Introduction to Sprinklr VoiceConnect
Updated
Sprinklr VoiceConnect is a cloud-based voice integration layer that connects traditional telephony systems with the Sprinklr Contact Center as a Service (CCaaS) platform. It supports inbound and outbound voice communication using standard telephony protocols and ensures secure, high-quality voice interactions.
This document introduces VoiceConnect’s capabilities, supported standards, integration options, and key technical components. It is intended for system architects, voice engineers, and implementation teams working with enterprise contact center solutions.
What is VoiceConnect?
Sprinklr VoiceConnect enables your contact center to handle voice calls over IP by integrating your telephony infrastructure—such as Session Border Controllers (SBCs), SIP trunks, or on-prem PBXs—with Sprinklr's CCaaS. It is designed to work in environments where:
You bring your own voice carrier (BYOC), or
You use Sprinklr's managed connectivity.
VoiceConnect acts as a SIP gateway, handling signaling and media traffic between your telephony provider and Sprinklr’s cloud services. It also includes optional encryption, codec transcoding, quality monitoring, and failover mechanisms.
Connectivity and Protocol Support
VoiceConnect supports a range of connectivity models and protocols to suit different deployment needs:
Supported Connectivity Models
Feature | Description |
Bring Your Own Carrier (BYOC) | You retain your existing telecom provider. Sprinklr helps establish SIP trunks and configure secure tunnels between your SBC and VoiceConnect. |
Standard SIP Connectivity | VoiceConnect uses SIP (RFC 3261) for establishing, managing, and terminating calls. Compatible with any SIP-compliant SBC. |
IP to IP Connectivity | Direct IP routing is supported for SIP traffic without using media gateways. Useful for low-latency or private network environments. |
Direct Connect / MPLS | For enterprise deployments, dedicated MPLS links or private peering can be used to reduce packet loss and jitter. |
TLS and IPSEC VPN | Tunnels can be configured using TLS or IPSEC to secure signaling and media traffic over public or hybrid networks. |
WebRTC Support
VoiceConnect supports browser-based calling using WebRTC:
Click-to-dial from Sprinklr UI.
Secure communication over public internet or VPN.
Support for STUN/TURN servers for NAT traversal.
Two-factor authentication for agent access.
Codec negotiation and transcoding for interoperability.
Reporting and Monitoring
VoiceConnect provides real-time visibility into voice quality and operational status:
Feature | Description |
MOS and Quality Score Reporting | Tracks call quality using standard voice metrics like MOS (Mean Opinion Score). |
Call Detail Records (CDR) | Detailed logs of call events, durations, and routing decisions. |
Voice Debug Console | Interactive console to trace SIP flows, diagnose call issues, and analyze media paths. |
Real-Time Alerts | Automatic alerting for call failures, quality drops, and configuration issues. |
Packet Capture (On Demand) | PCAP-level inspection available to troubleshoot media issues or investigate call anomalies. |
Security and Compliance
Security is built into VoiceConnect’s architecture with multiple controls:
Feature | Description |
High Availability and Failover | VoiceConnect supports automated failover between SIP trunks and regions. |
DDoS Prevention | Traffic filtering and rate limiting help protect against denial-of-service attacks. |
IP Whitelisting (ACLs) | Only trusted IP ranges are allowed to send SIP traffic. |
Intrusion Detection and Prevention (IDS/IPS) | Inline threat detection and mitigation for SIP and RTP traffic. |
TLS 1.2 Encryption | All signaling is encrypted using TLS 1.2 by default. |
Data Handling Policies | Compliance with regional voice storage and transmission standards (e.g., GDPR, HIPAA). |
Integration and Features
VoiceConnect supports a range of integrations and optional capabilities:
Feature | Description |
Microsoft Teams (Direct Routing) | VoiceConnect can route SIP traffic directly to Microsoft Teams tenants. |
Custom SIP Header Support | Add or modify SIP headers for integration with downstream systems. |
Speech Profiles and Transcription | Real-time voice recording with near real-time transcription support. |
Voicemail Recording | Store and retrieve voicemail messages via the Sprinklr interface. |
SIP REFER Support | Enables seamless call transfers using the SIP REFER method. |
Early Media & IVR | Stream prompts or announcements before a call is answered to improve user experience. |
Supported RFCs
VoiceConnect is compatible with standard SIP and RTP specifications, including:
RFC 3261: SIP: Session Initiation Protocol
RFC 3262: PRACK (Provisional Response Acknowledgement)
RFC 3264: SDP Offer/Answer Model
RFC 3311: UPDATE Method
RFC 3325: P-Asserted Identity
RFC 2833: DTMF Relay via RTP Payload
RFC 3960: Early Media
RFC 5009: NAT Keepalive Mechanisms
Deployment Value
VoiceConnect simplifies voice system integration by providing:
Centralized control of SIP routing, media handling, and monitoring.
Consistent call quality through codec negotiation and network optimization.
Tools to trace, monitor, and debug voice traffic without third-party systems.
Global scalability through distributed Points of Presence (PoPs).
Standards-compliant security and encryption options.
Sprinklr VoiceConnect is designed for organizations that need to connect existing telephony infrastructure with Sprinklr’s cloud contact center. It supports flexible integration models, strong security features, and in-depth diagnostics—all essential for modern, voice-enabled customer service environments.
For configuration details, refer to the VoiceConnect Quickstart Guide.