Products

vConductor

vConductor is SIPez's event-driven conversational data platform for capturing, structuring, processing, and operationalizing voice communications. It transforms raw SIP calls into structured vCon records that unify media, metadata, transcripts, and AI-derived insights in a portable, standards-based format.

Built for service providers, integrators, and product teams, vConductor enables real-time and lifecycle-aware processing of conversational data through APIs, event-driven workflows, AI analysis, and downstream integrations. It provides the backend infrastructure needed to support compliance recording, conversation intelligence, workflow automation, and AI-powered communications applications.

Core vConductor Capabilities

• SIP-based conversation capture
• Automatic vCon generation and structured conversation records
• AI transcription, labeling, summarization, and enrichment
• Event-driven processing pipelines and workflow triggers
• REST APIs for search, retrieval, processing, and integration
• Short-term and long-term storage options
• Compliance, retention, and lifecycle management support
• Single-host, containerized, and distributed deployment models

Media, Voice, and Communications Platforms

In addition to vConductor, SIPez continues to offer proven SIP, media, and communications platform components for customers building or extending voice infrastructure products. These capabilities draw on decades of experience in VoIP, SIP, media processing, high availability systems, and interoperability.

Wideband and HD Voice Conference Bridge, IVR Engine, and Applications

SIPez's ezIVR interactive voice response engine supports wideband and HD voice audio for conferencing and IVR applications. We can provide standard IVR applications or customize conference and IVR applications to satisfy your needs. Solutions range from small-scale single-system deployments to large-scale multi-host, high-availability environments. See the ezIVR product sheet for details.

Audio Processing

• Media Processing Engine
• RTP/RTCP
• Adaptive Dejitter
• Packet Loss Concealment
• Voice Activity Detection
• Active Gain Control
• Acoustic Echo Cancelation
• Conference Bridge Mixing
• Speaker Selection
• Side-bar Conversations
• Whisper Mixing
• Plugable Codecs
• Audio File Playout
• Audio Buffer Record and Playout
• Selective Recording

SIPez has a full-featured audio processing subsystem including RTP and RTCP support. The subsystem provides plugin codecs, audio file streaming, echo cancelation, bridge mixing, stream recording, adaptive dejitter, and lost packet reconstruction. These capabilities support high-quality audio performance even in adverse network conditions and are suitable for embedded systems, DSP-assisted platforms, media servers, gateways, and software-based voice applications.

Test Network Simulator

SIPez's Cumulus Network Simulator and related tools provide the ability to generate, capture, and reproduce adverse network conditions where packet loss and delay diminish audio quality and reliability. This is useful for validating improvements enabled by adaptive dejitter and packet loss compensation, or for providing complete performance testing services for voice products.

Hosted VoIP

SIPez provides solutions to power next-generation SIP hosted voice providers. Combined with our consulting and integration services, we can help deliver complete solutions for hosted communications platforms.

Media & Conference Servers

We provide enterprise-class media server solutions to handle interactive voice response (IVR) requirements. Depending on your needs, solutions range from fixed implementations to open and flexible VXML-based scripts.

SIP conference bridge functionality is also available. Please contact us with your port density and deployment requirements.

PBX Functionality

SIPez can help you leverage ready-made and tested PBX solutions for both server and embedded environments. We have experience building, customizing, and deploying solutions across Linux, Solaris, and Microsoft Windows environments.

Phones, Gateways, and Voice Endpoints

SIPez provides proven SIP endpoint and gateway software components for customers building communications products such as hard phone appliances, SIP to PSTN gateways, terminal adapters, media servers, IVR servers, and conference bridges. Our components are well tested, interoperable, and economical, allowing you to focus on your product differentiation rather than rebuilding core voice infrastructure.

• Call Processing
• Media Processing
• Conference Bridging
• IM
• REFER-based blind and consultative transfer
• STUN and ICE based NAT traversal
• Third Party Call Control

Proxy Servers

SIPez provides configurable SIP proxy, registrar, and redirect server technology. These servers can be extended to provide application-specific features through plugin APIs, customized for OEM solutions, and adapted for production deployments requiring RFC 3261 compliance and interoperability.

SIP Stacks

SIPez has experience with open source SIP stacks such as JAIN SIP, reSIProcate, and sipX stack, and can help you understand how to apply these or your preferred commercial stack. We understand the capabilities and tradeoffs of leading SIP stacks and can help you choose, integrate, and extend the right stack for your product or application.