Build your IP telephony infrastructure with Tivi SIP/RTP server

The SIP/RTP proxy is a part of our highly sophisticated and enhanced VoIP software package that combines the standard functions of a SIP proxy server and a SIP registrar with additional features to create an IP telephony infrastructure. The software has integrated RADIUS and SIP billing features, including built-in IVR functionality. Optionaly the SIP server may communicate with an external IVR server. Billing administration and the SIP proxy are managed via a Web interface and command line.

The SIP/RTP proxy is an excellent and cost-effective solution, which handles SIP signalling requests, creates a framework for call session management including RTP proxying and some useful elements of IMS and works seamlessly with standard audio and video codecs. Our software is optimised for running on a standard PC with Linux. The SIP server also complies with the RFC 3261 and supports connections from behind NATs, proxies and firewalls. The SIP/RTP proxy allows traffic anonymity during resale operations.

Routing methods:

• Least-cost routing (LCR) based on rate sheets in the billing system
• Priority routing based on quality
• Availability routing if primary routes fail

Performance benchmarks per one server unit (tested on an Intel 2.4 GHz PC)

• 40,000 SIP requests per second
• 5,000 SIP session initiations per second
• concurrent RTP sessions: 200 with transcoding or 2000 without

System security:

• IP address-based authentication
• Username and password with MD5 challenge authentication

Free SIP and RTP proxy:

• We also have a free version for smaller deployments - Download here