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Page 626 / 998 Scroll up to view Page 621 - 625
VoIP phone ports
Port configuration
iMG/RG Software Reference Manual (Voice Service)
6-62
•
Digit Map/Dial Mask
•
Voice Coder/Decoder
•
Voice Quality Management
•
Telecom Tone Management
6.3.1.1 Digit map
The
Digit Map
is a rule used by the access port to understand when dialling is to be considered completed and
the dialled number is ready to be processed by the call control layer. It works for outgoing calls (in the direction
from user to VoIP network).A digit map is defined either by a (case insensitive)
string
or by a list of strings.
Each string in the list is an alternative numbering scheme, specified either as a set of digits or timers, or as an
expression over which the port will attempt to find a shortest possible match. The following constructs can be
used in each digit map:
•
DTMF
A digit from '0' to '9' or one of the symbols ‘A’, ‘B’, ‘C’, ‘D’. Symbols ‘#’ or ‘*’, if necessary, have to be added
separately.
•
Timer
The symbol 'T' matching the timer expiry. The symbol 'T' at the end of Digit Map indicates that if user has
not dialled a digit for a time longer than the value of the inter-digit time, the dialled number shall be consid-
ered complete. If the symbol T appears in the middle of digit map expression is not considered and skipped
during expression evaluation.
•
Wildcard
The symbol ‘x’, which matches any digit (‘ 0’
to ‘9’ ).
•
Range
One or more DTMF symbols enclosed between square brackets (‘ [‘ and ‘]’).
•
Subrange
Two digits separated by a hyphen (‘ -’ ) that matches any digit between and including the two. The subrange
construct can only be used inside a range construct, i.e., between ‘[‘ and ‘]’.
•
Position
A period (‘.’), which matches an arbitrary number, including zero, of occurrences of the preceding construct.
Also, note that the whole
Digit Map
shall not exceed 128 characters.
Let’s consider an example where the user in an office wants to call a co-worker’s 3-digit extension. The
Digit
Map
is defined in such a way that the called number is processed after the user has entered 3 digits.
The command to set the
Digit Map
could look as follows:
voip ep analogue set prt0 digitmap xxx
This
Digit Map
specifies that after the user has entered any three digits, the call is placed. It's possible to refine
this Digit Map by including a range of digits. For example, if all extensions in the user company begin with 2, 3,
or 4, the corresponding
Digit Map
command could look as:
Page 627 / 998
Port configuration
VoIP phone ports
6-63
iMG/RG Software Reference Manual (Voice Service)
voip ep analogue set prt0 digitmap [2-4]xx
If the number dialled begins with anything other than 2, 3, or 4, the call is rejected and a busy tone is generated.
Another way to achieve the same result would be:
voip ep analogue set prt0 digitmap [234]xx
It is possible to combine two or more expressions in the same Digit Map by using the ‘|’ operator, which is
equivalent to OR. The left-most expression has precedence over the other expressions
Let’s consider the case of a choice: the Digit Map must check if the number is internal (an extension), or exter-
nal (a local call). Assuming that dialling ‘9’ makes an external call, the Digit Map could be defined with the com-
mand:
voip ep analogue set prt0 digitmap ([2-4]xx|9[2-9]xxxxxx)
In this case the
Digit Map
checks if the number begins with 2, 3, or 4 and the number has 3 digits
If not, it checks if the number begins with 9 and the second digit is any digit between 2 and 9 and the number
has 7 digits
It may sometimes be required that users dial the ‘#’ or ‘*’ to make calls.
This can be easily incorporated in a Digit Map with the command:
voip ep analogue set prt0 digitmap xxxxxxx#|xxxxxxx*
The ‘#’ or ‘*’ character could indicate that users must dial the ‘#’ or ‘*’ character at the end of their number to
indicate it is complete.
When the outgoing call is in process, the call control layer removes any '#'', '*' and 'T' symbols from the dialled
number.
6.3.1.2 Dial mask
The Dial Mask specifies the number of digits that must be removed from the dialled number
before
checking the
dialled number against the
Digit Map
.
6.3.1.3 Voice coder/decoder
The Voice system makes use of a specific DSP with an embedded sigma-delta Coder/Decoder to process voice
and data from/to access ports.
Different codec types are available in order to satisfy the requirements of different environments.
It's possible to specify more than one codec type for each port using the command VOIP EP ANALOGUE SET
CODECS.
The codec specified at the leftmost ends of the codec list has precedence over the other codecs.
Page 628 / 998
VoIP phone ports
Port configuration
iMG/RG Software Reference Manual (Voice Service)
6-64
The signalling protocol (SIP) will negotiate the active codec based on the capabilities supported by the other
peer involved in the VoIP connection. In the case of local calls, the call control layer performs codec negotiation
locally.
The following table lists the codecs available on the iMG units.
6.3.1.3.1 T.38 support
The iMG is designed to support the transmission of T.30 fax signals using T.38 Internet Fax Protocol (IFP) pack-
ets.
Although T38 is listed the as a supported codec for the iMG family, T.38 is not in itself a codec, but rather a
technical solution to map FAX signals into a dedicated IP protocol - overriding the limitations (e.g. signal distor-
tion) that are present when faxes are sent using codecs designed for speech applications.
When T.38 support is enabled and a fax must be sent or received, the intelligent Multiservice Gateway tries
firstly to negotiate T.38 support with the called or calling end-point respectively. If this fails, the iMG automati-
cally falls-back to a non-compressed codec such as G711 A-law or
μ-law
.
6.3.1.4 Voice quality management
To increase the voice/data quality additional parameters can be set on the voice system DSP. The following set-
tings are available on iMG models:
•
A fixed jitter buffer. Set to 120 ms with a jitter delay of 60 ms.
•
Separate TX and RX direction volume gain control. Adjustable between -48dB and +24dB.
•
ITU-T G.168 Line Echo Cancellation. Adjustable between 0 and 32 msec (a value of 0 disabled Line Echo
Cancellation).
•
Voice activity detection (VAD)/comfort noise generation (CNG).
TABLE 6-11
Codecs Available for iMGs
Codec
Notes
g711a
G.711 A law
g711
u
G.711 μ law
g729ab
G.729 Annex A and Annex B
g726-32
G.726 32kbps
T38
Media/codec negotiation for transmission of ITU-T T.30 fax signals via internet.
This is not an actual codec, but when specified in the codecs list indicates to the
iMG that ITU-T T.38 negotiation of media sessions & fall-back codecs should be
enabled.
Page 629 / 998
Port configuration
VoIP phone ports
6-65
iMG/RG Software Reference Manual (Voice Service)
•
Telecom tone management and DTMF relay.
•
This is a SIP protocol dependent solution used to transfer DTMF tones out-of-band either using SIP INFO
messages, or by means of RFC2833 ‘Named DTMF Events’ within the RTP stream. The underlying logic is as
follows:
•
When the iMG attempts to establish a call, it adds RFC2833 ‘Named Telephone Event’ to the capabilities
listed for RTP packets, but only if a compressed codec (g726 or g729ab) has been configured for the
Voice access port involved in the call.
•
If a call is then established using an uncompressed codec (i.e. g711u or g711a), the iMG will send DTMF
tones in-band - irrespective of whether or not the called endpoint supports RFC2833 Named Telephone
Events.
•
If however a the call is established using a compressed codec, the iMG will send DTMF tones using
RFC2833 Named Telephone Events, but only if the called end-point supports this mechanism - otherwise
it switches to the same path used for voice (accepting DTMF distortion).
When the intelligent Multiservice Gateway is going to accept a call, it adds to the capabilities list the RTP packet
Named Telephone Event only if a compressed codec (g726 or g729ab) has been configured for the Voice access
port involved in the call.
•
Then if the call is established using an uncompressed codec (i.e. g711u or g711a), the intelligent Multiservice
Gateway will send DTMF tone in-band (independently of whether the caller endpoint supports RTP packet
Named Telephone Event) on the same path used for voice.
•
If the call is established using a compressed codec, the intelligent Multiservice Gateway will send DTMF
tones using RTP packet Named Telephone Event only if the caller end-point supports it, otherwise it
switches to the same path used for voice (accepting DTMF distortion).
•
Inter-digit time/Inter-digit critical time.
Inter-digit time
is the maximum acceptable time between the dialling of one digit and the next. If a time
longer than the
Inter-digit time
elapses after the dialling of a digit, dialling is considered complete. The timer
‘T’ in the digit map expression uses the
Inter-digit time
value. To change the value of the
Inter-digit time
use
the VOIP EP ANALOGUE SET IDT-PARTIAL command.
Inter-digit critical time
is the maximum acceptable time between the off-hook event and the dialling of the
first digit. If a time longer than this has elapsed since off-hook and dialling has not yet started, then the con-
nection is closed and a busy tone is generated. To change the value of the
Inter-digit critical ti
me use the
VOIP EP ANALOGUE SET IDT-CRITICAL command.
•
Off-hook time / On-hook time / Flash-Hook time.
•
Off-hook time is the minimum time (msec) that the analogue line must stay in off-hook before the sys-
tem detects the off-hook state.
•
On-hook time is the minimum time (msec) that the analogue line must stay in on-hook before the sys-
tem detects the on-hook state.
•
Flash-hook. The flash-hool period may vary between countries, and the iMG flash-hook time parameter
allows the iMG user to allow for this. Note that flash-hook time can not be greater that on-hook time.
Page 630 / 998
VoIP phone ports
Port configuration
iMG/RG Software Reference Manual (Voice Service)
6-66
The iMG detects a flash-hook event when the on-hook period falls within a a time window. The lower
bound of this window is one third of the configured flash-hook time, and the upper bound is the lesser
of on-hook time and double the configured flash-hook time.
6.3.1.5 Country-specific telecom tones
The iMG is able to reproduce the same country-specific telecom tones used by Central Offices or Foreign
Exchanges simply by selecting the preferred country via the VOIP EP ANALOGUE SET COUNTRY command.
Dial Tone, Busy Tone
and
Ring Back T
one refer to ITU-T E.180 specifications as reported in the following
table:
TABLE 6-12
Country-specific Telecom tones
Country
Dial Tone
Busy Tone
Ring Back Tone
Frequency
(Hz)
Cadence
(msec)
Frequency
(Hz)
Cadence
(msec)
Frequency
(Hz)
Cadence
(msec)
Australia
425x25
Continuous
400
375 - 375
400x17
400 - 200 -
400 - 2000
Austria
450
Continuous
450
300 - 300
450
1000 - 5000
Belgium
425
Continuous
425
500 - 500
425
1000 - 3000
Canada
350+440
Continuous
480+620
500 - 500
440+480
2000 - 4000
China
450
Continuous
450
350 - 350
450
1000 - 4000
France
440
Continuous
440
500 - 500
440
1500 - 3500
Germany
425
Continuous
425
480 - 480
425
250 - 4000 -
1000 - 4000 -
1000 - 4000
Israel
400
Continuous
400
500 - 500
400
1000 - 3000
Italy
425
600 - 1000 -
200 - 200
425
200 - 200
425
1000 - 4000
Japan
400
Continuous
400
500 - 500
400x16
1000 - 2000
New Zealand
400
Continuous
400
500 - 500
400 + 450
400 - 200 -
400 - 2000
Norway
no tone
//
425
1000 - 4000
425
500 - 500
Russia
no tone
//
425
400 - 400
425
800 - 3200

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