New DSP Architecture
24bit Digital Audio
PROFESSIONAL QUALITY AUDIO
The Systembase C510ip IP audio codec has been designed and manufactured to
deliver unparalleled performance and reliability for professional real-time audio applications over IP, ISDN, ADSL and Satellite.
The C510ip incorporates the fast apt-X Sub Band ADPCM compression system which can deliver a coding delay of only 2.8ms,
in addition to L24, L16, G.722, G711, PCMA and PCMU.
AUDIO OVER IP (AoIP)
Systembase IP Codecs have been designed to enable multiple
simultaneous connections from a single unit. 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 codecs
NEW for 2016:- SIP
The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions. The inclusion of SIP on the C510ip provides a baseline of compatibility with most manufacturers of IP based Audio equipment. This would include Codecs, tablet based App's, mobile phones and desk top computers.
FORWARD ERROR CONTROL
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 neighboring packets. The Systembase FEC process does not require any additional IP communication and consequently minimises the end-to-end audio delay.
NEW for 2016:- SIP Proxy
The proxy server is an intermediary entity that acts as a and a host for the purpose receiving and making IP based calls. The C500 series IP codec can register on a proxy server using a secure username and password. Once registered, the codec can make and receive calls with other registered devices and PSTN based telephones. A subscription based Proxy provider will allocate you a telephone number that can be dialled from anywhere in the world via another proxy provider or standard PSTN device. This gives users the ability to connect to each other using just a telephone number.
MULTI CODEC MONITORING
New for 2015 is a monitor tool that allows all Systembase C500 series codecs to be monitored from a browser window. Each codec icon allows the user to monitor the audio input and output levels together with the SYNC status of the codec. A simple click of the OPEN CODEC tag will take the user directly to the associated codecs web browser interface.
The Group Security ID feature provides a very useful mechanism for securing a group of IP codec's. This feature is enabled by entering the same 8 digit ID code into all participating codec's. The ID code is then used by the codec to automatically reject all packets that do not have a matching code.
All Systembase IP audio 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. All Systembase IP (AoIP) codec's are supported by the BNCS control platform developed by ATOS and DataMiner developed by Skyline Communications. This feature makes the C510ip audio codec ideal for integration into a automated control room environment.
A Recent addition to the C510ip IP (AoIP) codec has been the provision for selecting the effective headroom at the codec's 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.
DIGITAL AUDIO 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 C510ip audio IP codec features a dedicated Ethernet connection reserved purely for WEB management and SNMP control.
This dedicated Ethernet connection provides essential isolation from the IP (AoIP) streaming connection, simplifying the process connecting the codec to the public internet.
The management interface allows each codec to be remotely controlled using the various standard web browsers such as IE, Chrome and Firefox.
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. In addition, this feature allows the Web manager to function correctly on MAC OS.
STUDIO TO TRANSMITTER LINK WITH BACKUP
The C510ip IP codec is ideally suited for
implementing a Studio-to-Transmitter Links (STL's) for FM and DAB radio,
providing up to 22.5KHz Stereo audio over a X.21 digital leased line or IP network.
When operating on either ISDN or X.21 the codec maybe configured to provide a set of end-to-end contact closures for relay based remote management as an alternative to the traditional aux data serial port.
4 WIRE PRODUCTION COMMS (TALKBACK)
The C510ip IP codec can provide two 4 wire
productions comms audio channels for Outside Broadcasts (OB). Connections can be established using either ISDN or IP (AoIP) over ADSL or iDirect.
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 studio.
DIGITAL HYBRID FACILITY (TBU)
The Digital Hybrid Facility offers a
direct replacement for a Telephone Balancing Unit (TBU). The Dual G711 encoding
& decoding facility replaces the requirement to install additional TBU's
and analogue telephone line. The C510ip 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.
WORLD-WIDE ISDN OPERATION
The C510ip IP (AoIP) codec can 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.