The Systembase C500ip range of digital audio codecs have been designed and manufactured to deliver unparalleled performance and reliability for professional real-time audio applications. Both models incorporate the fast apt-X Sub Band ADPCM compression system with a coding delay of only 2.8ms at a sampling rate of 48KHz, as well as Linear PCM and G.722. The C510ip ISDN model also supports G.711 coding algorithm to provide compatibility with other codecs. Additional design features include 24 Bit analogue and AES audio interfaces. The analogue interface is electronically balanced resulting in a high Common Mode Rejection Ratio (CMRR) for a significant improvement in performance.

The C510ip has been designed to enable two simultaneous connections. A connection can be made or broken without disrupting other connections that may already exist. When making a connection to a remote codec, the user can specify the audio to be duplex, receive only or transmit only. If the receiving codec is not able to implement the requested audio mode due to a conflict with an existing connection, Automatic Negotiation will take place to provide a suitable operating mode. Automatic Negotiation allows the remote codec to connect without any prior knowledge of the local codec's connection status.

IP connections are not always 100% error free and occasionally the IP network will loose a packet of data. In the case of a typical broadband ADSL connection packet loss can be fairly frequent. Forward Error Control (FEC) facilitates the transmission of additional information that allows the receive codec to automatically re-create the missing packet from data contained within the neighbouring packets. The Systembase FEC process does not require any additional IP communication and consequently minimises the end-to-end audio delay.

The Group Security ID feature provides a very useful mechanism for securing a group of IP codecs. This feature is enabled by entering the same 8 digit ID code into all participating codecs. The ID code is then used by the codec to automatically reject all packets that do not have a matching code.

The C500ip Codecs provide support for SNMP (Simple Network Management Protocol). SNMP allows the codec to be managed by industry standard, third party applications that are capable of monitoring and controlling equipment over LAN and WAN environments.

A Recent addition to the C500ip range of codecs has been the provision for selecting the effective headroom at the codecs analog interface. By default the analog headroom is set at +12dBu. However, the user may now change to headroom to +18dBu from either the front panel menu or web server interface.

AES3 digital audio interface is provided on all models, and is equipped with automatic sample rate conversion to simplify the problems of connecting between a studio reference clock and the ISDN network clock. The AES3 interface is supported via two transformer isolated XLR connectors located on the rear panel.

The C500ip range of audio codecs features a built in Ethernet connection. This facility allows each codec to be allocated a unique IP address and remotely managed using the standard windows web browser. To enable a hassle free experience the WEB server pages have been developed to avoid the use of active x components that may require administrative rights to install on the local computer.

The C500ip Codecs are ideally suited for implementing Studio-to-Transmitter Links (STL's) for FM and DAB radio, providing up to 22.5KHz Stereo audio over a digital leased line or IP network. To maintain program continuity, an automatic ISDN backup is initiated on failure of the primary signal path. The restoration of the primary signal path occurs automatically once the fault condition has cleared. To reduce the number of ISDN lines required, the codecs can be configured to monoise the input audio during ISDN backup, and to deliver the audio to both left and right outputs.

The C500ip codecs can provide two 4 wire productions comms audio channels for Outside Broadcasts (OB). The bandwidth of the audio channels is selectable to be either 3.7kHz or 7.5kHz and enables clear communications between the OB team and the producer located at the main studio.

The Digital Hybrid Facility offers a direct replacement for a Telephone Balancing Unit (TBU). The G711 encoding & decoding facility replaces the requirement to install additional TBU's and analogue telephone line. The C500ip can either dial up or receive calls via the ISDN line(s) and connect to a TBU or an analogue telephone, and can even connect to a mobile telephone.

Systembase codecs interface directly between the audio equipment and the ISDN network via standard ISDN RJ45 sockets. To facilitate world-wide operation, 12 international ISDN standards are supported, and can be selected by the user from either the front panel or the Web Server interface.