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SIP and VoIP

Voice over Internet Protocol (VoIP) is bringing about significant changes in the telecommunications industry with innovative services enabled by flexible and efficient packet transport. The convergence of IP networks and the PSTN has also led to a growing demand for circuit-based technology that is VoIP-ready.

At the heart of today’s VoIP networks is the Session Initiation Protocol, which is increasingly used for interworking functions between network services, as well as for signalling between devices. NMS offers an extensive set of board and software component and voice coding technologies for developers designing SIP-based VoIP solutions.

  • The SIP API for Natural Access, based on the popular Natural Call Control API, allows developers to create applications that can easily operate in either the PSTN or VoIP domains, or both simultaneously.
  • The CG Series of boards offers scalable, high-performance development platforms for converged PSTN and IP telephony solutions, designed to meet the connectivity, flexibility, and performance requirements of new applications such as VoIP gateways and IP media servers.
  • PacketMedia HMP gives developers a comprehensive software-only solution on a standard x86-architecture computer for creating a wide range of powerful IP media server applications ranging from announcement servers to voicemail and more.

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