Choose Your Deployment Model
dSIPRouter supports multiple use cases to match your SIP infrastructure needs.
SIP Trunking Services
Problem: Connecting on-premise PBX systems (FreePBX, FusionPBX, Avaya, etc.) to SIP carriers requires complex Kamailio configuration and routing logic.
Solution: dSIPRouter provides a simple web UI for managing SIP trunk connections with support for IP-based and credential-based authentication, plus carrier interconnect routing.
- IP-based authentication
- Credential-based authentication
- Carrier interconnect routing
Hosted PBX Services
Problem: Managing multi-tenant PBX environments requires complex SIP proxy configuration and domain-based routing.
Solution: dSIPRouter proxies SIP endpoints to multi-tenant (FusionPBX) or single-tenant (FreePBX) platforms with built-in integration for scalable routing.
- Multi-tenant PBX (FusionPBX)
- Single-tenant PBX (FreePBX)
- FusionPBX integration for scalable routing
Microsoft Teams Direct Routing
Problem: Connecting Microsoft Teams to existing PSTN infrastructure and SIP carriers requires an expensive SBC and complex configuration.
Solution: dSIPRouter provides SBC functionality that connects your existing voice infrastructure and carriers directly to Microsoft Teams via Direct Routing.
- SBC functionality for Teams
- Connect existing voice infra to Teams
- Carrier-to-Teams routing
WebRTC Proxy
Problem: WebRTC clients need to connect to PBX extensions that only support traditional SIP over UDP/TCP.
Solution: dSIPRouter registers WebRTC clients to PBX extensions, acting as a proxy between WebSocket-based WebRTC and traditional UDP/TCP SIP.
- WebRTC to SIP proxy
- Browser and mobile app support
- PBX extension registration