|---|---|
| Calls to a demo phone number (test environment) | ✅ Wetel-run demo, arranged on request |
| Outbound calls (Wetel-run internal tests) | ✅ Internal test only |
| Customer outbound calling | 🟡 Planned — not available |
| Production calls | ❌ Not available |
| Live transfer to a person | ❌ Not available (verbal offer only) |
| Call recording | ❌ Off by default |
Diagrams
Section titled “Diagrams”Inbound call
Section titled “Inbound call”sequenceDiagram participant Caller participant PSTN as Phone Network participant SIP as SIP Service (Cloud) participant Dispatch as Dispatch Rule participant Room as Voice Room participant Agent as Voice Agent (Worker) participant STT as Speech-to-Text participant Brain as Wetel Agent Workflow participant TTS as Text-to-Speech
Caller->>PSTN: Dials the test phone number PSTN->>SIP: SIP INVITE SIP->>Dispatch: Route Dispatch->>Room: Create room, dispatch agent Room->>Agent: Join Agent->>TTS: Greeting (AI disclosure, offer a person) TTS-->>Room: Audio Room-->>SIP: Audio SIP-->>PSTN: Audio PSTN-->>Caller: Hears greeting
loop Each conversational turn (barge-in supported) Caller->>PSTN: Speech PSTN->>SIP: Audio SIP->>Room: Audio Room->>Agent: Audio Agent->>STT: Utterance STT-->>Agent: Text Agent->>Brain: Message Brain-->>Agent: Reply Agent->>TTS: Text TTS-->>Agent: Audio Agent->>Room: Audio Room->>SIP: Audio SIP->>PSTN: Audio PSTN->>Caller: Hears reply end
Caller->>PSTN: Hang up PSTN->>SIP: BYE SIP->>Room: End sessionOutbound test lane (Wetel-run test)
Section titled “Outbound test lane (Wetel-run test)”This shows how Wetel engineers verify the system. Customer outbound calling is planned, not yet available.
sequenceDiagram participant Eng as Tester participant Script as Test Script participant SIPProvider as A SIP Provider participant Phone as Tester's Phone participant SIPService as SIP Service (Cloud) participant Trunk as Inbound Trunk participant Dispatch as Dispatch Rule participant Room as Voice Room participant Agent as Voice Agent (Worker)
Eng->>Script: Start (with consent, daytime only) Script->>SIPProvider: Place call to tester's own verified number SIPProvider->>Phone: Ring Phone->>SIPProvider: Answer SIPProvider->>Phone: Provider notice (trial) Phone->>SIPProvider: Acknowledge SIPProvider->>SIPService: SIP INVITE (via trunk credentials) SIPService->>Trunk: Match Trunk->>Dispatch: Route Dispatch->>Room: Create room, dispatch agent Room->>Agent: Join Agent->>Room: AI disclosure + conversation
Note over Phone,Agent: Same per-turn loop as inbound.<br/>Hard stop 300 seconds.System components
Section titled “System components”flowchart TB subgraph Phone["Phone Network"] CallerPhone["Caller's phone"] end
subgraph Cloud["Cloud SIP + Rooms"] SIPService["SIP Service"] DispatchRule["Dispatch Rule<br/>(one room per caller)"] VoiceRoom["Voice Room"] end
subgraph Worker["Voice Agent Worker"] AgentProcess["Agent process<br/>(one call at a time)"] end
subgraph Services["AI Services"] STT["Speech-to-Text<br/>(multi-language)"] TTS["Text-to-Speech"] Brain["Wetel Agent Workflow<br/>(staging)"] end
subgraph SIPProv["A SIP Provider<br/>(outbound test only)"] OutboundAPI["Outbound call API"] end
CallerPhone <-->|"PSTN"| SIPService SIPService --> DispatchRule --> VoiceRoom VoiceRoom <--> AgentProcess AgentProcess --> STT STT --> AgentProcess AgentProcess --> Brain Brain --> AgentProcess AgentProcess --> TTS TTS --> AgentProcess
OutboundAPI -->|"SIP INVITE"| SIPServiceWhat each part does
Section titled “What each part does”| Component | Role |
|---|---|
| SIP Service | Bridges the phone network to the cloud voice room. |
| Dispatch Rule | Routes an incoming SIP call to a new voice room and assigns an agent. |
| Voice Room | Real-time audio session between the caller and the agent. |
| Voice Agent Worker | Orchestrates each turn: listens, transcribes, thinks, speaks. Currently handles one call at a time. |
| Speech-to-Text | Converts each utterance to text (supports English and Malay). |
| Wetel Agent Workflow | The “brain” — runs the agent’s logic and returns a reply. |
| Text-to-Speech | Converts the agent’s reply to audio. |
| A SIP provider | Used only in the outbound test lane to originate a call to a verified tester phone. |
What you can try today
Section titled “What you can try today”- A demo call. On request, Wetel can arrange a test call to a demo phone number. Dial it and speak to the AI agent. The agent discloses it is AI and offers to connect a person (verbal offer only — live transfer is not yet built). Barge-in (interrupting the agent mid-sentence) is supported.
- Observe a Wetel-run outbound test. A Wetel engineer can demonstrate the outbound path by calling their own verified number with their consent, during daytime hours only.
Every call creates a session with a transcript, similar to chat.
What is not available
Section titled “What is not available”| Feature | Notes |
|---|---|
| Production calls | The agent workflow runs on the staging environment only. |
| Customer outbound calling | Designed; not built. Requires a SIP provider with local numbers, legal review of consent and recording requirements, and opt-in controls (off by default). |
| Live transfer to a human | The agent verbally offers it; the transfer itself is not implemented. |
| Local (Malaysian) phone numbers | Requires a SIP provider with Malaysian numbers; not available yet. |
| WhatsApp / Meta voice calling | Not part of this system. |
| Call recording | Off by default. |
Latency
Section titled “Latency”| Metric | Current | Target |
|---|---|---|
| Per-turn reply time | ~3.5–5 seconds (workflow or prompt agent); ~1.4–1.8 seconds (speech-to-speech option, no workflow) | 1.5 seconds |
| Speech-to-text per utterance | 0.65–1.8 seconds | — |
Phone audio is 8 kHz, which can reduce transcription accuracy for some accents. Latency improvements (streaming transcription, shorter replies, faster models, closer hosting) are on the roadmap.
Safety and consent
Section titled “Safety and consent”- The agent discloses it is AI at the start of every call.
- Outbound calls require explicit consent from the recipient.
- No recording by default.
- Per-key daily call caps are enforced in the staging environment.
- Customer-facing outbound calling, when built, will be off by default and require owner approval.