Systel IP 16

SYSTEL IP is a “call-in” system with multiconference capability that drastically reduces the costs for this type of communications. Further, it significantly improves the audio quality, increases the flexibility and integration with already existing telephone systems at the station.

SYSTEL IP allows to connect the broadcast telephone system to the current corporate PBXs, based on IP, avoiding maintaining conventional lines exclusive for broadcast.

SYSTEL IP allows for VoIP connection of 4-wire lines from intercom matrixes or audio consoles in order to establish multi-conference circuits or external coordination in radio or TV stations. Further, in a business environment it allows for the interconnection of several meeting rooms as well as audio routing between building locations and for example simultaneous translation systems, even if these are remotely located.

Main Features:

• SYSTEL IP does not operate on hybrids, but on a 4-wire digital matrix: all the lines can intervene live and simultaneously without loss of quality.
• Significant cost savings can be obtained by connecting the entire system to an Internet telephony provider, or as extensions of the IP PBX that is already in service in the corporation.
• SYSTEL IP 16 shares the IP lines in a very flexible and dynamic way with up to 4 studios through very simple analog or digital cabling not having to deploy special and expensive audio nodes. SYSTEL IP 16, also offers many channels of local audio input and output through IP, in Dante format, compatible with AES 67.
• It is possible to create much larger installation, with dozens of studios or even for a whole network of broadcasting stations. In such scenario, each SYSTEL IP16 will be a simple set of extensions and that can share phonebooks and users.
• Possibility to set the number of audio signals arriving at the studio console, allowing for level adjustment either through this SW application or the fader of the mixing console.

SYSTELSET+

SYSTELSET+ is a control terminal based on an IP phone with touch screen running a new configurable control application which can be adapted to the most varied operating ways. The need for a PC on each work place for full functionality is avoided.

On the terminal itself, using the function keys and the touch screen, calls are dialled or accepted, put on hold or pre-listened, their send and return levels are adjusted, they can be put on air, locked on air or hung up. The operational mode can be chosen between call queue or several calls on air simultaneously. The queue is built on the lines itself with an indication of next call to be put on air. It also allows for the management of a call book and call scheduling. Lines can be shared between different programs and the layout is adapted to the number of available lines at each moment, thus making the best use of the phone’s touch screen.

Audio specifications

• Analog inputs: input impedance: 20Kohm. Electronically balanced, professional line level.
• Nominal input level: +4 dBu. Max. input level: +24 dBu.
• Analog outputs: output impedance < 100 ohm. Electronically balanced, professional line level.
• Nominal output level: +4 dBu. Max. output level: +24 dBu.
• Digital inputs / outputs: AES / EBU interfaces, configurable as AES-3 or SPDIF. Inputs include SRC.
• AES 1 input can be used for external AES-11 synchronization.
• There are also Dante, AES 67-compatible inputs and outputs. Dual IP LAN interface compatible with Dante native redundancy. Synchronization is transported through the network.

Audio processing

• Phone audio in G.711, G.726, G.729, 50Hz – 3KHz.
• High-Definition audio with G.722 algorithm: 50Hz – 7KHz.
• Echo cancellation. Automatic gain control.
• Independent, digital gain control for all inputs and outputs with an adjustment range of +/- 12 dB and muting.
• Automatic gain control for telephone returns.

Inputs and outputs

• DB15 female audio multi-connectors. Two I/O each.
• 2 analog balanced inputs.
• 2 analog balanced outputs.
• 2 digital AES- EBU (AES3 or SPDIF) dual inputs.
• 2 digital AES- EBU (AES3 or SPDIF) dual output.
• 1 WAN IP port for 16 VoIP lines, plus 4 VoIP lines for control phones.
• 2 LAN IP ports for control and 32 AoIP inputs / outputs in redundant Dante / AES-67 format.
• 3 DB15 connector for 4 opto-coupled GPI and 4 GPO each one.

General characteristics

• Power supply. Universal 100-240 V. 50/60 Hz. 50 VA power supply.
• Silent operation: natural convection cooling.
• Weight: 4 Kg (8,8 lbs).
• Width: 482 mm (19″) 1U rack height = 44 mm. (1,75″).
• Depth: 356 mm. (14″).