AEQ Systel IP
Talkshow System
The AEQ SYSTEL IP is a “call-in” system with multiconference capability that drastically reduces the costs for this type of communication. Further, it significantly improves the audio quality, increases the flexibility and integration with already existing telephone systems at the station.
SYSTEL IP allows connecting 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.
- 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 IP16 shares the IP lines in a very flexible and dynamic way with up to 4 studios through very simple analogue 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 a much larger installation, with dozens of studios or even for a whole network of broadcasting stations. In such a 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.
- Analogue inputs: input impedance: 20Kohm. Electronically balanced, professional line level.
- Nominal input level: +4 dBu. Max. input level: +24 dBu.
- Analogue outputs: output impedance < 100 ohms. 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 is compatible with Dante native redundancy. Synchronization is transported through the network.
- 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.
- DB15 female audio multi-connectors. Two I/O each.
- 2 analog balanced inputs.
- 2 analogue 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 connectors for 4 opto-coupled GPI and 4 GPO each one.
- 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”).
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