The fastest way to stream SIP calls to your SIP Recording Server or AI engine in real-time.
Custom SIP Recording &
Real-Time Streaming Solutions
Most AI tools struggle to plug into traditional SIP-based dialers.
Individual integrations are slow. Compatibility is unpredictable.
And real-time analytics? Very complicated.
We fix that.
Our SIP Recording & Streaming Gateway acts as a universal translator, capturing SIP audio mid-call, recording it, and streaming it straight to your AI engine.
No changes needed on your dialer.
No delay.
No disruption.
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Key Features of Our Custom SIP Call Recording Solutions
Everything you need to connect your SIP calls to live AI engines.
We engineer every gateway to match your infrastructure, codecs, security needs, and downstream AI platform.
Real-Time RTP Forking
Audio is streamed as it happens; clean, compliant, and compatible with PCMU/ulaw codecs.
Softphone + WebRTC Support
Support SIP 2.0 and WSS-based WebRTC, with no changes required to the existing VoIP Infrastructure.
Built-in High Availability
Deployable in active-passive HA setups using Pacemaker or Keepalived.
Infra lacks recording? Get the perfect compliance call recording system (through SIP REC)!
Benefits of Custom SIP-to-AI Integration
Real-time insights, simplified compliance, and zero friction with your existing stack.
Works with Your Existing Dialers
No need to build dialer-level integrations or swap out softphones.
Smarter Coaching and QA
Enable AI models to analyze agent-customer interactions with both audio and metadata.
How Our Custom SIPREC Call Recording Solutions Work
Simple flow. Deep integration. Built to scale.
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. Media Engine modifies the SDP to become the media bridge.
SIPREC Session Begins
Once the call is active, the SIPREC INVITE is sent to the SRS (SIP Recording Server) from the SRC (SIP Recording Client). Upon acceptance of the INVITE packet, RTP streaming to the Recording Server begins.
Streaming to AI
Single-channel or dual-channel RTP streams are sent to the SIPREC Server, where call recordings are stored. RTP streams can also be pushed to the AI engine in the prescribed format.
Webhooks Fire
Call metadata is sent at the start and end, including the caller's number, callee's number, agent's name, timestamp, call ID, and more.
SIP Call Recording Use Cases We Empower
If you’re running call analytics, voice AI, or compliance recording over SIP, we can solve your core infrastructure bottlenecks.
Compliance-Driven Contact Centers
Enable secure SIP call recording that works with internal PBXs and VoIP switches.
Custom VoIP Deployments
Add SIPREC-based media forking to internal platforms that need smart call recording or audio routing.
Why Teams Choose Ecosmob for SIP Call Recording Solutions
Because real-time call streaming isn’t just about SIPREC, it’s about building it right for your stack.


Need real-time call streaming without disrupting your SIP flow?
SIP-to-AI Streaming Across Dialers
A quick look at an Ecosmob custom deployment.
A leading AI conversation analytics company needed to stream live SIP audio from multiple IP PBX and call center platforms to their AI engine for real-time transcription, call analysis, and agent coaching.
Integrating with each dialer individually was time-consuming, error-prone, and didn’t scale.
We deployed a VoIP proxy engine gateway with media engine and SIPREC support.

The system now:
- Translates WebRTC signals into SIP for call functions
- Proxies SIP calls from softphones
- Redirects RTP streams to SRS and to the clients’ AI engine
- Pushes webhook metadata to SRS so that against each call recording, data can be populated, and the same can be transmitted to the AI Engine to enrich AI processing
Our SIPREC Client enables real-time audio and metadata streaming from their VoIP servers to the SIPREC Server and then to the AI engine, requiring zero changes to the existing dialers or VoIP infrastructure.
Build a Smarter SIP Bridge
You don’t need to change your dialers.
You don’t need to rewrite how your softphones connect.
You just need a smarter SIP recording and streaming layer.
And we’ll build it for you.
SIP-compliant, built to connect with your AI engine, and custom-fit to your tech stack.
FAQs
How can I stream SIP call audio to an AI engine in real time?
You can use a SIPREC-enabled gateway that forks RTP audio mid-call and sends it to your AI system over a secure IP:Port stream, using supported codecs like PCMU or ulaw.
What metadata can be captured during SIP call recording and streaming?
Metadata includes call ID, caller and callee numbers, agent name, start and end timestamps, total call duration, and active call duration, all delivered via webhook in real time.
Are SIPREC-based recording systems secure and compliant?
Yes. SIPREC uses standards-based SIP signaling, and gateways can be configured with TLS and codec-specific controls to meet security and regulatory compliance requirements.
Can I record SIP calls without changing my softphones or dialers?
Yes. A SIP gateway can record calls by duplicating audio in the background, without needing any changes to your existing softphones or dialers.
Key Features You’ll Love in Ecosmob’s SIP Ingress Controller
The components that together make a complete, end-to-end solution for managing SIP traffic at scale.