Codecs Frequently Asked Technical Questions
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- My Zephyr either is stuck in Ethernet mode, or when I insert the ISDN line cord into the "U" connector, nothing happens, and the green LED next to the connector stays dark. What do I do?
- Most likely, something has damaged the ISDN card line input circuitry. Usual causes for this are close lightning strikes and mains voltage surges. Please contact Telos Support to schedule a repair.
- My Zephyr's screen shows INACTIVE INACTIVE. What do I do?
- If the Zephyr ISDN Transceiver shows INACTIVE, then its not connected to a working ISDN line. Check the wiring from where it enters the building, to where it connects to the Zephyr. If the wiring is good, then the problem is not in your plant! The next thing to do is to call your ISDN provider and request a trouble ticket. Be sure to have the ISDN line numbers and a description of the problem ready to report to the ISDN line supplier.
- My Zephyr appears to connect to the Zephyr at the other end, but the codec doesn't lock. Whats happening? How do I fix this?
Start by checking both Zephyrs in ISDN loopback mode. If you can lock to yourself in loopback mode, the great likelihood is that the difficulty is external to the Zephyr, and probably in the ISDN lines. Some customers have been experiencing inability to lock at 64 kHz bitrates, but are able to lock at 56 kHz. Within the past year or two, some telephone providers have discontinued support for 64 kHz service, so try 56 kHz instead and see if you achieve lock. You can also try connecting, and then selecting the G.722 codec. G.722 is a lower speed codec and occasionally locks when the higher speed codecs will not.
There could also be a problem with the long distance carrier handling data calls. Remember, all Zephyr codec calls are data mode calls and the ISDN provider needs to provision the ISDN lines for data transmission. So, if you can lock the Zephyr to itself in loopback mode, but not over the ISDN connection, then the next call should be to the ISDN local carrier and/or long distance carrier. The instructions for ordering the proper provisioning of ISDN lines for Zephyrs can be found toward the back of the user manual.
- Do I need to be careful what I plug into my Zephyr's audio jacks?
- Yes, be sure not to plug any input or output from any Zephyr into a microphone or intercom circuit connected to a live DC phantom power source. The input or output circuitry of the Zephyr could be damaged.
- In what format should the SPID numbers be entered into my Zephyr/Xstream?
- The most common SPID format for ISDN lines using the Nat'lI-1 protocol is the area code followed but the 7-digit phone number followed by a 4-digit SPID suffix. For example: 21678193100101 where "216" is the area code, "7819310" is the phone number and "0101" is the SPID suffix. Sometimes the telco will only provide 2 digits for the SPID suffix such as "01". In this case go ahead and add the second "01" to make it "0101". If you still are having problems getting the Zephyr to Ready / Ready please contact Telos Customer Support.
- Do I need to enter the MSN/DN numbers below the SPID numbers in my Zephyr Xstream?
- Yes. The (silver) Zephyr Xstream uses a different ISDN stack then the original (black) Zephyr. The Zephyr Xstream requires both the SPID numbers and the DN (directory numbers) to be entered. These directory numbers are normally just the 7-digit phone numbers that can be extracted from the SPID numbers. For example the DN for this SPID "21678193100101" would be "7819310". Please note that the Zephyr Xstream will need to be rebooted whenever changes to the SPID or DN numbers have been made.
- What does it mean when my little green LED on the ISDN NT-1 card is blinking?
- That means either no ISDN line connected, or the line is not functioning properly. With no line connected, the LED should flash evenly about 5 times per second. When a good ISDN line plugged in to the "U"port, the LED should revert to a steady "on" condition.
- What does "SPID Pending" mean?
- SPID Pending appears on the TEL menu as the Zephyr attempts to connect to the ISDN line, and is searching for the line's SPID number to match the SPID number entered into the TEL menu.
- What does "Cause xx" mean when I attempt to make an ISDN connection, and fail?
- There are many reasons an ISDN call may not connect properly. Often, the telephone company sends back an error code which the Zephyr interprets as a "Cause". A complete listing of the Cause codes is contained in the Zephyr manual, and also on line at the Telos website.
- The green LED on the NT-1 card is dark (not lit), Why?
- Something has damaged the ISDN card line input circuitry. Usual causes for this are close lightning strikes and mains voltage surges.
- I can call locally, but I can't make a connection with someone in another city.
- Be sure you have selected and assigned a Long Distance ISDN carrier, and informed both the local provider as well as the long distance provider. Also, try dialing 1010222 + 1 + area code + number this will force it through MCI, there is a good chance this will get your call to go through.
- Why does the sending end of the telephone circuit hear digital "breakup", while the receiving signal is clean or vice-versa)?
- After the two wire connection to the Zephyr through the "U" connector, from the demarc of the telephone company , the ISDN is a "4 wire" system. Its possible that one "direction" of the circuit is experiencing errors, while the other is pristine.
- I can make and receive digital calls fine, but my Zephyr will not function on POTS calls.
- The ISDN line needs to have both CSD (Circuit Switched Data) and CSV(Circuit Switched Voice) enabled.
- Why does my Classic Zephyr not retain my SPID's and Auto-Dial information when I turn it off?
- Its likely the internal battery has run down. Average internal battery life is 5 to 7 years. Contact us if you need to purchase one, we sell them for $20.
- What sort of ISDN line should I order for use with my Zephyr Xstream?
- To keep things easy, all Telos products that use ISDN use the same ISDN provisioning. In simple terms, what’s needed is a “dumb” ISDN BRI line, 2B+D, with most of the “automatic” services turned off. We recommend that you check your Telos product’s user manual for further details – there is a section (usually toward the back of the manual) that specifies what kind of provisioning needs to be ordered for an ISDN line. That section will instruct you on exactly how to request provisioning from your line provider. You can even scan and send these manual pages to your telephone company!
- I cannot hear any send audio from the front panel headphone jack on my studio’s Zephyr Xstream. What’s up?
- On the rack-mount Zephyr Xstream, the front panel headphone jack provides monitoring for incoming (receive) audio only. Only the Zephyr Xstream MXP, with the built-in mixer, allows switchable and adjustable local and remote monitoring simultaneously.
- I entered the SPIDs on my Zephyr Xstream, as provided by the telephone company, but can not receive incoming calls on line 2.
- Most likely, if the line is OK, there is no number entered into the MSN/DN field in the TEL menu. Rectify this and you should be up and running.
- I have my Zephyr Xstream’s receive codec set to Auto (Search). Why does it take almost a minute for my incoming audio to lock?
- The Zephyr Xstream is "searching" for a match to the incoming data stream. It works from a look-up table, and may take 30 or more seconds to find the “match” and sync up. Telos recommends that the Auto (Search) function only be used when the incoming stream's encoding algorithm is unknown.
- Can the Zephyr Xstream be remote controlled?
- Yes. All the menu functions of the Xstream can be accessed with an ethernet connection between a PC and the Xstream, using Internet Explorer, or another compatible browser.
- Is the Zephyr Xstream able to be used outside of the USA?
- Yes. The power input accepts mains voltages between 100 and 264 volts at 50 to 60 Hz. The ISDN "U" input is used in North America, and the "S" input is used in most other parts of the world. The ethernet connection is standard world wide.
- When I tested my Zephyr/IP yesterday, it connected quickly and reliably. Today, it’s much harder to connect. What’s going on?
- If real world networks had unlimited speed and bandwidth, there would always be plenty of "room" for all traffic. Real world networks are dynamic, with varying throughput speeds, varying rates of data transmission, and so on. Your internet connection may be going through dozens of servers, and each one has bandwidth and rate limits. Just like on a multi lane freeway, when traffic is light, you go the speed limit and have no blocked lanes. When traffic is heavy, speed falls and congestion builds. When traffic is extremely heavy, just as in automobiles, you can start and stop, and be restricted to slow lanes.
- Can I "guarantee" a fast and reliable Z/IP connection?
- Yes, if you have a QOS (Quality Of Service) guarantee on the network. Many ISPs have "big pipes" for their data traffic, but the Internet is extremely variable and, by its very nature, cannot guarantee QOS. The ZIP uses some very sophisticated algorithms to minimize the effects of "variable" internet connections, but, like the freeway, sometimes the blockage is pretty bad – and there’s not much you, as a user, can do about it.
- I tried VoIP a few years ago. The quality was okay for talking, but not good enough for broadcast transmission.
- To overcome any complications with using public IP for broadcast transmissions, Telos Systems engineered Agile Connection Technology™, or ACT. This exclusive technology from Telos Systems is designed to optimize public-IP broadcasting. Utilizing ACT, the Zephyr/IP dynamically overcomes packet loss, jitter, limited bandwidth, Firewalls, NATs, as well as dynamic IP address changes. With overcoming these factors, delay is minimized and the Z/IP intelligently adjusts to available bandwidth to provide excellent audio quality. This is done automatically without you constantly adjusting settings. The Zephyr/IP also uses MPEG AAC-ELD, a new codec designed specifically to address these issues.
- This sounds great, but complicated.
- Configuration is simple and allows you to assign the unit a unique name and group membership for easy identification. The unit's name and group allow it to be identified regardless of IP addresses and you control the units visibility to other Z/IPs on the network. You can configure the Z/IP through the full color VGA display or through a web interface. Wireless connections are configured in a simple four step process.
- Why use a VGA display? A web interface is good enough.
- There are situations where a web-interface is not convenient and can actually be limiting. In addition to a built-in web server, accessible by any web browser (even remotely), we give you a full-color VGA screen on the unit itself. This allows you to focus more on the important aspects of a broadcast.
- What makes this device unique?
- This is the only device on the market that continuously adapts itself to network conditions. During a five hour remote, you won't have to worry about constantly having to adjust your settings to accommodate network changes or make short calls.
Beyond basic stability, the Z/IP is extremely user friendly to you and other VoIP devices. We use open standards for signaling (SIP), transport (RTP) and codecs (G.711, G.722, MPEG codecs, linear PCM). This allows the Z/IP unsurpassed compatibility with other devices. The Z/IP can register and accept calls from your VoIP PBX as long as it supports SIP 2.0 and the G.711 and G.722 codecs.
- Digital is great, but I still have some analog gear. Will the Z/IP work with it?
- The Z/IP accepts both digital and analog inputs. Analog is input using standard XLR connectors and accepts both pro and consumer input levels with a peak of +22dba.
- I hear a lot of talk about compatibility and open standards.
- Compatibility and the use of open standards is an important factor to consider when investing in your broadcast technology. The longevity of true compatibility can only be achieved by using a set of mutually agreed upon open standards that participants implement. The Zephyr/IP is fully compliant with the N/ACIP International Standard. The Z/IP is compatible with any device that uses these standards. By using Session Initial Protocol (SIP) the Z/IP works with TCP, UDP, DNS and other protocols. The Zephyr/IP also uses the Real-time Transport Protocol (RTP) to transmit audio data. RTP is used by VoIP telephony and is becoming standard for broadcast codecs. In addition to standard signaling and transport protocols, the Zephyr/IP contains a suite of codecs that adhere to the N/ACIP standard. The Zephyr/IP may even be able to talk to your existing VoIP PBX.
- What is the Telos Z/IP Server? Do I need one of my own?
- The Telos Z/IP Server is a free service, provided and maintained by Telos Systems. It provides enhanced features for the Z/IP. It allows name or phonetic (soundex) and wild card searches, directory services, be visible and reachable behind NATs and firewalls, exchange status notifications and determine location of other end points. We provide this service, including full service redundancy, so you don't have to worry about it yourself.
If you don't want to use the Z/IP Server, the Z/IP will still work on any network including closed intranets, satellite links or other networks that are not connected to the Internet. When you are able to connect to the internet, the Z/IP Server will give you an enhanced experience.
- What about software updates?
- The Z/IP is fully updateable, for free, through the unit's web interface. They're downloadable at our website, and all of your settings are retained.
- What is error concealment?
- Error concealment is an inaudible technique of replacing lost audio packets without requiring retransmission and thus minimizes delay. The Codec keeps track of the audio frequencies used around the lost packet and creates a similar replacement. This is the best technique for handling packet loss in an interactive, two-way application. The algorithms employed on the Z/IP yield usable results with up to 20 percent random packet loss.
- Why don't you use Forward Error Correction (FEC) instead of concealment?
- Forward error correction works by sending the original audio packet along with some form of copy of this packet at a later time. If a packet gets lost it is replaced by the copy. This has two downsides: it increases the bandwidth usage (with lack of bandwidth being the likely reason why the packet was lost) and it increases delay. Using error concealment avoids both of these issues.
- Why don't you use TCP to send the audio streams?
- While TCP ensures that all of the data will get to the other side, it does this by using very large buffers and potentially very large delays. In most cases the delay is at an unacceptable level for two-way connections. We use RTP (transported over UDP) to get the minimum delay possible, suitable for two-way conversations.
- Can I send a stream to multiple locations?
- While the Z/IP generally supports one-to-one connections it is also capable of sending streams to multiple locations. We call this "push" mode in the manual.
- How can I tell what the network conditions are during a call?
- The Z/IP displays the approximate path taken by the audio on a world map along with the average delay introduced by each node. The path and node delay may change, but this information gives valuable clues about the network congestion. Additional screens display real-time statistics such as the number of packets lost, buffered, and concealed in both numeric and graphical formats.
- How do I configure the Z/IP to use a wireless device?
- Power up the Zephyr/IP without the device plugged in.
- Once fully booted up, and the Main Menu is displayed, plug in the approved wireless device into either of the USB ports on the back.
- Navigate to the main Network Menu and select Wi-Fi / WLAN. Enter the hostname for the wireless network (ESSID Network Name and Encryption Key as well for Wi-Fi). Press "Activate Settings" to commit your changes.
- ESC back to the main Network Menu and choose "Wi-Fi / WLAN" for a Wi-Fi device, or choose "UMTS" for a cellular device.
You will now be using the mobile device to make a call.
- What wireless devices does the Z/IP support?
- The Zephyr/IP uses common (non-proprietary) wireless devices. The Z/IP currently supports the following devices. Check back regularly since the list will be expanding:
- Linksys Compact Wireless - G USB Adapter, 24Ghz
EVDO / Mobile Broadband
- Sierra Wireless USB device AC595U for Sprint or Verizon
- Novatel Ovation U727
- How does the Zephyr/IP perform in real-time?
- A real-time, delay-free transmission requires a high-quality, stable, and dedicated connection such as a closed Intranet, satellite link or other (non-public) connection. The Zephyr/IP excels in performance over both controlled and non-controlled connections, such as public-IP. Using advanced codecs and Agile Connection Technology, the Zephyr/IP produces high-quality transmissions, with very little delay, without needing you to constantly 'tweak' the settings.
- Why audio over IP? Why not stick with ISDN or GSM?
- With the growth of the Internet, service providers are slowly discontinuing ISDN. This is also the case with GSM. In addition to IP's widespread availability, IP connections do not have multi-week lead times, costly set-up and per-minute charges. The Zephyr/IP uses exclusive technology to overcome the complications with public-IP audio transmission and is optimized for real-world conditions.
- I thought that your products only used Ethernet for transmission within a facility.
- The Zephyr/IP has been designed for exceptional transmission within a facility as well as over public IP and wireless mobile connections, such as at remotes. Not only does the unit provide superior functionality over public IP, the free Z/IP Server further enhances IP audio transmission by removing the complications. In addition to Ethernet connections, we include parallel, AES/EBU, USB, RS-232, analog, and even an extra PCI card slot.
- What about remotes?
- Among other features, the Z/IP Server provides dial-by-name functionality and traverses tough firewalls and NATs. If you have a long broadcast planned, you won't need to make multiple short calls. The Z/IP Server works hard to keep you connected and handles dynamic IP address changes. Dial-by-name functionality eliminates the need for reprogramming your unit if your area code is changed. The Z/IP Server can act as a media relay if both Z/IP units are behind especially troublesome firewalls or NATs. The new Z/IP Mixer combines the versatility of the Zephyr/IP with the utility of a digital four-channel stereo mixer, all in a rugged, road-ready portable chassis.
- Do I need to buy an upgrade for these codecs or functionality?
- The Zephyr/IP has the complete suite of codecs and dynamic functions without needing any upgrades or options purchased.
- I thought that quality over a mobile connection isn't that good.
- The Zephyr/IP uses Agile Connection Technology (ACT) to provide you with excellent quality audio over mobile connections. The Zephyr/IP is designed to work with standard existing mobile phone connections and optimize available bandwidth to provide excellent quality audio with very little delay.
- What about international connections?
- Since the Z/IP uses internationally adopted open standards, there is no need to worry about international connections or buying a special international model. Dial-by-name, directory look-up, and "group speed-dial" features make international calls as easy as calling a unit in the same room.
- Can the Z/IP place regular phone calls?
- The Zephyr/IP can place regular voice calls with very little configuration needed. If they Zephyr/IP has already been configured to use a wireless connection, all you need to do is change the streaming device in the main Network Menu. The Z/IP uses an intuitive display that presents context-sensitive help.
- I thought that packet based audio transmission through the Internet was bad.
- Agile Connection Technology solves the problems with packet based transmission to where it is the favorable protocol to use. ACT uses dynamic monitoring to adjust itself for optimum performance given the current network conditions. This includes dynamic buffering, jitter correction, and packet-loss concealment, just to name a few.
- What will happen if one of the public-IP transmission relay points, or hops, goes down?
- Every packet of your audio transmission can take completely unique routes. If a relay point goes down, the other packets will simply not take that route. The Zephyr/IP handles the unpredictable arrival of packets (jitter) by dynamically adjusting itself dynamically while optimizing quality and minimizing delay.
- I'm getting dropouts in one or both directions on my Internet ZIP connection. The Internet connection at both ends is good and I am using a reasonable bit rate and connecting through the ZIP Server. The dropouts are quick when they occur and can be quite regular in their timing. What can I do?
- Check your Min Buffer and Max Buffer settings. The 0 Min setting can only be used over a guaranteed QOS connection such as a direct SIP connection over a WAN or LAN. For Internet connections, we suggest a Min setting of at least 100ms. It may also be helpful to limit the entire range by changing the Max setting to around 3-5 seconds; this gives the ACT algorithm a better-defined working range.
- How do I connect two Z/IP ONE's together directly, without going through the ZIP Server?
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".
- Can I use my Z/IP ONE for point to multi-point streaming? If so, Is the parallel port end-to-end function available in that mode?
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.
- I need to configure my firewall for operation with Z/IP ONE . What ports do I need to open?
- You will need to make available ports 24 and 308 (used for updating), port 11926 (used for listening), and ports 5060 and 5061 (used for SIP negotiation). Also, default port for TCP is 5060, and 9150 for UDP.
- I've heard a little about the N/ACIP interoperability project. Can you tell me more about this?
- 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.
- Is Z/IP ONE compliant with N/ACIP?
- Yes, The Telos Z/IP ONE is N/ACIP compliant.
- If the Telos Z/IP ONE is N/ACIP compliant, that means I can use it with a codec from a different manufacturer?
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.
- Can I send and receive audio between my Z/IP ONE and my Zephyr Xstream using the SIP method?
- 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.
- Some of my settings are not being saved after I reboot my Z/IP ONE. What is going on?
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.