SIP Solution for VoIP Developers
If you are developing a VoIP project that needs reliable and cost-effective PSTN termination, you can get a significant head start by utilizing our APIs. Further, all of our back office functionality is exposed to a real-time routing client that comes pre-built into the OpenSBC open source session border controller.
OpenSBC is an ongoing attempt to create an flexible and versatile open-source Session Border Controller that is fully compliant with RFC 3261. OpenSBC can be used as a SIP router, media anchor for far-end NAT traversal, and SIP egress/ingress trunking solution, among other things. All grnVoIP/EZ Call traffic is routed through our OpenSBC cluster for realtime routing and rating, so we know it works and readily vouch for its capabilities.
Current OpenSBC functionalities include:
- SIP Registrar and Proxy
- Back-to-back UA Router
- Traffic Authentication
- Static Routing Rules
- ENUM Routing
- SIP Privacy Enforcement
- Upstream Registrar Support
- Media/Error Announcement Server
- LRN/Portability Routing Server
By combining your application with OpenSBC and our service, you can create a SIP application that dynamically routes calls to our resilient and scalable PSTN termination service, all controlled by your own application logic. For more information, please check out the OpenSBC project and our back-office API documentation.
