Media Terminal Toolkit

Our Media Terminal Toolkit (MTT) is built over an optimized, low-footprint Session Initiation Protocol (SIP) layer and is ready for implementation on reference silicon for telecom vendors. MTT is an application enabler that is reliable, architecture- and platform-independent, and easily customizable onto any customer-specified hardware platform. MTT provides much higher abstraction with simpler and fewer APIs to speed development of media terminals. MTT-built reference applications have been successfully tested for interoperability at SIPits and industry bakeoffs, and the MTT has been successfully integrated with leading third-party media managers.

Functionality

MTT toolkit enables the following functions:

Signaling functionality

SIP session control and management, which is SIP standard compliant (RFC3261)*

Media functionality

MTT provides APIs for the media manager interface and can be integrated with a third party media manager. The media interface supported by MTT has support for audio, T.38, DTMF through 2833 and video.

Management functionality

MTT provides configuration APIs, and remote management and troubleshooting support (e.g.,. statistics collection, error reporting, and multi-level tracing support). It also supports run time user driven configurations.

MTT enables a wide range of applications, including:

  • IP-based voice and video phones for the consumer or enterprise segment
  • WiFi VoIP phones
  • Dual-mode mobile phones
  • Softphone needed on PCs
  • Unified multimedia terminals

Key Features

Control Plane

  • Signaling protocol and session control to setup application level session with the SIP/IP core network.
  • In-built call session FSM to handle basic voice calls with intercept to applications at appropriate time for application specific function/override.
  • Registration, re-registration and de-registration with the SIP/IP core.
  • SIP signaling based session control compliant with RFC 3261.
  • Subscription and notification
  • IPv4 and IPv6
  • Subscriptions to any Event Package
  • Compact format of SIP headers
  • Multiple Line on terminal
  • INFO and OPTIONS
  • Receiving DTMF digits in RTP as per RFC 2833
  • Call hold, resume, mute, unmute, transfer, forwarding, and waiting

Media Plane

  • Interface with a media manager for the transfer of media in the network
  • Creation of media stream using RTP as media transport protocol
  • Audio/Video codecs
  • Media management functions (Adaptive Jitter Buffer, VAD, CNG, Silence Suppression, Packet Loss Concealment) as required

Configuration Plane

  • User IDs, proxy address, registrar address, service activation/subscription,
  • Scalability parameters like number of calls, number of lines, number of users etc.