Phoenix Stratos

Comprehensive, redundant, easy to use and controllable – for your station or network communications! This AudioCodec is the solution for any broadcaster with the need for a versatile but yet straight forward and efficient AudioCodec.

Features:

• Capacity to establish two independent stereo/dual connections or four mono channels to several different destinations.
• Fully independent dual channels for program, coordination or backup, each one with its respective returns.
• Front panel user interface and and control software.
• Local or remote control of a single or a “pool” of units.
• Remote monitoring: includes SNMP server that allows the user to monitor status, alarms, etc.
• Multi- and Unicast capability.
• Ancillary data channel both in IP and ISDN.
• 4 GPIs and GPOs.
• Includes encoding algorithms in compliance with N/ACIP EBU Tech3326 recommendations plus the AEQ LD low delay algorithms. AAC is optional.
• Balanced analogue audio Inputs and Outputs on XLR connectors and AES/EBU digital I/O’s

Front & Rear Panels:

1. Function Keys
2. Multi-function display with simple & intuitive configuration / operating menus
3. Escape
4. Rotary navigation encoder
5. Alphanumeric Keypad
6. VU meters
7. Comms status info and control
8. Headphone volume control
9. Headphone jack
10. 2 x L & R analog audio Inputs and outputs
11. IP interface
12. 2x AES EBU digital stereo inputs and outputs
13. X.21/V.35 interface
14. 4GPI & 4GPO
15. ISDN interface Euro + National 1
16. Serial Port
17. Power supply Redundant option

Applications:

• STL Link (Studio-Transmitter Link) – Through IP connections on private VLAN, IP Radio-links, WiMAx, WiFi, ADSL, Cable MODEM, etc., it is possible to send up to two stereo or four mono programs to the radio transmission sites (one or two destinations) and to remote control and supervise from the production centre.STL links can also be built using point-to-point V35/X21 synchronous links, switched ISDN, or IP connections, with optional backup on a synchronous switched ISDN network.
• Radio station networks – The unit can be used to interconnect the audio of the different radio stations in the network through IP. Since the unit provides bidirectional audio, a signal can be distributed to the different radio stations in the network and at the same time audio contribution can be established in the opposite direction.IP Multiple unicast mode allows a single STRATOS codec to send up to two different programs, each one to a group of correspondents, while receiving feedback from one in each group. This way, the number of required audiocodecs at the network headquarters can be reduced.Radio station networks can also be built using synchronous point-to-point V35/X21 links or switched ISDN networks.
• Outside broadcasts and contributions – The unit is able to establish IP connections with other Phoenix and Phoenix smartphone codecs (Stratos, Studio, Mobile, Mercury, PC, Pocket or Lite) or codecs from any other manufacturer that are N/ACIP compliant. Audio contribution can be performed on assorted IP networks, such as private VLANs, IP radio links, WIMAx, WiFi, ADSL, cable Modem, Inmarsat or similar IP satellite links, etc.It can be connected via ISDN with most audiocodecs on the market including, of course, AEQ Phoenix Stratos, Mobile, Studio, Eagle, Course ISDN, SWING, M-PAC and TLE02. The integrated ISDN module features both S and U interfaces on RJ45 and RJ11 connectors, and internally supports Euro ISDN and National 1 protocols for worldwide usage.

Control Software

Each STRATOS unit includes all the necessary controls for its operation and configuration on the front panel, so a PC is not required to setup or use it. However, studio codecs are often placed in rack/IT rooms or in a remote location not quickly accessible. For that reason, each unit is supplied with an individual control and configuration PC application, including friendly, well differentiated setup and operation windows. The application allows the user to work with the unit (the same as pressing the physical buttons on the unit), selecting encoding modes, connection methods and establishing, answering and ending calls. All configuration and operational functions are presented in a very intuitive way.

The software includes audio presence indicators as well as remote real-time VU-meters allowing the monitoring of incoming and outgoing audio levels for each device, no matter where they are physically
located. It also provides a call book management application with copy functions that allows the
user to generate a central contacts list and individual subsets for each codec in the network. If
desired, it is possible to open as many instances of the application as required in order to control all the
audiocodecs in your network from a single PC. If the quantity of units increases, purchasing the full license for multi-codec control becomes more convenient, enabling well organized exploit of the codecs pool from a single program instance.

Other features

• Automatic backup, selectable between IP, V35 and ISDN.
• Multicast IP: transmission and reception.
• Multiple-unicast in RTP-raw mode: allows the unit to send a same stream to up to 10 different IPs.
• SIP: according to EBU-Tech 3326 recommendation. Possibility of operation with or without SIP Proxy server.

Encoding algorithms

• OPUS with Fs= 48KHz, mono, stereo, with 3 mono and 4 stereo presets. Bit rates between 12 and 256Kbps. Audio bandwidth between 6 and 20 kHz.
• G711 A law, µ law (64kbps, low delay, 3.5 KHz audio bandwidth).
• G722 (64 Kbps, low delay , 7 KHz audio bandwidth).
• AEQ-LD Fs=16, 32 or 48KHz, mono or stereo. Bit-rates between 64 and 384 Kbps, audio bandwidths between 7 and 20 KHz.
• MPEG1 & 2-LII, Fs between 16 and 48 KHz, mono, stereo, dual channel and joint stereo. Bit-rates between 64 and 384 Kbps. Audio bandwidths between 10.5 and 16.5 KHz.
• AAC-LC* high quality, with Fs=24, 32 and 48 KHz, mono, stereo, MsStereo, bit-rates between 32 and 256 Kbps, audio bandwidths between 9 and 20 KHz.
• AAC-LD* high quality and low delay, Fs=48 KHz, mono, stereo and MsStereo. Bit-rates between 32 and 256 Kbps, audio bandwidths between 8 and 20KHz.
• PCM (linear) very low delay and transparent quality. Fs=48 KHz or 32 KHz @ 12, 16, 20 or 24 bits/sample, mono or stereo (bit-rates between 576 and 2304 Kbps), audio bandwidths between 15 and 20 KHz.
• Smart RTP call-initiation protocol that simplifies connection to compatible codecs.