Custom SIPREC Call Recording & Streaming Solutions
Listen Without Interrupting.
The FASTEST way to stream SIP calls to your SIP Recording Server or AI engine.
Build My SIP BridgeTrusted by 750+
The AI Data Gap in VoIP
Analyzing history isn’t the best use of your advanced AI model. Use it for now.
- The live call ends and disconnects.
- The static MP3 file is generated.
- The large audio file uploads.
- The AI analyzes the dead history.
Benefits of Custom SIP-to-AI Integration
Make your legacy dialer the smartest tool in the stack.
Real-Time Transcription & Sentiment Scoring
Works with Your Existing Dialers
Enterprise-Grade Compliance
Smarter Coaching and QA
Empower Agents with Real-Time AI Intelligence
Zero Friction With Your Tech Stack

Core Features of Our SIPREC Call Recording & Streaming Solutions
The golden missing link between telecom and modern voice intelligence.
Audio Streaming & Recording
- SIPREC-Based Call Recording
- Real-Time RTP Forking
- Dual-Channel Audio Streaming
- Encrypted Media Transport

Integration & Architecture
- Metadata via Webhooks
- Softphone + WebRTC Support
- Seamless Proxy Registration
- Built-in High Availability

The SIPREC Use Cases We Empower
For every company with the gold mine of voice logs.
AI Voice Intelligence Platforms
Compliance-Driven Contact Centers
Hybrid Agent Environments
Custom VoIP Deployments
Live Agent QA Software
Enterprise Fraud Detection
How Our SIPREC Call Recording & Streaming Solutions Work
The carefully engineered journey from a spoken word to an insight.
Client Registers
Softphone or WebRTC client registers via the VoIP Proxy Engine in pass-through mode.
Call Starts
SIP INVITE is routed to the dialer, and the Media Engine modifies the SDP to become the bridge.
SIPREC Session Begins
The SIP Recording Client sends an INVITE to the Recording Server to duplicate the session.
Streaming to AI
Single or dual-channel RTP streams are pushed directly into your designated AI engine.
Webhooks Fire
Rich call metadata is transmitted at the start and end of the session.
Insights Rendered
The AI platform takes in the packets and flashes live intelligence back to the dashboard.
Frequently Asked Questions
SIPREC (Session Initiation Protocol Recording) allows live SIP call audio to be duplicated to a secure recording server during an active session, without affecting the main call flow.
We deploy a SIPREC-enabled gateway that forks RTP audio mid-call and sends it to your AI system over a secure IP:Port stream using codecs like PCMU or ulaw.
A SIPREC recording and streaming platform captures the call ID, caller/callee numbers, agent details, and timestamps, delivering them instantly via webhooks.
Yes. A secure SIP call recording system uses standards-based signaling and strict TLS encryption to meet legal, financial, and healthcare compliance requirements.
No. The media forking happens concurrently, meaning the VoIP SIPREC audio is duplicated and streamed without adding any latency or interference to the active conversation.















