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.
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.
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".