ViA User Manual v2.0

Important Notes:

•You cannot edit a program when it is currently loaded in the codec.

•You can lock a loaded custom program in a codec to ensure the currently loaded program cannot be unloaded by a codec dialing in with a different type of program.

•Some drop-down menus and settings may be greyed out intentionally depending on features available and the transport selected (e.g. IP or ISDN).

•It is possible to save audio stream or program settings at several points throughout the program wizard and use default settings to save configuration time.

•Failover and SmartStream PLUS redundant streaming are not available with SIP or sessionless IP connections.

Important Note: When you decide to use an existing program as a template, the new program inherits all the settings of the template program and you can adjust these settings as required by continuing through the program wizard.

Routing Type Options:

Default

No Dial Route or Answer Route is configured. An incoming call will be routed to an audio stream on a first-come, first-served basis in a multi-stream program. Note: By default IP streams are routed using audio ports.

Deterministic

Select a Dial Route or Answer Route to configure deterministic routing of multiple audio streams using transports like ISDN or POTS. Use of Dial and Answer Routes is not necessary over IP. See Configuring ISDN Answering or Configuring POTS Answering for more information.

Line Hunt

Create line hunt groups for multiple incoming callers on a first come, first served basis. This is ideal for separating groups of inputs and outputs between different studios or stations. See Line Hunt Call Answering for more information.

Important Note: The G3 profile setting supports maintaining specific G3 codec settings when answering a call from a G5 codec.

1.Auto: The codec will dial the G3 codec and connect in mono or stereo.

2.Dual Program: This allows the codec to dial a G3 codec with a Dual Program profile loaded and support two simultaneous mono connections.

3.Runtime: The G3 codec will retain runtime settings when answering a call from a G5 codec.

4.Custom: The G3 codec will load a specified profile, e.g. profile 6, which is the first custom profile number.

Important Note:

•If you select Sessionless as the Session Protocol select UDP/IP +RTP for RFC-compliant IP streaming.

•See RS232 Data Configuration for detailed information on RS232 data and see Enabling Relays and RS232 Data for more information on relay operations.

For IP connections configure the IP address, ports, and then specify which streaming interface is used to dial this connection, e.g. Primary (port LAN1) or Secondary (port LAN2). Note: By default Any will select LAN1 if it is available and LAN2 if it is unavailable.

Session Port 9002 and Remote Audio Port 9000 are used by default for the first IP audio stream. The audio for this audio steam will use analog or digital input and output 1 for both the dialing and answering codecs. If you need to change default port settings on the codec from which the call originates, this will also need to be adjusted in the answering codec using a custom program.

Click Next Stream to configure the first audio stream with default algorithm, jitter and FEC settings which are physically entered in the codec. Alternatively, click Next to specify individual algorithm, jitter buffer and FEC settings and configure a failover connection or SmartStream PLUS for this audio stream (recommended).

Click the drop-down arrows on the right-hand side of each text box to adjust the Encoding, Sample rate or Bit rate options.

For IP connections click to configure:

•Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer priority. It is also possible to configure the Minimum depth and Maximum depth of jitter over the connection. See Configuring the Jitter Buffer for more details.

•Alternatively, select a Fixed Buffer Level and enter the Jitter Depth, which must be between 12ms and 5000ms depending on the algorithm you select.

•Local and Remote FEC settings if required.

Click the check-box to select Enable Redundant SmartStream PLUS and configure configure dual redundant IP streaming. Alternatively, click Next to configure Auto Reconnect or a failover connection, whereby the alternative connection is dialed if the primary connection fails.

By default, primary IP streaming is via LAN1. To achieve the maximum level of redundancy select Secondary to configure redundant streaming from the secondary IP port LAN2. The redundant stream uses Remote Audio Port 9001 by default and the Local Audio Port allocated is Automatic. Note: Automatic indicates that the codec will arbitrarily allocate the local port value and send this information to the codec to which you are dialing.

Important Note:  Dual SmartStream PLUS redundant streaming over two IP interfaces mitigates lost packets on either link and will provide IP network backup if an IP link is lost. To learn more about SmartStream PLUS redundant IP streaming visit http://www.tieline.com/Transports/SmartStream-IP

For ISDN connections enter a number and select which B channel to use. Select the Enable bonded connections check-box to configure and bond multiple B channels.

Next, click Save Program to save the program with default algorithm settings, or click Next to specify a different algorithm and configure a backup connection if required. (recommended).

Dialing settings for this ISDN audio stream are now complete.

Select POTS Codec in the Mode drop-down menu to encode/decode using POTS, or select Analog Phone to configure a standard analog phone call, then click Next.

Next, enter the phone number of the codec or device you want to dial. When multiple POTS modules are installed, click the Via drop-down menu and select Module 1 or Module 2 to specify which POTS module will dial. Next, click Save Program to save the program with default settings, or click Next to specify algorithm settings and configure a backup connection if required (recommended).

Dialing configuration settings for this POTS audio stream are now complete.

Important Note: When Auto Reconnect is enabled, the dialing codec will continue to attempt a connection with the remote codec until Disconnect is pressed either on the dialing codec's keypad, or in the Web-GUI.

Screen Display

Description

1

Threshold

The percentage of lost data measured during a given time frame

2

Time Frame

The time frame against which lost data is measured

3

Keep Alive

The keep connection alive time before failing over to a backup connection; Tieline RTP pings every second to confirm connectivity

4

Failback Parameters

Select the check-box to fail back to a higher priority connection

5

Stable Time

The amount of time a primary connection must remain stable before attempting to fail back from the backup connection

6

Maximum Retries

The maximum number of fail back retries a codec can try before ending fail back attempts

7

Time Frame

The time frame used to measure the number of fail back retries attempted

Important Notes for Rules:

•Rules for connecting or disconnecting an audio stream are configured in the Program Manager panel. Rules for connecting or disconnecting a program are configured in the Rules panel. See Creating Rules for more information.

•A non-WheatNet-IP Tieline codec can be configured to trigger a logic IO in a Tieline WheatNet-IP codec. Up to 64 logic IOs are available in Genie Distribution and Merlin PLUS WheatNet-IP codecs, as well as 4 physical CONTROL PORT GPIOs.

•Connection-related rules are not displayed in Answer only programs.

Important Notes: Session Port 9002 and Remote Audio Port 9010 are used by default for the second IP connection. If you connect to a G3 codec using IP2 you will need to change the session port on the answering codec to 9012 for the second audio stream connection, as this is used by default on G3 codecs for IP2.