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RTS Telex
http://www.rtsintercoms.com
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Enhanced communication via modern Session Initiation Protocol telephony

September 20, 2016
  • SIP server enables VoIP phone calls interfacing thanks to state-of-the-art Session Initiation Protocol (SIP) technology
  • Future-proof solution to connect intercoms to digital phone systems; old analog phone lines no longer required
  • SIP phone line provides backup solution for communication, operating independently of the trunking system
Amsterdam/Straubing, September 2016 – With the new SIP server functionality, RTS provides a cost-effective way for digital RTS ADAM or ADAM-M matrices to make phone calls via IP telephony. Interfacing with state-of-the-art Session Initiation Protocol, this allows users to feed into a telephone switching system (PBX) and ultimately into the public switched telephone network (PSTN). Unlike telephone interfaces, the SIP server can provide an unlimited number of digital telephone lines; each SIP line is capable of supporting one phone conversation.

The SIP server is a functionality based on RTS VLink software, a fully inter-connected software intercom solution for RTS matrix systems. RTS VLink enables remote users to interface with RTS matrix intercoms, allowing control and flexibility from anywhere in the world. The RTS VLink system feeds intelligent trunking links into an RTS intercom matrix to provide full support for RTS intercom alphas and matrix access for standard communications workflows.

A simple ticking of the checkbox in the RTS VLink configuration software activates the SIP server functionality. A complete system requires an RTS matrix with OMNEO IP technology on board, a PC with the RTS VLink software and Dante Virtual Soundcard as well as SIP-lines, an IP network and the respective software components and licenses. The SIP-server can also be connected via an analog connection. In this case, the RTS matrix’s analog signals must be converted into IP Dante signals.

Wherever RTS VLink software uses the SIP server functionality, an RTS TrunkMaster system is no longer required – thus reducing the required investment. For broadcasters already using TrunkMaster, the SIP server can be utilized as a professional backup communication system, adding more resilience and redundancy to the existing intercom system.

The SIP server can transmit several audio streams to the matrix. OMNEO is the preferred IP technology; it is available for digital ADAM and ADAM-M matrices. Users can also opt for an analog connection, which is available for all RTS matrices, or MADI (Multichannel Audio Digital Interface).

The AZedit configuration software from RTS has been updated with the SIP server functionality; an optional Tally Screen for monitoring each connection is also available for the SIP server. The latter creates a real-time display of the SIP line usage, with levels on both incoming and outgoing audio.

The SIP server from RTS will be available worldwide at the beginning of 2017. For further information, please visit www.rtsintercoms.com

About Bosch OMNEO IP technology: By utilizing both industry and open public standards, OMNEO provides future-proof technology with the highest level of interoperability, flexibility, reliability and resilience. OMNEO is based on two key technologies – the media transmission component Dante (from Audinate) and the system-control component OCA (Open Control Architecture). OMNEO enables a secure set up at competitively low system cost due to the use of standard IT components, along with lower installation and maintenance costs.


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