Other Settings

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You can adjust a number of miscellaneous settings through this dialog which normally aren’t altered. Note, however, that the correct configuration MUST be made for CODEC_NAME_ARIAL12_001 to operate!

 

Other settings configuration dialog:

 

others003

 

G.711 law

North America/Japan and Europe use a different coding method for ISDN telephony.

Thus, the following alternatives are available:

A-law (as used in Europe – uses non linear PCM, with 13 segments)
µ-law (as used in USA and Japan – also non-linear PCM; but 15 segments)

Note:

If G.711 law is not set properly then strong disturbances occurs at G.711 transmissions.

 

G.711 Attenuation

Sometimes G.711 signals (i.e. normal telephone signals) are over modulated and therefore their too high audio levels can cause distortions. Therefore G.711 attenuation can be set to:

0 dB (no attenuation)
6 dB
12 dB

 

G.722 SRT Timeout

- defines the G.722 ‘timeout’ (measured in Seconds).

Generally, there are two methods to synchronize G.722 data streams.

Generally, there are two methods of determining the ‘octet-start’ (byte-start) in G.722 data streams.

They are:

G.722 with H.221 (in-band signalling)
H.221 uses a small amount of the data capacity (i.e.1.6 kbit/s of 64 kbit/s) to transmit the synchronization information in-band.
[Although G.722/H.221 is a recommended EBU standard, it’s not very widespread!]
G.722 SRT (Statistical Recovery Timing)
Synchronization requires finding the start bit of every G.722 octet, and is reached by evaluating the data pattern of the signals statistically. We DON’T recommend that you use a pure sine wave signal whilst trying to achieve synchronization, using G.722 coding!

 

CODEC_NAME_ARIAL12_001 codecs will auto-detect a G.722/H.221 data-stream since they evaluate the H.221 in-band signalling information. However, as G.722 SRT signals do not have such signalling, CODEC_NAME_ARIAL12_001 only switches to the G.722 SRT setting when the ‘G.722 timer’ expires and framing has yet to be achieved. This G.722 timer starts once the first ISDN line is connected.

It is recommended that the G.722 SRT timeout be set to ~30 Seconds. This value is high enough not to interrupt any auto-detection process, but should still be acceptable for framing up.

Note:
You can enter values other than those shown, but the maximum possible value is 9,999 seconds

 

Disconnect Timeout

- defines the period (in seconds) after which a connection is dropped, if no valid framing is achieved. Ideally, it should be set higher than the G.722 SRT timeout value – its default value is 45 Seconds.

Note:
You can enter values other than those shown, but the maximum possible value is 9,999 seconds.

 

ISDN Answer Delay

- defines the ‘telegram’ interval (used in ISDN connections) - in milliseconds. It depends on the system software version how it works

codec_name_arial12bold_001 system software < 2.1.0.2:

With an ISDN connection, the called CODEC_NAME_ARIAL12_001 sends out a ‘telegram’ once the first B-channel is connected and the ISDN answer delay has expired.

The default value (of 3,000 ms) is usually long enough, but if there are framing problems with multiple B-channel ISDN connections, this value can be increased.

codec_name_arial12bold_001 system software >= 2.1.0.2:

With an ISDN connection, the called CODEC_NAME_ARIAL12_001 sends out its ‘telegram’ 10 times till the ISDN answer delay expires. The default value is 5,000 ms.

Note:

You can enter values other than those listed, up to the permitted maximum of 30,000 ms

 

ISDN Active Layer 1

- determines whether ISDN Layer 1 stays active all the time, or not.

When CODEC_NAME_ARIAL12_001 is used in the Netherlands, this value should definitely be set to ON.

 

Dial In Protection

- selects whether Dial-in Protection is activated for:

ISDN
IP
ISDN and IP

When Dial-in Protection is set to On, only connections from ISDN numbers (or IP addresses, as appropriate) which are stored in the Phone book will be accepted.

 

Accept Mode

- determines how Incoming connections are accepted.

auto:
Accepts all calls that are permitted by Dial in protection automatically
manual:
Accepts all calls manually
dbpos:
Accepts all calls that are permitted by Dial in protection manually
dbneg:
Accepts all calls that are not permitted by Dial in protection manually, the others are accepted automatically

 

Dual Mode At B-channel

With ISDN codec_name_arial12_001 can be used as two separate mono codecs. However, the following restrictions will apply:

Maximum sample rate:        32 kHz
Mode:                        mono
Maximum bit rate :                64 kbit/s (except for MPEG L2)

This setting defines on which B-channel the ‘second (sub) codec’ starts to communicate.

For example:

‘Dual mode on B-channel = 2’ means that B-channel 1 is assigned to the first (sub)codec and all other B-channels are assigned to second one.

If dual mode at B-channel = 0 then dual mode is de-activated, and CODEC_NAME_ARIAL12_001 functions as a single codec, without the above restrictions.

Note:

Dual mode just can be set if the communication interface of both sub codecs is set to ISDN. If this is not true dual mode can not be set or does not work properly pending on the system software version.

 

Select Sub codec

Although at special applications (e.g. Dual mode, Backup, Gateway) CODEC_NAME_ARIAL12_001 supports more than one sub code, only one input interface is available. This setting lets the user select to which internal (sub) codec the input and display applies to.

If dual-, backup or gateway-mode is used, the selected sub codec (either 1 or 2) is displayed in the main program window. This differs with a codec_name_arial12_001 4000/4001, which contains 4 codecs!

The sub codec can also be selected in the main window with help of short cut keys. More info about short cut keys can be found in the help chapter Wheel button dialog and special keys/Short Cut Keys.

 

Multi Point Return Channel

- determines which return Channel is used on a Point-to-Multipoint transmission.

[Using ISDN, Point-to-Multipoint means that a CODEC_NAME_ARIAL12_001 codec can transmit one audio signal out to multiple ISDN destinations. However, with Point-to-Multipoint (PMP) CODEC_NAME_ARIAL12_001 just can decode one return channel. This option allows you to define which of the return paths you wish to use.]

PMP functionality works with any of the following:

codec_name_arial12_001
MERKII
Worldnet Tokyo
Millennium 2
CDQPrima
CDQ2000/2001

Note:

If PMP value is not set to 0, then the PMP return is also displayed in the main window.

If the value IS set to 0, PMP is deactivated.

The number of possible return channels is calculated by:

number of PMP-channels = (number of B-channels*64kbit/s) /encoder bit rate

Note 2:

With system software version 2.0.0.57 (or later) even negative return channels can be selected. A negative return channel means that all multi point locations still receives the audio but the central codec_name_arial12_001 does not decode any audio.

 

UDP Broadcast Mode

- enables UDP broadcast mode. All UDP audio datagrams are now addressed to the UDP “broadcast” address (255.255.255.255), and UDP broadcast reception is enabled.

 

Backup

- enables to configure the backup feature of the codec_name_arial12_001 if the referring additional function is implemented (see menu item Expert/Set Key Code). Backup feature means that codec_name_arial12_001 establishes automatically another connection if framing is lost. Two Backup parameters can be set:

Backup time
In the list box the backup time can be selected in ms. Backup time is the time the backup event (= framing lost) must minimum last before the backup connection is established. If ‘off’ is selected the backup functionality is switched off.
Note:
You can enter values other than those listed, up to the permitted maximum of 30,000 ms.
Backup connection
In the right edit field the backup connection can be selected. If it is empty then the last connection will be used. Additionally a phone book entry can be selected:
- by typing in the number of the phone book entry
or
- by mouse double clicking to open the phone book

Note:

A general description of the CODEC_NAME_ARIAL12_001 backup feature can be found at chapter Miscellaneous/Backup.

 

Gateway

- enables to configure the gateway feature of the codec_name_arial12_001 if the referring additional function is implemented (see menu item Expert/Set Key Code). Gateway functionality means that codec_name_arial12_001 routs received audio from one interface to another one (e.g. MPEGL2 audio received at ISDN is sent out as MPEGL3 audio via IP) The following parameters can be selected:

off: Gateway functionality is disabled
last: Routing to last established connection
phone book: Routing to a phone book entry

The phone book entry can be selected in the right edit field:

by typing in the number of the phone book entry

or

by mouse double clicking to open the phone book

Note:

A general description of the CODEC_NAME_ARIAL12_001 gateway feature can be found at chapter Miscellaneous/Gateway.