Z/IP ONE IP Broadcast Codec Overview
Z/IP ONE is a 1 RU rack-mount IP codec for remote broadcasting. It’s a single-space rack unit perfect for studios, TOCs and remote kits. Z/IP ONE Includes a full range of codecs including AAC-ELD, AAC-HE, AAC-LD, MPEG 4 AAC, MPEG 2 AAC, MPEG Layer 2, G.711, G.722 codecs, plus linear audio and optional aptX® Enhanced coding. Z/IP ONE supports SIP 2.0 protocol and conforms to N/ACIP standards; it also works with VoIP devices and connects to compatible SIP PBXs. A full complement of I/O, including Livewire® AoIP, analog and AES/EBU, is standard.
Z/IP ONE IP Broadcast Codec Features
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Works with wired and wireless IP connections including WiFi, WLAN and UMTS/EVDO networks.
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Exclusive Agile Connection Technology (ACT) automatically senses network conditions and adapts codec performance to provide the best possible audio.
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Largest choice of high-performance codecs: AAC-ELD, AAC-HE, AAC-LD, MPEG Layer-2, MPEG-4 AAC- LC, MPEG-2 AAC-LC, G.711, G.722, and linear PCM. Enhanced aptX coding optional.
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Dual Ethernet ports for separate streaming and control, LAN for local control with Livewire audio and GPIO; separate WAN for secure connection to wide area networks.
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Livewire, analog and AES/EBU I/O standard.
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Easy browser setup via built-in Web server.
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“Push Mode” for one-way network transmission.
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“Multiple Push Mode” for audio distribution to multiple destinations.
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Distributed Z/IP Server directory service, with multiple geolocations, lets you easily connect to other Z/IP ONE devices without the need for an IP address and also provides sophisticated NAT traversal support.
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Transparent, time-aligned RS-232 channel for remote control or metadata, e.g., RDS.
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Time-aligned 8-bit parallel GPIO port for signaling and control.
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Slim 1RU form factor is perfect for studio racks, remote kits or road cases.
Z/IP ONE IP Broadcast Codec In Depth
Z/IP ONE: It's the Zephyr® for IP
These days, you can get broadband Internet just about everywhere, which makes it ideal for live remotes. But public Internet can also be erratic. You could be lucky enough to get a good connection, but it might deteriorate during your broadcast. What to do? Cross your fingers and hope for the best? Or reduce your bit rate, sacrificing audio quality in hopes of making it through your show?
With Z/IP ONE (the “Z/IP” stands for “Zephyr IP”), you don’t have to compromise audio quality for a solid connection. Z/IP ONE helps you get the best possible quality from public IP networks and mobile data services — even from connections behind NATs and firewalls. Telos® collaborated with Fraunhofer (the developers of MP3 and many AAC breakthroughs) to develop a unique coding control algorithm that adapts to changing Internet conditions on the fly, helping you maintain quality and stability.
We call it ACT, short for Agile Connection Technology, and only Telos has it. Using ACT to sense and adapt to the condition of your IP link, Z/IP ONE delivers superb performance on real-world networks. ACT adapts dynamically to minimize the effects of packet loss and jitter. When the bits are flowing smoothly, you’ll benefit from the lowest possible delay and the highest possible fidelity. If congestion starts to occur, Z/IP ONE automatically lowers bit rate and increases buffer length to keep audio flowing at maximum quality. You’ll get reliable audio even when network conditions are unpredictable — and you won’t need to fiddle with settings or codecs to do it.
To make certain your remote broadcast has excellent audio quality even when IP connections are not- so-excellent, Telos engineers employed AAC-ELD (Advanced Audio Coding-Enhanced Low Delay) to produce excellent fidelity at low bitrates, with nearly inaudible loss concealment and very little delay. Standard high-performance codecs are a part of the Z/IP ONE toolkit as well, such as AAC-HE, AAC-LD, MPEG4 AAC-LC, MPEG2 AAC-LC, G.711, G.722 and even linear PCM. And if apt-X is part of your codec cache, you can add it to your Z/IP ONE as a small extra-cost option.
It’s from Telos, so of course you expect that Z/IP ONE will be easy to set up and easy to use. And it is — the front panel controls are intuitive and friendly, and the built-in Web server makes short work of configuration or remote control via any PC with a Web browser. And our exclusive worldwide Z/IP Server service, free to Z/IP owners, lets you easily get around NATs and network firewalls for fast connections to your favorite locations. For even more flexibility, Z/IP ONE can connect to third-party apps such as LUCI LIVE and LUCI LIVE Lite to receive on-the-go reports from smartphones and tablets.
Around back, you’ll find analog and AES3 XLR ins and outs, a Livewire LAN port for quick connection to Axia® networks, and a separate WAN port for safe connection to “the outside world.”
Z/IP ONE is also wireless-capable and connects natively to IP networks via Wi-Fi. A parallel port is provided for end-to-end, time-aligned GPIO contact closures; Z/IP ONE can also transport RS-232 serial data (using an inexpensive USB-to-Serial adaptor cable), synchronized with audio delivery — useful for RDS/RBDS data, as well as other serial data, at up to 9600 bps.
Z/IP ONE IP Broadcast Codec Specifications
Conformance and Compatibility:
- Conforms to N/ACIP (Open) Standards. Fully supports Session Initiation Protocol 2.0 (SIP) Compatible with TCP, UDP, DNS, Zephyr Xtream®, Uncompressed PCM, and other Internet Protocols.
Codecs:
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SIP: G.711, G.722, MPEGLayer2, MPEGAAC, MPEG4AACLC, MPEG2AACLC, Linear PCM, MPEG AAC-Enhanced Low Delay (ELD), HighEfficiencyAAC.
Connections:
Analog
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1x Stereo input, presented on two XLR-F connections
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1x Stereo output, presented on two XLR-M connections
Livewire
- 1x 100BASE-T connections, presented on RJ-45
AES/EBU
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1x Stereo Input, presented on one XLR-F connection
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1x Stereo Output, presented on one XLR-M connection
Network
- 2x 100BASE-T connections, presented on RJ45 (1x LAN, 1x WAN)
USB
- 2x A-Type, Female
Parallel (GPIO)
- 1x DB25, Male
Audio:
Analog Line Inputs:
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Input Impedance: 6K Ohm differential
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Input Range: Selectable, Line (+4 dBu nominal), Microphone (-50dBu nominal)
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Selectable Phantom power
Analog Line Outputs:
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Output Impedance: 50 Ohm differential
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Output Clipping: +22dBu
Digital Audio Inputs And Outputs
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Reference Level: +4 dBu (-20 dB FSD)
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Impedance: 110 Ohm, balanced
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Signal Format: AES3 (AES/EBU)
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AES3 Input Compliance: 24-bit with sample rate conversion
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AES3 Output Compliance: 24-bit
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Digital Reference: Internal (network timebase) or external reference 48 kHz, +/- 2 ppm
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Internal Sampling Rate: 48 kHz
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Input Sample Rate: 32 kHz to 192 kHz
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Output Sample Rate: 48, 44.1 or 32 kHz, or “sync to input” (auto-matches rate and clock from
AES/EBU input)
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A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling
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D/A Conversions: 24-bit, Delta-Sigma, 256x oversampling
Frequency Response
- Any input to any output: +/- 1dB 25– 20 kHz
Headroom
- 18 dB
Dynamic Range
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87dB Unweighted
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90 dB “A” Weighted
Total Harmonic Distortion + Noise
- < 0.03% @ +12dBu, 1 kHz Sine
Crosstalk Isolation
- > 80 dB
Power Supply AC Input
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Auto-ranging supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
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Power consumption: 14.2 Watts
Operating Temperatures
- 0-40 degrees C (32-104 degrees F), stirred air
Dimensions
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19” (48.3 cm) standard rack mounting front panel
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1.75” (4.5 cm) height, 6.5” (16.51 cm) depth
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Shipping Weight: 8 lbs. (3.62 kg)
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Shipping Dimensions: 24” x 14” x 6” (61 cm x 35.6 cm x 15.25 cm)
Regulatory
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Click here to view the current regulatory compliance.
It’s possible that your Z/IP ONE is being rebooted too quickly after making settings changes. There is an intentional delay after a setting is changed, before it is written to the unit’s flash memory. This is to prevent file system corruption should power be removed before the write is complete. For this reason, settings changes are not committed to the file system until ten seconds after the last change.
Just like you would never save a file and then pull the power cord on your PC, there is a procedure to reboot a Z/IP ONE safely. The safe way to reboot a Z/IP ONE is to go to the System->Software menu, and choose the option to reboot to the bank that the Z/IP ONE is currently booting from. This will ensure that all writes are done properly, and set the system to reboot as soon as it is safe to do so.
Yes, you can use a point to multi-point connection from one ZIP ONE to another ZIP ONE using the RTP "push mode" connection. Point to multipoint connections can not be used with the end-to-end contact closure feature. End-to-end contact closures are only available using the TSCP connection mode.
Also, be aware that, because push mode is a one-way connection only, Z/IP ONE can not utilize the ACT error correction/concealment technology. ACT is also unavailable when connecting between two Z/IP ONE using connected via SIP. ACT is only available during bi-directional point-to-point connections utilizing the ZIP Server or direct-dial TSCP mode.
Partially. You will be able to establish a connection, and the Z/IP ONE will decode audio from the Zephyr Xstream, but the Xstream will not decode audio from the Z/IP. If you still wish to do this, you’ll need to place your Zephyr Xstream in the “SIP” interface mode (see the Codec menu for this setting). Set your Xstream’s encode/decode mode to AAC, preferably 128 kbps mono or stereo. On the Z/IP ONE side, set it for MP2-AAC coding. Note that this procedure will only work if your Z/IP ONE is NOT registered with a SIP server, since the Zephyr Xstream is not SIP capable and will not receive calls routed through SIP servers.
Yes, The Telos Z/IP ONE is N/ACIP compliant.
N/ACIP is a technical project group from the EBU. The N/ means it is a project group from the Network division managed by the Network Management Comittee (NMC) of the EBU ACIP stands for Audio Contribution over IP. The EBU has established a project group, N/ACIP, to work in close cooperation with manufacturers to develop an interoperability standard for equipment for audio contribution over IP. The group will also create EBU recommendations on operational practices allowing its members and others to share experience and knowledge and help each other to get the best out of audio contribution links established over IP connections.
The answer to this question is "yes and no." Allow us to explain.
Telos is a member company of the N/ACIP workgroup. So are our friends, Comrex. As a result, there is some compatibility between the Telos ZIP ONE and the Comrex BRIC, such as when using g.722 mode. However, because the quality and reliability of public IP connections can be wildly inconsistant, both Telos and Comrex have developed their own sophisticated, proprietary techniques to improve the quality and reliability of coded audio over inconsistent links. For instance, Z/IP ONE uses our exclusive ACT Agile Connection Technology to dynamically adjust the buffers, bitrates, codec algorithms, and perform multiple levels of error correction and concealment. So, if you connect two Z/IP ONEs together, you'll get this "best effort/tech" using a combination of standard and proprietary tech - and Comrex devices will likely do the same. But these techniques are NOT part of a N/ACIP compatible mode. In other words, two N/ACIP compatible devices from different manufacturers will work together using their most basic settings, but best performance over the public Internet is going to come from having the same device at each end of the connection.
You can use a direct "TSCP" call which maintains all of the ACT advantages. This can be useful for applications such as backup or temporary STL service. You may even consider asking your ISP to provide QOS service for this. You will need your ISP to provide outside IP addresses for each of the 2 units in any case.
Configure each of the ZIP ONE's WAN port network parameters with a normal internal IP address and the proper Gateway address (usually the address of the Router). To call the other unit, Press the "CONN" button and set the "Device Name" to the outside IP address of the other unit. Delete any "Group Name" that is present so that it indicates "(For TSCP Calls)" and set the "Call Type" to "TSCP". You should now be able to connect to the far-end unit. Be sure to set the Codec parameters as appropriate for the available bandwidth. For STL use, "regular" AAC is suggested at 256kbps or higher (bandwidth permitting). It would also probably be a good idea to set General Settings / "Autoredial Broken Connections" to "Forever".
Yes. Analog inputs and outputs are still available with configurable gain, and even an available microphone boost setting.
No. The change to support AES/EBU requires both a new main board and new chassis.
Yes. Since the Z/IP ONE encodes just one stereo pair, you must choose which of the input options to use for encoding. However, decoded audio is simultaneously available on Analog, AES/EBU, and Livewire outputs. You can even set the inputs to ‘fail over’ from Livewire to AES/EBU to Analog in case of input failures.
Loss of frame synchronization is used to determine whether to change input sources from Livewire or AES/EBU. Silence is not considered.
The AES/EBU design has a new main board with a new digital audio controller. Aside from minor changes necessary to configure AES/EBU, the new Z/IP ONE design has the following major differences:
- Different analog input gain settings, particularly with respect to microphone gain
- Two additional XLRs on the back panel
- Ground stud near the power entry module on the back panel
