ZEPHYR XPORT USER’S GUIDE

Phone

Places a call to a standard telephone. DSP hybrid technology prevents send audio coloration due to leakage, however the audio quality will still be only 300 to 3.3 kHz telephone quality. If the ISDN interface and line are present at the time of bootup, the call will go over ISDN. Otherwise, the call will be placed over the POTS interface.

h HOT TIP!

You can use this mode to call the on- air telephone system of any radio station, or to a telephone.

Zephyr

NOTE: This option will only be present if the optional ISDN interface is present and your unit is connected to ISDN.

Places a data call using the ISDN interface to a Zephyr Xstream at the far end. This mode is only available if the ISDN interface is present. This mode offers very high quality with very low delay, and is recommended whenever ISDN is available at the remote site where the Xport will be used.

The Zephyr mode uses AAC- LD - the far end Xstream should be set to:

Xmt = AAC- LD Mono 64, RCV = AAC- LD 64, Sample = 48 kHz.

DROP

The <DROP> key is also straightforward. Pressing <DROP> twice will drop any call active.

2.3.3 The rear panel

The rear panel connections are shown below. For additional information, see Section 3 (The Details).

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Telos Zephyr Xport manual Rear panel, Phone

Zephyr Xport specifications

The Telos Zephyr Xport is a cutting-edge audio-over-IP codec that seamlessly revolutionizes the way audio is transmitted over networks. Renowned for its high-quality sound, this powerful codec is designed for both radio broadcast applications and professional audio environments, enabling efficient, reliable, and high-fidelity audio transport over IP networks.

One of the main features of the Zephyr Xport is its ability to handle multiple audio formats, including PCM, AAC, and MP3, which provides users with flexibility depending on their specific needs. Its ability to encode and decode audio ensures that broadcasters can deliver outstanding quality both on the air and in streaming applications.

The technology behind the Zephyr Xport includes advanced algorithms that minimize latency, making it ideal for live broadcasts where real-time transmission is crucial. With a low latency rate, users can enjoy audio transmission that is nearly instantaneous, a defining feature for any live event or remote broadcasting situation.

Another standout characteristic of the Zephyr Xport is its support for multiple network protocols, including RTP/RTCP, SIP, and POTS. This versatility allows integration with a wide range of existing equipment and infrastructures, facilitating easy implementation in various settings. Additionally, the codec boasts robust error correction and adaptive bitrate control to ensure audio clarity even in fluctuating network conditions, reducing the chances of dropouts or interruptions.

The user-friendly interface of the Zephyr Xport features an intuitive LCD display, making it easy for operators to configure settings, monitor audio levels, and manage network connections. With built-in web access, users can make adjustments and control the unit remotely, providing added convenience for situations where operators are away from the hardware.

Moreover, the Zephyr Xport is designed for durability and reliability, constructed to endure the demands of a fast-paced broadcast environment. Its compact design makes it suitable for rack mounting or portable use, allowing broadcasters to take it on location without hassle.

In summary, the Telos Zephyr Xport combines advanced audio processing technologies with user-friendly features, making it an essential tool for modern broadcasters. Its high-quality audio performance, low latency communication, versatile protocol support, and ease of use exemplify the future of audio transmission in the ever-evolving landscape of digital broadcasting. As the industry continues to embrace IP-based solutions, the Zephyr Xport stands out as a vital asset for any professional audio application.