Let every call be handled by a professional AI voice agent
No system replacement, no architecture refactor, and no coding required. As long as network connectivity is available, you can add AI voice capabilities to existing SIP contact centers or IPBX (FreeSWITCH / Asterisk), with both orchestrated ASR→LLM→TTS and end-to-end Omni, and first-response latency consistently under 1 second.
Non-invasive rollout: keep existing SIP/IPBX trunks and architecture, then enable AI by scenario
No-code delivery: configure agents, scripts, and routing policies entirely in the console
Deploy privately: keep call data local with RBAC & audit trails
No legacy-system changes · No coding · Go live with network connectivity
Live Console
Dual Engine · Observable · Auditable
Inference engines
3-stage orchestration
Swap ASR, LLM, and TTS independently for fine-grained control.
Omni end-to-end
Native speech modeling shortens the path and lowers first-token latency.
End-to-end latency breakdown
3-stage example: VAD / ASR / LLM / TTS
VAD
35ms
ASR
240ms
LLM
430ms
TTS
210ms
* Sample data for illustrating observability.
Concurrency
12
Error rate
0.1%
Pipeline
<1s steady
Give traditional SIP contact centers / IPBX an AI brain
No architecture rewrite and no replacement of existing systems. Configure, deploy, monitor, and iterate in one console with seamless 3-stage and Omni switching.
Go live fast
Keep your existing SIP/IPBX architecture and launch the first AI voice agent in 5 minutes once network is connected.
Visual configuration
No-code composition for ASR/LLM/TTS and dialog strategies. Drag-and-drop flows available (optional).
Dual-engine routing
Run both 3-stage and Omni in one tenant, and route by scenario for best performance.
Secure and controlled
Private deployment supported. 100% of call data can stay on-prem.
From dialing to post-call review in 3 steps
Dial → Hear AI response → Review logs, memory writes, and <1s latency breakdown for both 3-stage and Omni.
Demo uses simulated data to illustrate observability.
Dial
SIP
INVITE → 200 OK → RTP
Supports orchestrated ASR/LLM/TTS or end-to-end Omni mode
TTS
Play AI voice
AI Response
Shows VAD/ASR/LLM/TTS outputs in real time
Idle
—
Observability
—
You'll see
• First-response latency breakdown (every segment visible)
• Knowledge/memory evidence (sources and write records)
• Exportable review (JSON/logs) for PoC and compliance
Control comes from dual-mode architecture
Without changing traditional SIP contact-center or IPBX architecture, every hop from SIP ingress to model calls remains replaceable, observable, and auditable; choose 3-stage or Omni based on business goals.
Pipeline overview (uplink/downlink)
Click a mode card or KPI chip to switch the flow diagram.
Uplink (caller → AI)
Downlink (AI → caller)
1. Caller
2. SIP Gateway
3. VAD
4. ASR
5. LLM + mem0/MaxKB
6. TTS
7. Voice back
Give AI the call volume. Keep the outcomes.
Example layout only. Replace with real customers and real numbers after launch.
Typical customers
Finance · City bank
E-commerce · Support center
Gov · Hotline
Finance outbound
Connect rate +37% · Labor cost -¥2.1M/year
Automate high-frequency outbound calls and intent collection; agents focus on high-value users.
E-commerce after-sales
Auto-resolution +52% · Avg. call duration -18%
Answer high-frequency questions and sync tickets so the team can focus on complex cases.
Gov hotline
Peak queue -35% · Satisfaction +0.6
Handle peak traffic with consistent answers to avoid congestion and inconsistency.
"We now manage multiple trunks in one console. Debugging moved from listening to recordings to reading traces."
"Models and knowledge can be swapped independently—no need to redo SIP integration."
No voice-algorithm team required—bring AI brains to your support system
We set up SIP ingress, model pipeline, observability, and governance so you can focus on scripts and knowledge.
Private/hybrid deployment · PoC checklist available · Delivery timeline depends on scope