KEY FEATURES
New DSP Architecture
24bit Digital Audio
32bit Signal Processing
Selectable headroom
(+12dBm or +18dBm)
Audio Over IP
L16, L24, APTX, G.722
G711, PCMU, PCMA
FS 8KHz to 48KHz
Forward Error Control
Digital Clock Recovery
10/100 Base-T
Unicast, Simulcast
Multicast (IGMP V2)
DHCP
SIP
SIP Proxy
Fast Bootup
Ethernet Support
Dedicated IP (AoIP) port
Dedicated Web Server Port
Remote Management
SMTP Email Alerts
Backup User Settings
Restore User Settings
Firmware Updates
SNMP Management
SNMP Traps
DHCP
Multi-Algorithm
G.711 (Telephone)
PCMA (Telephone)
PCMU (Telephone)
G.722
APTX (64->384Kbps)
L16
L24
ISDN
ISDN: 128Kbps
Dual G.711 (Telephone)
Dual PCMA (Telephone)
Dual PCMU (Telephone)
Dual G.722
APTX
Automatic Backup/Restore
External Trigger Dial
12 ISDN Standards
Automated Backup
X.21->ISDN
X.21->IP (AoIP)
IP (AoIP)->ISDN
AES3
Transformer Isolated
XLR
User Interface
24x2 Character VFD
Full VU Metering
Back Space Key
Telemetry Features
2 x End-End Relays
9600,4800,2400 Aux Data
C510ip-S
PROFESIONAL QUALITY AUDIO
The Systembase C510ip-s 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-s 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.
The C510ip-s is fully compatible with the existing C310xr ISDN codecs when operating over ISDN and X.21 circuits. In addition, the C510ip-s also supports Dual G.711 and Dual G.722 coding to provide enhanced compatibility with ISDN codec's from other manufacturers. 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.
Following a power failure, the C500 series of audio codecs can fully reboot to the ready state in approximately 1 second.
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 connection status.
In the event of power failure the C510ip-s can re boot, negotiate and restore an IP Audio connection to its streaming state in approximately 10 seconds. This feature is perfect when the codec is being used for commercial applications such as transmitter links or live commentary.
HEADROOM ADJUSTMENT
A Recent addition to the C510ip-s 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.
SIP
The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions. The inclusion of SIP on the C510ip-s provides a baseline of compatibility with most manufacturers of IP audio codecs. This would include hardware codecs, tablet based App's, mobile phones and desk top computers.
All Systembase SIP codecs have been certified for use with the SIPitPro cloud based plaform.
Multi Codec Monitoring
Easymon 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.
SNMP MANAGEMENT
All Systembase IP (AoIP) 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-s audio codec ideal for integration into a automated control room environment.
WEB SERVER
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.
All C500 series IP codecs have a facility to apply firmware updates via the WEB interface. This feature allows for hassle free maintenance in environments where physical access is not always possible. During the update process the upgrade file, typically 400Kb, will be checked for integrity prior to being flashed into the system. Program disruption is minimized to approximately 20 seconds.
LINK SECURITY
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.
4 WIRE PRODUCTION COMS (TALKBACK)
The C510ip-s 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.
If using IP (AoIP) for the connection, the bandwidth required for a basic 3K7 talkback circuit is only 83Kbps.
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.
On the latest version of the codec firmware it is possible to independently control the dialing of individual ISDN B channels from the WEB browser interface.
WORLD-WIDE ISDN OPERATION
The C510ip-s 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.