Upgrade your existing system, with our on prem AI speech enhancement SIP module

Designed for offline, on prem operation
Our solutions can be deployed into any setup - and give you the modular flexibility you need.
Desk phone and soft phone compatible
Cleans up ingoing and outgoing sound
<10ms real-time latency
No internet or cloud connection
Safe level control
Trained like the brain, not like a bot
Seamless, non-intrusive use flow, avoiding change fatigue
Non-intrusive, AI Speech Enhancement.
On prem. On virtual or physical machines.
Real-time speech enhancement
Real-time speech enhancement
Sharpi SIP Bridge performs real-time speech enhancement on VoIP calls by processing RTP media streams.
Keep your operations unchanged
Keep your operations unchanged
SIP Signalling and call control remain unchanged - while your audio becomes crystal clear.
Codec and protocol as received
Codec and protocol as received
Audio is processed by Sharpi frame-by-frame, and forwarded using the same codec and protocol as it was received.
Media processing made to perform
RTP audio frames (typically 20ms are processed)
RTP audio frames (typically 20ms are processed)
Sharpi SIP bridge can process audio from even the noisiest of environments - such as emergencies.
Speech enhancement is applied in real time
Speech enhancement is applied in real time
Preserved output - crystal-clear hearing
Preserved output - crystal-clear hearing
Output RTP stream preserves codec, payload and timing.
End-to-end processing latency: ~10–20 ms
Processing latency is configuration dependent, but with Sharpi SIP bridge, your typical end-to-end processing latency will be best-in-market.
Record and reconstruct calls
Sharpi SIP Bridge supports recording of original calls. Processed calls can be recorded directly or reconstructed by post processing.
Modular and configurable deployment
Both our integrated media processing module and Standalone SIP server deployment options use standard SIP/RTP interfaces and are compatible with SIP-based telephony systems without vendor-specific extensions.
Media processing module (e.g Asterisk/SBC)
Media processing module (e.g Asterisk/SBC)
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Processes RTP frames inline, similar to a codec or DSP module
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No changes to endpoints, phones or soft clients
Standalone SIP Server
Standalone SIP Server
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Deployed as an independent SIP node
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Inserted via standard SIP trunk (SIP/RTP pass-through)
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Bypass trunk for fail-safe operation
Not sure what the right option is?
Not sure what the right option is?
Speech clarity, redefined by AI
Instant voice separation
Our Sharpi technology instantly separates voice from noise - giving you full control over what you want to hear.
Universal speech detection
Recognizes human speech across all languages, supporting global teams and diverse users.
Trained like the human brain
Not just AI, but a deep neural network, trained on 25+ years of sound scenarios.
Military grade
Delivers reliable performance for emergency and operational centers where accuracy is essential.
Effortless integration
Integrates seamlessly with call centers, headsets, and heading devices for immediate deployment.
No data transmitted
Not online, not cloud based. Absolute privacy for your operations and customers.