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| Section 9
Direct Connection
Instead of using the ZIP Server, you can make direct TSCP calls as well. To do so, you will dial the IP address and
listen port of the remote Zephyr/IP in the standard address:port notation. Since this is a serverless connection, you
will not have the benefits of the directory server or NAT traversal assistance.
9.2 N/ACIP Session Initiation Protocol
The European Broadcasting Union, recognizing that broadcasters increasingly rely on IP networks to transport
audio, defined a standard for connecting between codecs from various manufacturers. This standard is N/ACIP
(for “Network / Audio Contribution over IP”). Put simply, N/ACIP defines a connection method, an audio
transport method, and a list of codecs that are mandatory, recommended, and optional.
The features of this protocol are:
♦
Directory service (optional)
♦
Cross-product compatibility
♦
Bidirectional audio transfer
♦
Agile Connection Technology (on applicable codecs)
Of these things, the part that needs configuration in the Z/IP ONE is the connection method. For this, N/ACIP uses
Session Initiation Protocol, or SIP. SIP is a signaling protocol widely used for Voice Over IP (VoIP) applications.
Using SIP, the Z/IP ONE is able to accept and make calls to a variety of devices, including some VoIP phones.
SIP can be used in two modes: direct connection, or with a SIP server. Direct mode is what you would expect: one
device connects directly to another. A SIP server entry is used when you have established VoIP service, and all VoIP
calls are directed by that provider. Another case when a SIP server may be used in when your facility has a VoIP
PBX, so the calls may be routed through the PBX. A SIP server acts a bit like the ZIP Server, passing information to
its clients so that they may make connections to each other.
SIP server details are entered on the front panel under Setup->Network->SIP Server, on the Network web page. If
you use a SIP server, enter the hostname or IP address of the SIP server as the “Registrar Hostname,” and then your
login details under “Registrar User Name” and “Registrar Password.”
The format of the “registrar user name” is unfortunately dependent on the SIP provider. Your login details will be
covered in your SIP server documentation, but there are three common schemes:
♦
An extension number. This is common with a PBX, especially those used with VoIP desk phones.
♦
A raw user name. This is less likely with a PBX, but it is still possible that a user name is associated with an
extension number, and used as the login to the server. In this case, the ‘registrar user name’ field would be
something like JohnSmith
♦
username@servername. This is most common with Internet SIP servers, but it is possible that you
could see it in a PBX as well. This format is used when a single SIP server may have multiple identities. It
uses the ‘@servername’ portion so that each of its identities can have usernames that would otherwise
conflict. You can think of it as the ‘group’ portion of a TSCP identity.
If you have a SIP server defined, all SIP calls will be made using it. The formats to call other users will also be defined
by the server. SIP to SIP calls are made like sending an email – [email protected]. However, a PBX may allow you to put
in extension numbers, or dial out to the public switched telephone network by entering a phone number in the SIP
connection field. Again, consult your SIP server, PBX, or IT department for information on using your SIP server.
