1 SYSTEL IP 12 SYSTEL IP 4 IP TELEPHONY FOR LIVE RADIO AND TELEVISION TALK-SHOWS AND MULTICONFERENCE APPLICATIONS HD VoIP SYSTEMS
2 Product requirement 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. The investment required is very small and will be amortized very rapidly through simple saving of costs. Business telephone systems are rapidly migrating to VoIP technology, integrating IP switchboards, Asterisk or similar type distributed, or virtual, allowing access to new alternative telecommunication service providers. At the same time telephony or call-in systems for broadcast applications have so far been an isolated island with important operational costs and stagnant technology. Background Multi-line telephone systems for the broadcasting industry have been available for more than 20 years. AEQ has continuously been offering innovative solutions and in line with the available technology: In 1994 AEQ developed the Systel 3000 conference system with control for digital telephone hybrids on conventional telephone lines in console multiplex format. Other manufacturers were offering call-queue oriented interfaces. In 2004 AEQ launched the Systel 6000, with a new architecture: High quality ISDN lines with AudioCodecs, POTS, leased lines and point to point-to-point IP Audio (RTP). The system incorporated a 4-wire digital matrix that allows console format multiplex and multi-conferencing of up to 40 different channels. Now, AEQ presents its third generation of SYSTEL talk-show and multi-conference systems. The system continues being built around a digital router and we are using lines from IP telephony systems implementing a flexible and dynamic control. Further, the call-in-queue is controlled simultaneously with the multiplex functions for the comfort of all users. Four basic concepts regarding VoIP In order to understand the technology behind the SYSTEL IP, we will try to explain the four basic concepts with regards to its working environment. IP Telephony or VoIP Currently, the way to functionally enable communications between conventional telephony and the majority of private switchboards (PBX): if a call is generated with a conventional phone, the generated audio signal will be converted into a digital signal, compressed and encapsulated under internet protocol (IP) within a gateway. From the gateway the signal will be forwarded to the recipient s phone within a computer network - WAN. The audio that reaches the conventional phone, has previously traveled through the network to the "gateway" where IP packets have been converted into audio for the earpiece. If the phone is an IP phone, the phone itself generates and receives IP packets. In this way telephone voice signals are converted into and treated as computer data and flows through Internet networks via switches, routers, ADSL lines etc... Asterisk At this point, it is not difficult to imagine that a telephone is a kind of computer with specialized software. The manufacturer Digium has provided the community with a powerful IP PBX software application for Linux called Asterisk - open source and free! A vast majority of existing PBXs are based on Asterisk code and there are many technicians everywhere in the world capable of configuring very powerful and cost efficient IP PBX systems: Simple PC software based on Asterisk, which has become the heart of many professional facilities, including the most prestigious brands. VoIP Providers These area Internet-based service providers that are able to route calls through the network with access to traditional telephony in different cities and countries, allowing international rates adjust to a little more than the cost of a local call. The VoIP service providers (their services) are accessed through a trunking IP (Internet access: DSL, cable modem, fiber optic, WIMAX...). Some offer virtual PBX service: connect all IP phones to an office trunking with a switch without the need of a switchboard. SIP SIP is a signalling protocol for VoIP (Voice over IP) to route calls between locations and equipment. SIP is also available in the AudiCodecs that meet the N / ACIP EBU (European Broadcasting Union) and in many Softphones that allows establishing calls from computers, PDAs, etc. using the telecommunication companies data networks. SIP allows both partners to negotiate and establish audio codecs high quality calls (HD) if both phones support it. 2
3 Main features SYSTEL IP does not operate on hybrids, but on a 4-wire digital matrix: all the lines (4 or 12 depending on model) 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, complementary or alternative to a conventional telephone company. The relatively low cost of the equipment makes it profitable even by just replacing the analog telephone hybrids in a studio with a SYSTEL IP4, turning the phone lines into IP telephony through an FXO gateway. The SYSTEL IP12 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. Based upon the free software ASTERISK, it is possible to create much larger installation, with dozens of studios or even for a whole network of broadcasting stations. In such scenario, the SYSTEL IP4 or IP12 will be a simple set of extensions. The SYSTEL IP control terminals are extremely powerful, flexible, economical and practical: A handset (phone terminal) to respond to calls and a Web browser that can be operated from any PC. If using an ipad or a tablet instead of a PC, there is a customizable desktop control terminal which allows all the usual functions of a talk show without having to use a PC. AEQ has designed a convenient swivel stand for Tablets or ipad available as an option. Multiple control terminals with internal and individual labelling and chat lines can be used in a studio, thus dividing the work among producers, technicians and presenters. Control format can be multiplex console or call queues, allowing the assigning of VIP correspondents on exclusive fader. Possibility to set the number of audio signals arriving at the studio console, allowing for level adjustment either through ths SW application or the fader of the mixing console. System parts SYSTEL IP4 SYSTEL IP12 The heart of the system is a 19" rack format equipment in two versions: IP4 Systel, 1RU for 4 simultaneous IP phone lines and a total of 3 inputs, 3 audio outputs and a connector for a phone "handset", typically enough to service a studio. IP12 Systel, 2RU for 12 simultaneous IP phone lines and a total of 12 inputs, 12 audio outputs, and four connectors for phone "handsets", typically enough to serve up to 4 studios. Both units behave like multi-line IP Phones with SIP signalling protocol. Compatible with Asterisk PBX, SIP Trunking and virtual PBX. Analogue and ISDN lines supported through gateways. Encryption algorithms include the proper ones in telephony: G726, G729 and low bit rate G711 with higher quality. G722 also incorporate coding with extended bandwidth to 7 khz, which characterize them as "HD" and makes them compatible with N / ACIP AudioCodecs and SIP-Phones (Any AEQ Phoenix AudioCodec and most PC telephony software). PC FOR CONFIGURATION AND WEB SERVER The operation of the system requires an application for web server control and configuration to be opened on a PC that can be shared for other usages. In this application the circuits to be used in the Studio are assigned, the agendas or phonebooks are managed and service is provided to the system control terminals for the studio operators and producers. By opening the application in a web-browser it is also possible to assume the function of a control terminal. TERMINALS The control terminals can consist in elements: one, two or three A web browser based application (essential element of the Systel IP) that runs on a PC, tablet or IPad, regardless of operating system. A phone handset that incorporates a pre-amplifier allowing to place it over 100 meters (109 yds.) from the equipment rack (highly recommended). A support for a tablet and handset (optional) If the control terminals are mounted on tablet, and that most probably won t have an Ethernet connector, these can easily connect through WiFi router to the Web server control PC. 3
4 Examples of use SYSTEL IP4 for one studio The SYSTEL IP4 behaves as a combination of 4 IP telephone terminals. Calls are received through the WAN Ethernet connection from an Asterisk PBX and / or gateways and / or through SIP trunking from the virtual exchange of IP telephony service provider. Application for configuration and control The control is provided through the Ethernet LAN by a PC in the studio in which the Web server control and configuration application has been installed. The control terminals will access the control PC via WiFi if the control terminal is installed on Tablet and through Ethernet if installed on a PC. The SYSTEL IP4 has a special connector for a phone handset. Also and since the unit has 3 outgoing audio lines, if the handset is used by the producer, the technician can establish an additional intercom circuit with an order output from the mixing console and an input from telephone to CUE. It also has an output and an input that can be used to record the phone line. If you don t need an additional CUE recorder or handset to be assigned to the system, two system faders can be assigned on your mixing console; one for a VIP guest and another one to queue the three remaining lines. TELEPHONY: ASTERISK SWITCHBOARD and/or POTS OR ISDN GATEWAY AND/OR TRUNKING SERVICE PROVIDER Wi-Fi ANCILLIARY WITHOUT TALKBACK TO CUE SPEAKER From the studio mixing console an aux bus is sent without the telephone audio in order to add the rest of the telephone lines and to be able to provide customized return for each caller. Wi-Fi OTHER STUDIOS STUDIO 2 STUDIO 3 STUDIO 4 TELEPHONY: ASTERISK SWITCHBOARD AND/OR POTS OR ISDN GATEWAY AND/OR TRUNKING SERVICE PROVIDER STUDIO TABLET Wi-Fi PRODUCER OR HANDSET ANCILLIARY WITHOUT TALKBACK TO STUDIO 1 CUE SPEAKER PC with application for configuration and control Recording of Welcome messages and Calls ADDITIONAL TABLET STUDIO TABLET PRODUCER OR SYSTEL IP12 for four studios The SYSTEL IP12 behaves as a combination of 12 IP telephone terminals that can be shared in a flexible and dynamic way between the four studios. Calls arrive by the WAN Ethernet connection from an Asterisk PBX and / or gateways and / or through SIP trunking from the virtual exchange of IP telephony service provider. The control is provided through the Ethernet LAN by a PC in the studio in which the Web server control and configuration application has been installed. The control terminals will access the control PC via WiFi if the control terminal is installed on Tablet and through Ethernet if installed on a PC. The SYSTEL IP12 has four special connectors for phone handsets. There are 8 analog and 4 digital inputs and outputs inputs on the unit, allowing, for example, that if the handsets are assigned to the producers, the technician can attend calls using the t back or order circuits and monitoring the CUE bus. Also, it is possible to set-up automatic call recorders, CUEs and Greeting messages in the selected studios and until the equipment s maximum I/O capacity have been reached. From each studio mixing console an aux bus is sent without the telephone audio in order to add the rest of the telephone lines and to be able to provide customized return for each caller. The amount of tablets or control PC is not limited by the number of handsets. Control tablet can be installed in addition to those that are linked to the handsets. The number of handsets doesn't limit the number of call positions. SYSTEL IP12 in a a studio with producers and sever Audio circuits to/from the mixing console Application for configuration and control 4 The 8 analog and 4 digital inputs and outputs on the SYSTEL IP12 allows us to send Audio to 2, 4, 6 or even more faders of the Studio Mixing console. From the mixing console an aux bus is sent in order to add the rest of the telephone lines and to be able to provide customized return for each caller. It is also possible to trigger a PLAY command through a GPO related to an Audio input channel. For example, a Welcome message or a generic WAIT audio can be useful before passing the caller to monitor the program signal (if the automatic OFF-HOOK is activated) on the phone. The Welcome Audio could be common to all the incoming lines assigned to a program. There are handset connectors for technician, presenter and two producers. These can exchange chat messages, tag lines and highlight annotations for each one of these. TELEPHONY: ASTERISK SWITCHBOARD AND/OR POTS OR ISDN GATEWAY AND/OR TRUNKING SERVICE PROVIDER STUDIO PRODUCER Wi-Fi HANDSETS CHANNELS TO MIXER Recording of Welcome messages and Calls ANCILLIARY WITHOUT
5 Product details. SYSTEL IP4 & SYSTEL IP12 Link indicators Error indicators Power indicator Analog or digital audio connectors IP, control and voice Networks GPIO connectors Handset connectors Digital audio connectors Analog audio connectors CONFIGURATION AND OPERATION SOFTWARE An important system part of the SYSTEL IP consists in an application for setup and another one for the actual operation.the setup application works on a PC, and as its name indicates is used to configure the system and managing the address or phone books. The application for control and operation runs on a web browser and operates the equipment: calls are generated or received, put on hold or queued, send and return levels adjusted, auxiliary circuits are diverted, put on air or hung up. The operating modes can be call queues or multiplex console. It also lets you tag calls, comment or establish chats between producers, presenters and control operators. 5
6 System Components 6 EXCLUSIVE COMPONENTS Engine for 4 IP lines: SYSTEL IP4 All the processing power and connectivity for 4 IP lines is concentrated into a 1RU height rack frame: IP connectors for control and voice, 2 analog inputs and 2 outputs, one pair of selectable analog or digital input / output, a handset port, 4 GPI and 4 GPO. Includes configuration software, Web server and Web client for an unlimited number of terminals. Engine for 12 IP lines: SYSTEL IP12 All the processing power and connectivity for 12 IP lines is concentrated into a 2RU height rack frame: IP connectors for control and voice, 8 analog inputs and 8 outputs, 4 digital inputs and 4 outputs, 4 handset ports, 12 GPI and 12 GPO. Includes configuration software, Web server and Web client for an unlimited number of terminals. Handset SYSTEL IP HS Handset with powered preamplifier that is connected to a special port of SYSTEL IP4 or SYSTEL IP12. It is remote powered at 48V, and has electret microphone supply. Compatible with many professional headsets (operator micro-headphones) ANCILLARY COMPONENTS In order to setup a System IP system in a particular environment, you may need to add some additional computing or telephone devices that are readily available on the market or even already installed in any office or radio / TV station. The requirements are quite relaxed, but in case you would require, AEQ can recommend or even provide those complements, certified by our System Engineering department. Switch Ethernet The equipment is connected to a network for control (LAN) and another one for VoIP (WAN). If not already created for other purposes, an Ethernet switch needs to be installed for each network. Configuration PC and control Web server Almost any PC running Windows Vista, 7 or 8 is suitable to perform this task, even shared with others required at the studio or CPD. It must be continuously running, unattended except when the system needs to be reconfigured or when massive data insertion to the call book is to be performed. Tablet - ipad The two compatible tablets for the control client are Apple ipad 2 and Samsung Galaxy TAB , as an alternative to a PC. Almost any 10 tablet with IOS or Android operating systems should work adequately, however we will be delighted to help you confirm this or otherwise find a suitable device at AEQ. WIFI access point When an Ipad or tablet is used, it may be easier to connect them to the control network through a Wi-Fi Access Point than using an Ethernet equipped docking station. There are also combined Access-Point & Ethernet switch units. We have tested D-Link DIR 615 that also has a router function, although not required for this application. SYSTEL IP ST Support for SYSTEL IP HS microheadphone and a 10 Tablet / Ipad. It is initially apt for Samsung Galaxy, Apple ipad 2 and Pad 3. Please ask us for an updated list of compatible devices However, if your Tablet / ipad is not in the list of compatible devices, please check whether the position of the tablet connections is at least compatible. In particular, check that the supply and speaker connectors have enough clearance when the device is assembled in the support. The tablet can be tilted to avoid reflections. Wiring Accessory FR CAB INP DB15 male connector to four unscreened balanced pairs of a 6 meter cable (other end open-end ), to facilitate the wiring of 2 inputs and 2 audio outputs SYSTEL IP12. Maximum 6 required per Unit. Wiring Accessory CAB FR GPIO DB15 male connector to 6 meter cable (other end open-end ), to facilitate the wiring of 2 GPI and GPO SYSTEL 2 IP 4 (max. one (1) needed per unit) or IP SYSTEL 12 (max. three (3) needed per unit). Operator s micro-headphone combination Some producers that are continuously receiving calls need the capacity to connect operator headsets. Both wired and wireless units can be found in the market. They must have a RJ-9 connector, in order to be connected as a substitute to the SYSTEL IP HS headset, which must be previously disconnected. Some wireless headsets also provide an auxiliary connector in order to be able to have both Handset and headset connected. POTS FXO Gateway Converts conventional ISDN lines into IP lines. There are models for 1, 2 and 4 basic (BRI) or primary (PRI) access. ISDN Gateway Converts conventional ISDN lines into IP lines. There are models for 1, 2 and 4 basic (BRI) or primary (PRI) access. Asterisk PBX SYSTEL IP doesn t strictly require an Asterisk PBX, but it must receive the IP calls from somewhere: Gateway, SIP Trunking or IP PBX. Hence, if you take the opportunity to totally migrate your station s telephony to IP when installing the system, you can use a simple PBX that we provide fully configured on a Synologic NAS DS112 server, including a solid-state disk, with capacity for more than 30 simultaneous calls. IP Phone SYSTEL IP doesn t need phones, as it incorporates its own specific phone service terminals and can even use a PC microphone / speakers, or even the coordination circuit of an audio mixing console. But if you take the opportunity to migrate your station s telephony to IP, you should use IP phones in reception and all offices.
7 Systel IP Features GENERAL FEATURES Operating Features SIP communications protocol: compatible with VoIP trunkings, Asterisk PBX, SIP Phones such as Phoenix Pocket or Phoenix Lite, N/ACIP compliant Audiocodecs such as Phoenix Mercury, Phoenix Studio, Phoenix Venus or Phoenix Mobile and POTS, ISDN, E1 and T1 FXO. Based on non-blanking digital switching matrix, all lines (4 or 12 depending on the model) can be simultaneously live participating in a program with no loss of quality. GPI/O.- programmable functions: WAIT / ON AIR, CUE, PLAY. 4 GPI, 4 GPO and power supply on each DB15 female connector. All functions are replicated over TCP / IP in the control network. 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. Encoding Algorithms Phone audio in G.711, G.726, G.729, 50Hz - 3KHz High-Definition audio with G.722 algorithm: 50Hz 7KHz. Echo cancellation Independent, digital gain control for all inputs and outputs with an adjustment range of +/- 12 db and muting. Automatic gain control for telephone returns. Configuration and web server control Software Windows 32 and 64 bit operating systems: Windows Vista, Windows 7 and Windows 8. Functionality Assigns audio, handset, IP phone and chat circuits to the different studios, univocally. Renames circuits Defines and manages phone books, allowing the user to share, edit and copy them. Defines PFL signals assigned to each studio. Defines auxiliary and master signals assigned to each studio. Configure the initial audio levels for each line and each study. Configures the format of the client screens, defining the number of lines per program, console operation, and the use of one or two call queues. SIP configuration for communication with an IP PBX, FXO gateway and external (Internet) or internal (LAN or WAN) service providers. Web control client There is at least one compatible web browser for each one of these operating systems: Windows, Android and IOS for ipad Functionality Call establishment: by number dialing, with SIP identifiers, or from call book entries. Optical tally and acoustic RING signal. Caller identification. Accept incoming calls, either manual or automatically. Register new contacts in the call book. Talk by means of the headset or microphone / headphone with the people at the remote line end. Put calls on hold, while the caller can listen to the program. Put calls ON AIR so they can contribute to the program. Send the signal to any of the PFL (pre fader listen) circuits defined in the studio, in order to listen the signal from the remote line end before going to the program. Send the signal to any of the auxiliary circuits defined in the studio, in order to perform recordings on a recording device or establish party lines on lines connected to a same auxiliary circuit. Changes input and return levels for every phone line in the studio. Display the status of all the phone lines and where they are being routed to. Distinguish between producer, operator and presenter roles: label calls, and chat among the different controllers assigned to the program. Systel IP4 : 4-lines IP engine Inputs and outputs XLR type audio connectors SYSTEL IP HS handset RJ45 connector 2 analog balanced inputs. 2 analog balanced outputs 1 input, analog / digital AES- EBU (AES3 or SPDIF) selectable 1 output, analog / digital AES- EBU (AES3 or SPDIF) selectable 1 input / output port for Systel IP handset 1 WAN IP port for 4 VoIP lines 1 LAN IP port for control 1 DB15 connector for 4 opto-coupled GPI and 4 GPO. Power supply, weight and measurements Universal V. 50/60 Hz. 25 VA power supply Silent operation: natural convection cooling. Weight: 3,5 Kg (7,7 lbs). Width: 482 mm ( 19 ) 1U rack height = 44 mm. (1,75 ). Depth: 170 mm. (6,7 ). SYSTEL IP12 12-lines IP engine Inputs and outputs DB15 female audio multi-connectors 4 SYSTEL IP HS handset RJ45 connectors 8 analog balanced inputs. 8 analog balanced outputs 4 digital AES- EBU (AES3 or SPDIF) inputs 4 digital AES- EBU (AES3 or SPDIF) output 1 input / output port for Systel IP handset 1 WAN IP port for 12 VoIP lines 1 LAN IP port for control 3 DB15 connector for 4 opto-coupled GPI and 4 GPO each one. Power supply, weight and measurements Universal V. 50/60 Hz. 50 VA power supply Silent operation: natural convection cooling. Weight: 5 Kg (11 lbs). Width: 482 mm (19 ) 2U rack height = 89 mm. (3,5 ). Depth: 330 mm. (13 ). SYSTEL IP HS remote powered active Handset Includes 48V remote powered preamplifier with output for electret microphone supply. RJ45 connector for Cat5 or better wiring. Output RJ9 connector for your standard micro-telephone or micro-headphone operator s set (electret mic.) Weight and measurements Weight: 0,5 Kg (1,1 lbs). Width; 85 mm (3,33 ), Height: 44 mm. (1,75 ). Depth: 220 mm. ( 8,66 ). SYSTEL IP ST support for SYSTEL IP HS micro-headphone and 10 tablet / ipad Valid for most 10 tablet or ipad. Compatible for Samsung Galaxy, Apple ipad 2 and ipad 3. Minimum tablet size: 22x15 cm (8,66 x 5,9 ). Maximum tablet size: 28x20 cm (11 x 7,9 ). Includes support for SYSTEL IP HS at the left. Adjustable tablet tilt. April Specifications subject to evolutionary changes. Download the latest version or 7
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