Custom SIP Recording &
Real-Time Streaming Solutions

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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.

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      SIPREC-Based Call Recording

      Duplicate live RTP streams using SIPREC & send them to a designated recording server or AI engine.

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      Metadata via Webhooks

      Structured call context sent in real time: agent name, timestamps, call ID, caller/callee, call durations.

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      Seamless Proxy Registration

      Calls and registrations pass through transparently to your existing Dialers and PBXs.

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      Real-Time RTP Forking

      Audio is streamed as it happens; clean, compliant, and compatible with PCMU/ulaw codecs.

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      Softphone + WebRTC Support

      Support SIP 2.0 and WSS-based WebRTC, with no changes required to the existing VoIP Infrastructure.

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      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.

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      Real-Time Transcription & Sentiment Scoring

      Send audio to AI as it happens, not after the call ends.

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      Enterprise-Grade Compliance

      SIPREC and TLS-based transport ensure you meet call recording regulations without compromising call quality.

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      Empower Your Agents with Real-time AI Intelligence

      Our solutions stream live agent-customer conversations to an AI engine, instantly displaying transcriptions on the dialer screen and providing relevant, in-the-moment answer suggestions to guide agents.

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      Works with Your Existing Dialers

      No need to build dialer-level integrations or swap out softphones.

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      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.

    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.

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      AI Voice Intelligence Platforms

      Feed live call audio and context into transcription, scoring, and AI coaching engines.

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      Hybrid Agent Environments

      Unify recording and streaming across a mix of WebRTC and traditional SIP environments.

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      Compliance-Driven Contact Centers

      Enable secure SIP call recording that works with internal PBXs and VoIP switches.

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      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.

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      Designed for your dialers, IP PBXs, and VoIP Switches

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      Support for hybrid WebRTC + SIP environments

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      Real-time streaming + webhook tagging or notification

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      Customized metadata handling to match your reporting needs

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      Secure, HA-ready deployments that scale with your traffic

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      SIP recording infrastructure that fits your compliance goals with call metadata

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    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.

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    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

    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.

    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.

    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.

    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.

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