Voice over IP OXO

VoIP parameters Page 1 of 19
VoIP parameters
General description
This screen contains the parameters for voice programming on IP. It consists of the following tabs:
VoIP: Parameters / General
VoIP: Parameters / Gateway
VoIP: Parameters / Gatekeeper (parameters moved to different configuration screens as of OXO
Connect R3.0)
VoIP: parameters / DSP (parameters moved to different configuration screens as of OXO Connect
R3.0)
VoIP: Parameters / DHCP (parameters moved to different configuration screens as of OXO Connect
R3.0)
VoIP: Parameters / Fax (parameters moved to different configuration screens as of OXO Connect
R3.0)
VoIP: Parameters / SIP Trunk
VoIP: Parameters / SIP Phone
VoIP: Parameters / Codecs (parameters moved to different configuration screens as of OXO
Connect R3.0)
Note: If SIP (Session Initiation Protocol) is selected in the General tab, no tab is displayed for the
Gatekeeper (H.323). If H.323 is selected, no tab for SIP is displayed.
Buttons description
OK This button is used to validate your modifications
Cancel This button is used to cancel your modifications
This button, available as of OXO Connect/OXO Connect Evolution R3.0, provides a

Advanced complementary and basic configuration of parameters available in previous releases
as global VoIP noteworthy addresses.
System Limits
DSP are provided by the PowerCPU EE board, optionally equipped with an ARMADA daughter board.
The PowerCPU EE board natively provides 16 DSP resources and supports an optional ARMADA VoIP
daughterboard with additional DSP resources.
The PowerCPU EE board equipped with an ARMADA VoIP32 daughterboard supports up to 48 DSP channels.
The PowerCPU EE board equipped with an ARMADA VoIP64 daughterboard supports up to 60 DSP channels
(multi-codec) or 76 DSP channels (G711).
The maximum number of VoIP trunks is 48 (76 with an ARMADA VoIP64 daughterboard).
file:///C:/Users/USER/AppData/Local/Temp/~hh8AAC.htm 7/5/2024
The number of external VoIP accesses depends on a software key.
VoIP parameters / General
Fields description
VoIP Channels mode
This field gives the combinations of multi-codecs and G711 codecs, available for the current
Description
hardware.
l OXO Connect Evolution: the field is greyed and shows Multi-codecs

l OXO Connect without Armada daughterboard:the field is greyed and shows Multi-
codecs [16] (also applicable for OmniPCX Office RCE up to R10.3).
l OXO Connect with an ARMADA VoIP32: the field is grayed and shows Multi-codecs
[48]. (also applicable for OmniPCX Office RCE up to OmniPCX Office RCE R10.3).
l OXO Connect with an ARMADA VoIP64, the dropdown list offers two choices:
Values
¡ Multi-codec [60]
¡ Multi-codecs [30] + G711 [46]
When activating Extended mode (multi-codec + G711 channels), the OMC displays a
text to inform that the system and IP phones must be configured for the preferred
G711 codec.
Whenever the VoIP channels codec mode is modified for Armada64 VoIP
daughterboard, the OMC displays a popup warning that a system reset is required.
Number of VoIP Trunk Channels

Note: This parameter is available only in versions of OXO Connect up to R2.x.
The number of VoIP access channels defines the number of channels assigned for
communications in IP mode for external calls. It is linked to the number of VoIP-subscriber
channels. These channels must be entered in a trunk group (see Procedure).
The link categories and barring parameters of the trunk group and the IP
Description
terminals must be defined. The number of external VoIP accesses depends on
a software key.
ATTENTION:
VoIP Subscriber channels and Trunk channels must not exceed the available
physical channels (multi-codec channels + G711 channels) and license
limitations. Trunk channels have priority over Subscriber channels.
Procedure File Configure Voice over IP (individual case) (Stage 3)
Number of VoIP Subscriber Channels

The number of VoIP-subscriber channels is the part of DSP channels reserved for IP
Description subscribers (from the total amount of DSPs available on the PowerCPU EE board, optionally
equipped with an ARMADA daughter board).
Up to 48 (absolute maximum), but the effective maximum is given by:
Total no. of VoIP channels - No. of VoIP-trunk channels
VoIP Subscriber channels and Trunk channels must not exceed the available physical
channels (multi-codec channels + G711-only channels) and license limitations. Trunk
Values channels have priority over Subscriber channels.
If the total number of VoIP channels is 20 and the configured number of VoIP-
EXAMPLE: trunk channels is 13, the number of available VoIP-subscriber channels is 20 - 13
= 7.
Number of VoIP Channels

This parameter is visible on this screen as of OXO Connect 3.0.
This field indicates in read-only mode the total number of VoIP channels available on the
Description
global pool of VoIP channels of the system.
The size of this pool is hardware (for OXO Connect) or software (for OXO Connect Evolution)
Values
dependent
VoIP Channels for trunks with reservation

This field indicates in a read-only mode the number of VoIP channels (DSPs) that have been
Description
explicitly reserved by the VoIP trunk(s).
Values [0 - Number of VoIP Channels]
VOIP Channels for IP phones and trunks without reservation

This field indicates in a read-only mode the number of VoIP channels (DSPs)
available in the global pool and usable by both IP-subscribers and VoIP-trunks.
Description
In others words, it represents the number of VoIP channels that have not been
reserved for VoIP trunk(s).
Number of Trunk Channels for trunks without reservation

This field indicates in a read-only mode the number of trunks channels (dedicated to
Description VoIP trunks) for which no VoIP channel has been reserved. These trunk channels may
use VoIP channels available in the global pool.
Number of 'SIP Trunk channels' licenses

This field indicates in a read-only mode the maximum number of SIP Trunks
Description channels licenses. This information is also available in the Software key features
window (see: Software key features )
'SIP Trunk channels' licenses for trunks with reservation

This field indicates in a read-only mode the number of SIP Trunks channels licenses
that have been explicitly reserved by the VoIP trunk(s).
Description The field name is highlighted in red when the number of licenses used in the
configuration of VoIP trunks (via the Reserved mode option) is greater than the
number of available licenses.
Values [0 - Number of SIP Trunks channels licenses]
'SIP Trunk channels' licenses for trunks without reservation

This field indicates in a read-only mode the number of SIP Trunks channels licenses
Description that have not been reserved by VoIP trunk(s) (Reserved mode option deactivated in
VoIP trunk settings).
Values [0 - Number of SIP Trunks channels licenses]
IP quality of service
The quality of service must correspond to that specified in the LAN network. The
characteristics of a network (transmission speed, etc.) can be measured, improved and, in
Description some ways, guaranteed in advance through the choice of a quality of service.
As of OmniPCX Office RCE R10.0, it is possible to define a separate QoS value for RTP flow
and SIP Trunk signaling.
Procedure File Configure Voice over IP (individual case) (Stage 3)
VoIP Protocol
Description Determines the VoIP protocol type to use: H.323 (listed under Gatekeeper tab) or SIP.
Select either H.323 or SIP (Session Initiation Protocol).
Procedure NOTE: After you select a protocol and the system resets, If SIP is selected, no tab is
displayed for Gatekeeper. If H.323 is selected, no tab for SIP is displayed.
RTP direct
This parameter is only displayed in a consistent SIP protocol configuration.
Note: This field is visible only in versions of OXO Connect up to R2.x. As of OXO Connect R3.0, this
parameter is accessible from the window: Gateway parameters in latest versions.
Description This field is available only if in VoIP protocol, the SIP is selected.
Value Checked or unchecked
Codec Pass-Through for SIP phones

This parameter is only displayed in a consistent SIP protocol configuration.
Description This field enables or disables codec pass-through for SIP phones.
Value Yes or No. Default value = No (unchecked)
Codec Pass-Through for SIP trunks

This parameter is available as of OmniPCX Office RCE R8.1 and can be configured only when RTP direct is
enabled.
This field enables or disables codec pass-through for SIP trunks.

Description NOTE: This field is taken into account only if the ARS route used has its "Codec/framing"
parameter set to 'default'.
G711 Codec for Music on Hold and preannouncements

This parameter is available as of OmniPCX Office RCE R9.0.
This field allows to use the G711 codec in priority for Music on Hold on SIP trunks and for
preannouncements on SIP trunk incoming calls.
This field is only relevant when the following conditions are true:
l The "codec/framing" trunk gateway parameter is set to "default"

l The Default Codecs List defined in OMC includes G711
Description For MOH, the G711 codec is available in the first position of the codec list, for codec
negotiation proposal.
For preannoucements, if the offer includes the G711 codec, the first G711 codec is selected
for the answer.
RTCP attribute in SDP (as of OmniPCX Office RCE R10.0)

This field covers the need for the declaration of an RTCP attribute to document the RTCP
port used in SDP (RFC3605).
When activated, an RTCP attribute is added by the PCX in all requests containing SDP. When
Description receiving requests containing this attribute, the PCX interprets it and uses the specified port
for RTCP flow.
This parameter is only available when VoIP relies on the SIP protocol. In all other cases, this
field is not displayed.
Value Yes or No. Default value = Yes (checked)
VoIP parameters / Gateway
General description
This screen shows all the timer parameters for remote IP access and the value of SIP Trunk Signal Source
Port (as of OmniPCX Office RCE R9.1).
Fields description
The default values of the timer parameters for remote IP access are as follows:
l RAS Request Timeout: 20 s (this field is relevant only for versions of OXO Connect up to R2.x)
l Remote Gateway Presence Timeout: 50 s (this field is relevant only for versions of OXO Connect up
to R2.x)
l Connect Timeout: 500 s
l H.245 Request Timeout: 40 s (this field is relevant only for versions of OXO Connect up to R2.x)
l H.323 End of Dialing Timeout:5.0 s
NOTE: This parameter applies also to SIP trunks.
The default value of SIP Trunk Signal Source Port is 5060.

NOTE: This field is visible only in versions of OXO Connect up to R2.x. As of OXO Connect R3.0, this
parameter is accessible from the window VoIP parameters / SIP trunk.
NOTE: Any modification of this value requests a system restart.
NOTE: This value must be different from 0 and from the SIP Phone Signal Source Port value configured in
VoIP: Parameters / SIP Phone screen.
Select the End of Dialing table used and RTP Tickets activation, where applicable.
NOTE: The End of Dialing table used field is relevant only for versions of OXO Connect up to R2.x.
See also the remote proxy SIP configuration in the ARS table.
VoIP parameters / Gatekeeper
This screen applies to OXO Connect up to R2.x. It is no more available as of OXO Connect R3.0
General description
This screen is used to set parameters to protect IP communications from the external environment. This
protection is carried out using a Gatekeeper (secured directory server or gatekeeper). The system offers:
l either the use of the Gatekeeper integrated into the CPU board,
l or the use of an external Gatekeeper by specifying its IP address.
NOTE: This screen is used for H.323, it is not visible when SIP is selected.
Fields description
Integrated Gatekeeper
If this box is checked, the system will use the integrated Gatekeeper in the CPU board by
Description
default.
Procedure File Configure Voice on IP (individual case) (Stage 3)
Identification of the Gatekeeper
If the Gatekeeper box is not checked, the IP address of the Gatekeeper used must be
Description
specified here.
VoIP parameters / DSP
General description
This screen is for the parameters of the DSP paths.
Fields description
Law Mode
Description This field cannot be modified. It specifies the encoding law used.
A: for all countries except Asian countries
Values
Mu: for Asian countries
Echo cancellation
Checking the box will cancel the echo on the IP links. However, if these links are poor, it is
Description
best to leave the box unchecked.
Active voice detection
Description Checking the box helps in reducing the bandwidth consumption for VoIP trunks/calls.
VoIP parameters / DHCP
General description
This screen is used to manage the assigning of IP addresses to all the elements on the LAN network.
Fields description
Activate integrated DHCP server
Checking this box allows IP addresses to be assigned automatically to all the elements on
the LAN network by using the internal DHCP (Dynamic Host Configuration Protocol: a
Description software application that enables IP addresses to be assigned dynamically). The start and
end of these addresses are found in the Start and End fields of IP addresses in the Dynamic
range box.
Dynamic range
The Start and End fields of IP addresses must contain the IP addresses assigned to the
Description elements on the LAN network. These fields cannot be modified if the Activate integrated
DHCP server box is checked.
VoIP parameters / Fax
Note: This tab is visible only in versions of OXO Connect up to R2.x. As of OXO Connect R3.0, parameters
from this tab are accessible from the window: Gateway parameters in latest versions.
General description
This screen is used to manage the parameters specific to Fax over IP, in T38 mode.
Fields description
UDP Redundancy
In order to ensure the reliability of the UDP transmission, the packets are sent several times
Description to ensure that the information reaches its destination; this mechanism is called "UDP
Redundancy".
Value Between 0 and 2 (default = 1).
Framing
In order to reduce bandwidth use, the framing mechanism (RTP Packetization) allows the
Description
compression of packets of the same type of compression as Fax packets.
Value Between 0 and 5 (default = 3).
Error Correction Mode
As of OmniPCX Office RCE R10.0, this parameter enables/disables the FAX ECM mode on IP
Description trunk.
This parameter is available in both H.323 and SIP VoIP protocol mode.
Value This feature is disabled by default.
VoIP parameters / SIP trunk
General description
This screen is used to set specific SIP-trunking parameters. You can also set parameters to protect IP
communications from the external environment.
The values required in this tab are defined by your SIP provider. Refer to the subscriber form, given to you
by your provider, to complete the following fields.
Note1: To set additional SIP authentication details, see Gateway parameters in previous versions or
Gateway parameters in latest versions
Note2: This tab is available if the parameter VOIP Protocol is set to SIP in the VOIP Parameters General
tab (VoIP Parameters/General).
The parameters "REGISTRATION", "AUTHENTICATION" and "LOCAL DOMAIN NAME" must be

NOTE:
configured either in Numbering > ARS > Gateway Parameters or Automatic Routing: Prefixes.
Fields description
Timers
Timer T1 Enter the value of the T1 timer (in milliseconds). Default value = 1000.
Enter the allowed number of message retransmissions between 1 and 6.
Number of Retries
Default value: 6
Eth0 parameters
IP Quality of Service for Signal
Allows to set up the QoS value specific to SIP signalling.

The default value and behavior is identical as for the IP Quality of service field
under VOIP Parameters -> General -> IP Quality of Service, see:
Description
VoIP_Parameters_General
Note: When the OMC is connected to OmniPCX Office RCE before R10.0, this
field is grayed out and cannot be modified.
SIP Trunk Signal Port (as of OXO Connect R3.0)
Enter the value of SIP Trunk signal source port.

Default value: 5060
Description NOTE: Any modification of this value requests a system restart.
NOTE: This value must be different from 0, from the SIP Phone Signal Port value
configured in VoIP: Parameters / SIP Phone screen and from the SIP TLS Trunk
Signal Port configured in this screen.
SIP TLS Trunk Signal Port (as of OXO Connect Evolution R3.2)
Enter the value of SIP TLS Trunk signal port.

Default value: 5061
NOTE: In case of data base restore from R3.1 or lower to R3.2 or higher, this
Description field is set to default value.
configured in VoIP: Parameters / SIP Phone screen and from the SIP Trunk Signal
Port configured in this screen.
UDP to TCP Switching
This parameter offers the possibility to deactivate the transport switching of

UDP to TCP Switching UDP to TCP.
Default value: Activated (deactivated in Orange Business System configurations)
Eth1 parameters (relevant only for OXO Connect Evolution R5.0 or higher)
IP Quality of Service for Signal
Allows to set up the QoS value specific to SIP signalling.

Description The default value and behavior is identical as for the IP Quality of service field
under VOIP Parameters -> General -> IP Quality of Service, see: VoIP
parameters / General
SIP Trunk Signal Port

Default value: 5060
configured in VoIP: Parameters / SIP Phone screen and from the SIP TLS Trunk
Signal Port configured in this screen.
SIP TLS Trunk Signal Port
Enter the value of SIP TLS Trunk signal port.

Default value: 5061
NOTE: In case of data base restore from R3.1 or lower to R3.2 or higher, this
Description field is set to default value.
configured in VoIP: Parameters / SIP Phone screen and from the SIP Trunk Signal
Port configured in this screen.
UDP to TCP Switching
This parameter offers the possibility to deactivate the transport switching of

UDP to TCP Switching UDP to TCP.
Default value: Activated (deactivated in Orange Business System configurations)
DNS Authentication
For DNS-enabled ARS lines, this parameter enables acceptance control of SIP
incoming calls, based on the IP address of the remote caller.
DNS Authentication
Default value: Deactivated (activated in Orange Business System
configurations)
Registration
From version OmniPCX Office RCE R8.0 onwards, the parameters "REGISTRATION", "AUTHENTICATION" and
"LOCAL DOMAIN NAME" must be configured either in Numbering > ARS > Gateway Parameters or Automatic
Routing: Prefixes.
Validate this option to enable the Registration feature.

Requested
When this option is not validated, the parameters below are unavailable.
Registered User Enter your Registered User Name.

Name
Registrar IP
Enter the IP address of the Registrar server of your provider.
Address
Enter the port number to use on the Registrar server.
Port
5060 is the standard value.
This check box describes the status of the DNSSRV feature. By default, DNSSRV is
disabled.
If checked (DNSSRV enabled), the Registrar IP Address, Port and
DNS SRV Outbound Proxy IP fields are set to their default values and are not editable.
The DNS server used is defined in: Gateway parameters.
If unchecked (DNSSRV disabled), the Outbound Proxy field is set
to its default value and is not editable.
Parameter is a 50+1 characters string corresponding to the Registrar’s hostname.
Registrar Name
Default is an empty string.
This parameter describes the next SIP hop - the very next SIP equipment -
the PBX must send the messages to, using a hostname format.
Parameter is a 50+1 characters string, which is accessible when DNSSRV is enabled.
Outbound Proxy List of authorized characters:
abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ1234567890-.
Default is an empty string when the DNSSRV feature is enabled.
Value is reset to default when the DNSSRV feature is enabled or disabled.
This parameter describes the next SIP hop - the very next SIP equipment -
the PBX must send the messages to, using an IP address format.
Parameter is an IP address which is accessible when DNSSRV is disabled.

Outbound Proxy Default is a null 0.0.0.0 IP address (which is not displayed) when the DNSSRV
IP feature is disabled.
Value is reset to default when the DNSSRV feature is enabled or disabled
Expiration Time Enter the life time (in second) of the registration. Default value = 3600.
Authentication
From version OmniPCXOffice R8.0 onwards, the parameters "REGISTRATION", "AUTHENTICATION" and
Routing: Prefixes.
Username Enter your Username.

Shared secret Enter your Password.
Registered realm Enter the validity domain of the Authentication.
Blacklist behavior
Note: The following fields are visible only in versions of OXO Connect up to R2.x. As of OXO Connect R3.0,
these parameters are accessible from the window: Gateway parameters in latest versions.
A Proxy server is declared in denial of service when it transmits more than a

defined number of messages in less than a defined time.
Message Peak Number
Enter the number of messages allowed before declaring a denial of service.
Default value = 90.
Enter the period (in seconds) for a denial of service detection. Default value =
Period Peak Detection
3.
Enter the delay (in seconds) during which a Proxy server declared in denial of
Quarantine Time
service is not polled.
When a SIP proxy is considered as unreachable by the system, it is inserted in a
list of unreachable proxies to avoid further requests to be directed to it.
The quarantined SIP proxy shall be removed automatically from the unavailable
list after the current timer, whose value can be modified.
This timer is specified in minutes, with a default value of 10 and allowed
Unreachable Proxy List range of [0, 1440].
Timer
1440 corresponds to a one day duration.
As of OmniPCX Office RCE R10.0,this parameter covers the implementation of
the sipgw_dns_dubl data from VOIPnwaddr noteworthy address.
As of OmniPCX Office RCE R10.0, setting this parameter to value 0 entails that
the unreachable proxy list is not used.
Local Domain Name

From version OmniPCX Office RCE R8.0 onwards, the parameters "REGISTRATION", "AUTHENTICATION" and
Routing: Prefixes.
Enter your Local Domain Name. If your Local Domain Name has not been defined
DNS name
by your SIP provider; enter your IP address.
RTP Proxy: Fixed ports
This parameter allows to either fix the ports during a communication or let
them change dynamically (default behavior).
This parameter is enabled and grayed out when a SIP gateway is associated to
RTP Proxy: Fixed ports the Eth1 LAN Interface of an OXO Connect Evolution.This parameter is not
automatically disabled when any or all SIP gateways associated to Eth1 LAN
interface are deleted. This parameter is configurable and can be disabled when
all SIP gateways associated to Eth1 LAN Interface are deleted.
Security (as of OXO Connect R6.0)

When OMC R60.0 is connected to OXO Connect in a version prior to R6.0, these fields are empty and
grayed out.
Enter the IP address of OXO Connect Evolution Front End used to provide SIP
OCE FrontEnd IP TLS/SRTP service to OXO Connect.
Default value: empty.
Port number of OXO Connect Evolution Front End is set to 5060 and cannot be
OCE FrontEnd Port modified.
VoIP parameters / SIP Phone
General description
This screen is used to define the server parameters associated to all SIP phones (8002/8012 DeskPhone,
8082 My IC Phone, 8088 Smart Deskphone, 4135 IP conference phone, 8135s IP conference phone and
generic SIP phones).
Fields description
Default Transport Mode
Select the default transport mode to be used by 8002/8012 DeskPhone, 8088

Smart Deskphone and 8082 My IC Phone sets.
Description Default value: UDP
Select TCP if the SIP packet length is greater than the MTU value or if some
SIP endpoints require it.
Domain Name
Description Enter the domain name of the SIP phones.
Authentication Realm
Description Enter the authentication domain in which user and password are defined.
SIP Phone Signal Port (as of OmniPCX Office RCE R9.1)

Default value: 5059
NOTE: This value must be different from 0 and from the SIP Trunk Signal Port
and SIP TLS Trunk Signal Port values configured in VoIP parameters / SIP
trunk screen.
Timers
The following parameters are visible on this screen as of OXO Connect R3.0.
Enter the allowed number of message retransmissions between 1 and 6.
Number of Retries
Default value: 6
Registration
Enter the time after which a SIP phone will retry to register after a failed
Register Retry Time registration. Default value = 300s. Minimum value = 60s. Maximum value =
36000s.
Enter the time after which the registration expires. Default value = 120s.
Register Expire Time Minimum value = 60s. Maximum value = 36000s.
Codec Pass-Through for SIP phones

Description This field enables or disables codec pass-through for SIP phones.
RTCP attribute in SDP (as of OmniPCX Office RCE R10.0)

This field covers the need for the declaration of an RTCP attribute to
document the RTCP port used in SDP (RFC3605).
When activated, an RTCP attribute is added by the PCX in all requests
Description containing SDP. When receiving requests containing this attribute, the PCX
interprets it and uses the specified port for RTCP flow.
This parameter is only available when VoIP relies on the SIP protocol. In all
other cases, this field is not displayed.
Value Yes or No. Default value = Yes (checked)
Nonce Caching
Description This field enables or disables Nonce caching for SIP phones.
Yes or No.
l By default, Nonce caching checkbox is checked (Enabled) in both online
(for R3.1) and offline OMC.
Value l When OMC 31.0 is connected to OXO Connect Release R3.0 this
checkbox is checked and greyed out.
l When OMC 31.0 is connected to OXO Connect Release < R3.0, this
checkbox is unavailable.
Law Mode
Description This field cannot be modified. It specifies the encoding law used.
A: for all countries except Asian countries
Values
Mu: for Asian countries
Default Framing
This droplist is used to define the default framing value associated to a codec.
Values: 20, 30, 40, 50, 60, 90, 120 (depending on the codec)
Dynamic Payload
This field allows the definition of the dynamic payload for DTMF used for SIP
trunk and IP sets. On IP trunks, DTMF is always exchanged according to the
DTMF RFC4733 standard (RTP with specific payload type).
The range value is between 96 to 122
The default value is 106 (101 for Orange Business Systems configurations)
This parameter is visible on this screen as of OXO Connect R4.0.
This field allows the definition of the dynamic payload for OPUS.
OPUS The range value is between 96 and 122.
The default value is 102.
The values used for DTMF and OPUS payload MUST be different. In case of overlap, the OMC displays an
error message at validation, when pressing the OK button.
VoIP parameters / Codecs
General description
As of OmniPCX Office RCE R9.1, this screen is used to define the default codecs list used in the
Codec/Framing field of the Automatic Routing: Prefixes table. It is relevant only when VoIP relies on the
SIP protocol. In all other cases, all fields are grayed and not modifiable.
Fields description
Available Codecs
This area contains the list of all available codecs.
Default Codecs List (renamed Selected Codecs in latest OMC versions)
This area contains the Default Codecs List, built using the sided arrows (==>,<==)
Default Framing (renamed Preferred Framing) in latest OMC versions)
This droplist is used to define the default framing value associated to a codec.
Values: 20, 30, 40, 50, 60, 90, 120 (depending on the codec)
NOTE: Whenever codec G723 is included in the list of Default Codecs, the Default framing selected must
be a multiple of 30.
Dynamic Payload
As of R10.0, two parameters allow to define the dynamic payload used for SIP trunk and IP sets for DTMF
and for G.722.2.
This field allows the definition of the dynamic payload for DTMF used for SIP trunk and IP
sets. On IP trunks, DTMF is always exchanged according to the RFC4733 standard (RTP with
DTMF specific payload type).
The range value is between 96 to 122
The default value is 106 (101 for Orange Business Systems configurations)
This field allows defining the dynamic payload for G.722.2 used for SIP trunk and IP sets. RTP
Payload used for G.722.2 is in the dynamic range specified by RFC3551.
G722.2
The range value is between 96 to 127.
The default value is 117.
The values used for G722.2 Dynamic payload and DTMF Dynamic payload MUST be different. In case of
overlap, the OMC displays an error message at validation, when pressing the OK button.
Buttons description
==>
This button is used to add an entry from the Available Codecs List to the Default Codecs List.
<==
This button is used to remove the selected codec from the Default Codecs List.
Factory Default (renamed Reset in latest OMC versions)
This button is used to return to the factory default list of codecs.
NOTE:
l For all targets except Orange Business Systems, the default list is: G729.a, G723.1, G722.2, G722,
G711.a, G711.µ. (with this priority order) and a framing of 30ms.
l For Orange Business Systems on the other hand, the default list is: G722, G711.a, G729.a (with this
priority order) and a framing of 20ms.
VoIP parameters / Topology
General description
As of OmniPCX Office RCE R10.0, this screen is used to define public data parameters, relevant only when
VoIP relies on the SIP protocol. In all other cases, all fields are grayed and not modifiable.
NOTE: To activate the Static NAT feature, Static NAT must also be enabled in the section ARS/Gateway
parameters (protocol Tab)
Fields description
Static NAT
IP address
Description This field defines the IP address for the public interface CPE router.
SIP/SIP TLS Port (UDP/TCP)
Description This field defines the UDP/TCP port number for the public interface CPE router.
Values By default, this port is set to 5060.
Range ports for RTP (UDP)
This field defines the UDP range of ports of the public interface CPE router used for RTP. A
continuous range of 256 ports must be indicated.
Description
To abide by this constraint, only the beginning of the range must be specified. The end of
the range is automatically adjusted.
Values The default ports range is [32000-32255]
Range ports for T38 (UDP)
This field defines the UDP range of ports range of the public interface CPE router used for
T.38. A continuous range of 96 ports must be indicated.
Description
To abide by this constraint, only the beginning of the range must be specified. The end of
the range is automatically adjusted.
Values The default ports range is [6666-6761]
NOTES:
l The values of the ports must be within the 1024-65535 port range
l The use of a port must be exclusive. In other words, there must be no overlap between the 3 types
of ports definition (SIP/RTP/T.38). In case of overlap, the OMC displays an error message at
validation, when pressing the OK button.
Buttons description
OK This button is used to validate your modifications
Cancel This button is used to cancel your modifications

Voice over IP OXO

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Voice over IP OXO

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VoIP parameters Page 1 of 19

OK This button is used to validate your modifications

Cancel This button is used to cancel your modifications

This button, available as of OXO Connect/OXO Connect Evolution R3.0, provides a

The number of external VoIP accesses depends on a software key.

VoIP parameters / General

l OXO Connect Evolution: the field is greyed and shows Multi-codecs

Number of VoIP Trunk Channels

Procedure File Configure Voice over IP (individual case) (Stage 3)

Number of VoIP Subscriber Channels

Number of VoIP Channels

VoIP Channels for trunks with reservation

VOIP Channels for IP phones and trunks without reservation

Number of Trunk Channels for trunks without reservation

Number of 'SIP Trunk channels' licenses

'SIP Trunk channels' licenses for trunks with reservation

'SIP Trunk channels' licenses for trunks without reservation

Codec Pass-Through for SIP phones

Codec Pass-Through for SIP trunks

This field enables or disables codec pass-through for SIP trunks.

G711 Codec for Music on Hold and preannouncements

l The "codec/framing" trunk gateway parameter is set to "default"

RTCP attribute in SDP (as of OmniPCX Office RCE R10.0)

VoIP parameters / Gateway

The default value of SIP Trunk Signal Source Port is 5060.

VoIP parameters / Gatekeeper

Identification of the Gatekeeper

VoIP parameters / DSP

Active voice detection

VoIP parameters / DHCP

VoIP parameters / Fax

Error Correction Mode

VoIP parameters / SIP trunk

The parameters "REGISTRATION", "AUTHENTICATION" and "LOCAL DOMAIN NAME" must be

Allows to set up the QoS value specific to SIP signalling.

SIP Trunk Signal Port (as of OXO Connect R3.0)

Enter the value of SIP Trunk signal source port.

Enter the value of SIP TLS Trunk signal port.

UDP to TCP Switching

This parameter offers the possibility to deactivate the transport switching of

Allows to set up the QoS value specific to SIP signalling.

SIP Trunk Signal Port

Enter the value of SIP Trunk signal source port.

SIP TLS Trunk Signal Port

Enter the value of SIP TLS Trunk signal port.

UDP to TCP Switching

This parameter offers the possibility to deactivate the transport switching of

Validate this option to enable the Registration feature.

Registered User Enter your Registered User Name.

Parameter is an IP address which is accessible when DNSSRV is disabled.

Username Enter your Username.

A Proxy server is declared in denial of service when it transmits more than a

Local Domain Name

RTP Proxy: Fixed ports

Security (as of OXO Connect R6.0)

VoIP parameters / SIP Phone

Select the default transport mode to be used by 8002/8012 DeskPhone, 8088

Description Enter the domain name of the SIP phones.

SIP Phone Signal Port (as of OmniPCX Office RCE R9.1)

Enter the value of SIP Trunk signal source port.

Codec Pass-Through for SIP phones

RTCP attribute in SDP (as of OmniPCX Office RCE R10.0)

This parameter is visible on this screen as of OXO Connect R4.0.

VoIP parameters / Codecs