Panasonic KX-NCP500 User manual

Thank you for purchasing a Panasonic Pure IP-PBX.
Please read this manual carefully before using this product and save this manual for future use.
KX-NCP500/KX-NCP1000: PBMPR Software File Version 2.0200 or later
Pure IP-PBX
Document Version: 2009-12
IP Networking Guide
Model No. KX-NCP500
KX-NCP1000
3DQDVRQLF7HOHSKRQH6\VWHPV
www.voicesonic.com
Phone [phone]
Panasonic KX-NCP500 KXNCP500 KX NCP500 KX-NCP1000 KXNCP1000 KX NCP1000

Table of Contents
1Introduction ..............................................................................................3
1.1 Overview ............................................................................................................................4
1.1.1 Establishing a VoIP Network with the Pure IP-PBX .........................................................4
1.2 Network Management .......................................................................................................7
1.2.1 DHCP (Dynamic Host Configuration Protocol) Server .....................................................7
1.2.2 VLAN (Virtual LAN) ..........................................................................................................8
1.2.3 Gatekeeper .....................................................................................................................10
1.3 Packet Control Features .................................................................................................11
1.3.1 Jitter Buffer .....................................................................................................................11
1.3.2 Voice Activity Detection (VAD) .......................................................................................11
2 Guidance for VoIP Installation ..............................................................13
2.1 VoIP Requirements .........................................................................................................14
2.1.1 Bandwidth Assessment ..................................................................................................14
2.1.2 Network Configuration ....................................................................................................17
2.1.3 Network Devices ............................................................................................................20
2.1.4 QoS (Quality of Service) .................................................................................................22
2.2 VoIP Requirements Checklist ........................................................................................23
3 Connection to the LAN ..........................................................................25
3.1 Connecting the IPCMPR Card to the LAN .....................................................................26
3.2 Connecting the IP Telephones .......................................................................................28
4 Programming ..........................................................................................31
4.1 Programming the PBX ....................................................................................................32
4.1.1 Installing the Virtual IP Cards to the PBX .......................................................................32
4.1.2 Installing Additional Activation Keys ...............................................................................33
4.2 Programming the Virtual 16-Channel VoIP Gateway Card ..........................................36
4.2.1 Assigning the Hunt Pattern .............................................................................................36
4.2.2 Programming the Address Translation Table .................................................................37
4.2.3 Programming the Network Settings ................................................................................39
4.3 Programming the IPCMPR Card ....................................................................................44
4.3.1 Assigning the IP Addressing Information .......................................................................44
4.4 Programming the IP Telephones ...................................................................................47
4.4.1 Assigning the IP Addressing Information .......................................................................47
4.4.2 Setting the VLAN Parameters ........................................................................................64
4.4.3 Setting the Diffserv Parameters .....................................................................................67
4.4.4 Configuration of IP Ports ................................................................................................70
4.5 Registering IP Telephones .............................................................................................77
4.5.1 Registering IP Telephones .............................................................................................77
4.5.2 De-registering IP Telephones .........................................................................................88
5 Appendix .................................................................................................91
5.1 Revision History ..............................................................................................................92
5.1.1 KX-NCP500/KX-NCP1000 PBMPR Software File Version 1.01xx .................................92
5.1.2 KX-NCP500/KX-NCP1000 PBMPR Software File Version 2.02xx .................................93
5.2 Error Message .................................................................................................................94
2 IP Networking Guide Document Version 2009-12
Table of Contents

Section 1
Introduction
Document Version 2009-12 IP Networking Guide 3

1.1 Overview
1.1.1 Establishing a VoIP Network with the Pure IP-PBX
Panasonic KX-NCP500/KX-NCP1000 Pure IP-PBX supports Panasonic KX-NT series IP proprietary
telephones (IP-PTs), Panasonic IP softphones, and SIP (Session Initiation Protocol) Extensions (hardphones
and softphones) for communication on a Voice over Internet Protocol (VoIP) network. These IP telephones
can be used as extensions of the PBX when the local office LAN is connected to other LANs at different
locations.
The Pure IP-PBX also enables VoIP communication with PBXs installed at different locations. Since the
communication does not take place over conventional telephone network, the high cost of long distance
communication is virtually eliminated.
The following diagrams illustrate VoIP network with (i) a remote office LAN and (ii) another PBX installed at
different location.
Note
Panasonic IP Cell Station (IP-CS) units are also supported by the Pure IP-PBX for communication on a
VoIP network. For details, refer to the Quick Installation Guide for the IP-CS.
(i) VoIP Network with Remote Office LAN
DHCP Server
SIP Extension
Extn. 102
IP-PT
Extn. 101
DHCP Server
Remote Office LAN
Local Office LAN
Router
Switching
Hub
PBX
IP Softphone
Etxn. 103
IP Network
IP-PT
Extn. 104 Switching
Hub
IP Softphone
Extn. 105
Router
SIP Extension
Extn. 106
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1.1.1 Establishing a VoIP Network with the Pure IP-PBX

(ii) VoIP Network with Another PBX
IP Network
Router
Switching Hub
Router
Switching Hub
PBX
PBX
BranchBranch
HeadquartersHeadquarters
Document Version 2009-12 IP Networking Guide 5
1.1.1 Establishing a VoIP Network with the Pure IP-PBX

Network Parameters
You will need to have the following IP addressing and QoS information to establish VoIP communication on
your network. This information is typically supplied by a network administrator.
Consult your network administrator for specific values.
Parameter Description
IP telephone IP Address Identifies the location of IP telephones on the network. Each IP
telephone must have a unique IP address.
Subnet Mask Address
Defines which digits of an IP address are used for the network
address and the host address at each network location. The IP
addresses of the IP telephones and the PBX must fall within the same
subnet as that of the default gateway (e.g., router) of the LAN.
Default Gateway Address
Identifies the IP address of the primary gateway (typically a router or
similar device) that exchanges IP packets with the other gateways on
the VoIP network.
PBX IP Address Identifies the location of the PBX in the network during VoIP
communications.
VLAN ID
Identifies the ID of the logical segment within the corporate LAN,
through which voice packets from IP telephones travel. For details,
refer to "1.2.2 VLAN (Virtual LAN)".
DiffServ (DS) Identifies the value for the DS field in the header of IP packets, which
determines the priority given to packets traveling from IP telephones.
For details, refer to "4.4.3 Setting the Diffserv Parameters".
Types of IP Network
The speech quality depends on the type of IP network in use. Managed IP networks provide better speech
quality compared to unmanaged networks such as the Internet, where quality of service cannot be guaranteed.
Examples of recommended IP networks
•Digital Leased Line
•IP-VPN (Virtual Private Network)
•Frame Relay
Not recommended
•Internet (including an Internet VPN)
Note
•Peer-to-peer calls between KX-NT300 series IP-PTs installed at different locations may not be possible
if packet communication cannot be established between the respective networks.
In this case, you need to configure the network settings (e.g., a VPN router when using an IP-VPN) to
establish packet communication.
•Unlike an IP-VPN, which is set up over a network provider’s own IP network, an Internet VPN is set up
over the Internet. Internet VPNs are not recommended for VoIP communication because transmission
delays and loss of data are likely to occur.
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1.1.1 Establishing a VoIP Network with the Pure IP-PBX

1.2 Network Management
1.2.1 DHCP (Dynamic Host Configuration Protocol) Server
To establish communication over a VoIP network, IP addresses must be assigned to IP telephones and the
PBX to identify their locations on the network. While these addresses can be assigned manually, it is also
possible to use a DHCP server to automatically assign IP address information. Using a DHCP server allows
you to centrally manage and automate the assignment of IP addresses.
Assign IP address
information to an IP
telephone
When the IPCMPR/DSP cards are
connected to the network
When an IP telephone is
connected to the network
Request IP address
information
1
Assign IP address
information to the
IPCMPR/DSP cards
2
Request IP address
information
3
4
IP Telephone
DHCP Server
IPCMPR/DSP Cards
Note
•The PBX is not able to act as a DHCP server. To use the DHCP client function of IP telephones and
the IPCMPR/DSP cards, a separate DHCP server is required on the network, as shown above.
•An IP telephone and the IPCMPR/DSP cards cannot request IP addresses from a DHCP server on
another LAN (connected through an IP network). They can only receive IP addresses from a DHCP
server on the same LAN. Therefore, when IP telephones are located on several LANs, a DHCP server
is required on each LAN. If a DHCP server is not present on the LAN, IP addresses for IP telephones
and the IPCMPR/DSP cards on that LAN must be assigned manually.
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1.2.1 DHCP (Dynamic Host Configuration Protocol) Server

1.2.2 VLAN (Virtual LAN)
VLANs are logical segments within a corporate LAN. By assigning VLAN settings to IP telephones, it is possible
to separate the packets transmitted by an IP telephone according to the type of data and specify which VLAN
each data type will be sent over. This allows you to avoid generating unnecessary network traffic on each
segment and to reduce the load on the network. As a consequence, speech quality can be assured. Therefore,
we recommend using the VLAN feature to perform VoIP communication effectively.
Some IP telephones (e.g., KX-NT300 series) are equipped with 2 ports, primary and secondary, for packet
communication. Allocating these ports to different VLANs enables you to split the paths for packets depending
on whether the packet contains voice signals or data.
VLAN settings (VLAN ID and VLAN priority) for the primary port affect voice data transmitted by the IP
telephone, whereas VLAN settings for the secondary port apply to data transmitted by a PC connected to the
IP telephone. When sending packets, the IP telephone can attach information on which VLAN the packets are
to be transmitted over (VLAN Tagging). The switching hub that receives these packets reads the VLAN
information and sends the packets over the appropriate VLAN. This helps to ensure bandwidth for IP telephone
voice transmissions.
In this way, an IP telephone with 2 ports can transmit voice packets from the primary port with higher priority
than other packets from the secondary port.
PBX
IP Telephone with 2 ports
VLAN 1VLAN 1
VLAN 1
VLAN 2
VLAN 2
VLAN 1
Segment for Voice Data
VLAN 2
Segment for Other Data
Voice Packet
Other Packet
PC
PC
Other Packet
VLAN 2
VLAN-capable Switching Hub
Primary Secondary
Note
•This VLAN feature complies with IEEE (Institute of Electrical and Electronics Engineers) 802.1Q.
•The PBX receives VLAN settings only from the connected switching hub. Therefore, VLAN settings for
the PBX must be assigned at the switching hub.
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1.2.2 VLAN (Virtual LAN)

•Some PC LAN cards allow VLAN settings to be assigned. However, when using a PC connected to an
IP telephone with 2 ports, the VLAN settings for PC communications must be assigned only to the
secondary port of the IP telephone. Any VLAN settings assigned to the PC LAN card must be disabled.
These settings can usually be identified by "802.1Q", "802.1p", or "VLAN" in their name.
•If you are using an IP telephone with a primary port only (e.g., KX-NT265*1), a PC cannot be connected
to the IP telephone.
*1 Only KX-NT265 IP-PTs with a software version of 2.00 or later are supported by KX-NCP500/KX-NCP1000.
Document Version 2009-12 IP Networking Guide 9
1.2.2 VLAN (Virtual LAN)

1.2.3 Gatekeeper
The following are the general functions of a gatekeeper:
•Dialed number-to-IP address translation
•Authentication
•Bandwidth control
The gatekeeper provides these network management functions to registered clients. To register with the
gatekeeper, you need to configure the V-IPGW16 card to use the gatekeeper and program the GK Settings
table through system programming. For details, refer to "3.9 [1-1] Slot—Shelf Property - Virtual IP Gateway
—Gatekeeper Available" and "3.10 [1-1] Slot—Shelf Property - Virtual IP Gateway—GK Settings" in the
PC Programming Manual. After programming, the V-IPGW16 card attempts to register with the gatekeeper
using registration information such as the IP address of the IPCMPR card, and destination telephone numbers
specified in the GK Settings table.
Note
•For more information about gatekeeper functions, consult the documentation of the gatekeeper.
•When using a gatekeeper, make sure to choose a compatible model. For more information about
gatekeeper compatibility with the V-IPGW16 card, consult a certified dealer.
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1.2.3 Gatekeeper

1.3 Packet Control Features
1.3.1 Jitter Buffer
When voice signals are packetized and transmitted, individual packets can take different paths through the
network and arrive at the destination at varied timings. This is referred to as "jitter", and it can cause degradation
in speech quality. To compensate for jitter problems, the "jitter buffer" accumulates the packets temporarily for
processing.
To set the size of the jitter buffer, refer to "3.4 [1-1] Slot—Card Property - IPCMPR—VoIP-DSP Option" in the
PC Programming Manual.
1.3.2 Voice Activity Detection (VAD)
The VAD conserves bandwidth by detecting silent periods during a call and suppressing the packets of silence
from being sent to the network. This feature can be enabled or disabled for each available codec: G.711 and
G.729A.
To configure the VAD feature, refer to the appropriate section in the PC Programming Manual.
Note
•To use the VAD feature for a certain codec, be sure to enable it for that codec on both the local and
remote gateway devices.
•When connecting the PBX to a KX-TDE or KX-TDA series PBX:
–The VAD feature cannot be used between the V-IPGW16 and IP-GW4 cards since the
V-IPGW16 card does not support the G.723 codec (although calls can be made and received as
normal).
–The VAD feature between the V-IPGW16 and IP-GW16 cards can be enabled through system
programming. Refer to "Main— Connection for IP-GW16" under "3.11 [1-1] Slot—Shelf Property
- Virtual IP Gateway—GW Settings" in the PC Programming Manual.
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1.3.2 Voice Activity Detection (VAD)

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1.3.2 Voice Activity Detection (VAD)

Section 2
Guidance for VoIP Installation
Document Version 2009-12 IP Networking Guide 13

2.1 VoIP Requirements
2.1.1 Bandwidth Assessment
When using the IP telephones and V-IPGW16 card, you must ensure that the IP network in use has enough
bandwidth to support VoIP communications. If the amount of bandwidth required for VoIP communications is
more than the network can accommodate, speech quality will be compromised. In addition, there may be an
adverse effect on the performance of other applications (e.g., email or web applications) that use the same
network. Therefore, care must be taken when assessing bandwidth requirements.
Inform your network administrator of the required bandwidth, and make sure that the network can support VoIP
communications even under conditions of maximum network traffic.
Bandwidth Assessment for Virtual IP Extension Card
Required Bandwidth per IP telephone for a Call
The required bandwidth depends on what combination of codecs and packet sending intervals is used. Keep
in mind the following points about the type of codecs and packet sending intervals, in terms of speech quality:
•The speech quality of the codecs varies as follows: (High) G.722, G.711, G.729A (Low)*1
•The shorter the packet sending interval, the higher the speech quality.
•The higher the speech quality the IP telephones provide, the more bandwidth the IP telephones require.
*1 When the preferred codec of each party differs, the call will be established using the lower codec. For example, if the caller prefers
G.711 while the called party prefers G.729A, the call will be established using G.729A.
Codec
Packet Sending Interval
20 ms 30 ms 40 ms 60 ms
G.722*1/G.711 87.2 kbps 79.5 kbps — —
G.729A 31.2 kbps 23.5 kbps 19.6 kbps 15.7 kbps
*1 G.722 is only available for the KX-NT300 series IP-PTs and some SIP Extensions that support this codec during peer-to-peer
communication. For details, refer to "13.1.8 Peer-to-Peer Connection" in the Feature Manual.
Required Bandwidth for Each Virtual IP Extension Card
To allow all IP telephones to make calls simultaneously, it is necessary to keep available the bandwidth required
by a virtual IP Extension card with the maximum number of IP telephones connected.
Provided below is the formula to calculate the amount of bandwidth required for each virtual IP Extension card.
Required Bandwidth = (Required Bandwidth per IP telephone ´ 32)
Bandwidth Assessment for V-IPGW16 Card
Required Bandwidth for One VoIP Channel
The required bandwidth depends on what combination of codecs and packet sending intervals is used. Keep
in mind the following points about the type of codec and packet sending interval, in terms of the speech quality:
•The speech quality of the G.711 codec is higher than that of the G.729A codec.
•The shorter the packet sending interval, the higher the speech quality.
•The higher the speech quality the V-IPGW16 card provides, the more bandwidth the card requires.
14 IP Networking Guide Document Version 2009-12
2.1.1 Bandwidth Assessment

Via LAN
Codec
Packet Sending Interval
20 ms 30 ms 40 ms 60 ms 90 ms
G.711 87.2 kbps 79.5 kbps 75.6 kbps 71.7 kbps —
G.729A 31.2 kbps 23.5 kbps 19.6 kbps 15.7 kbps —
Via WAN (PPP: Point-to-Point Protocol)
Codec
Packet Sending Interval
20 ms 30 ms 40 ms 60 ms 90 ms
G.711 84 kbps 77.3 kbps 74 kbps 70.7 kbps —
G.729A 28 kbps 21 kbps 18 kbps 14.7 kbps —
Bandwidth Calculation
Provided below is the formula to find out the amount of bandwidth required for VoIP communications:
Required Bandwidth
= (No. of Fax Machines ´ Required Bandwidth for the G.711 codec) +
[(16 - No. of Fax Machines) ´ Required Bandwidth for Voice Communication]
Example
Consider the following case as an example:
•Communication: via LAN
•No. of Fax Machines: 2
•G.711 Packet Sending Interval: 20 ms (requiring 87.2 kbps per channel)
•G.729A Packet Sending Interval for Voice Communication: 20 ms (requiring 31.2 kbps per channel)
In this case, the required bandwidth will be as follows:
Required Bandwidth
= (2 ´ 87.2) + [(16 - 2) ´ 31.2]
= 611.2 (kbps)
Therefore, inform your network administrator and make sure that the network can support a bandwidth of 611.2
kbps even when the network is under conditions of maximum traffic.
Note
It is recommended that all cards on a VoIP network have the same packet sending interval.
Additional Information
As described above, it is possible to control the required bandwidth by selecting a certain combination of codec
and packet sending interval. However, it is also possible to control required bandwidth by limiting the number
of available virtual VoIP channels.
The V-IPGW16 card supports a total of 8 ports, each having 2 separate channels. By disabling some of the
ports, you can reduce the bandwidth required for VoIP communications.
To limit the number of VoIP channels:
Set the status of the ports you wish to disable (starting from the highest-numbered port) to OUS.
Document Version 2009-12 IP Networking Guide 15
2.1.1 Bandwidth Assessment

For example, if you wish to use only 10 of the available 16 virtual VoIP channels (i.e., disable 6 channels), set
ports 8, 7, and 6 to OUS as shown below:
In this case, the equation for bandwidth calculation, based on the previous example, will change as follows:
Required Bandwidth
= (No. of Fax Machines ´ Required Bandwidth for the G.711 codec) +
[(10 - No. of Fax Machines) ´ Required Bandwidth for Voice Communication]
= (2 ´ 87.2) + [(10 - 2) ´ 31.2]
= 424 (kbps)
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2.1.1 Bandwidth Assessment

2.1.2 Network Configuration
You must evaluate the structure of the existing network to see if a VoIP network can be implemented. Below
are the points that should be evaluated.
Is the IP network a managed network?
A VoIP network should be implemented on a managed IP network such as Frame Relay, Leased Line, or
IP-VPN (Virtual Private Network).
An unmanaged network, such as the Internet (including an Internet VPN), cannot be used to employ a VoIP
network because delays and loss in data transmission can cause huge degradation in speech quality.
Is it possible to have static IP addressing?
IP telephones on the network always perform VoIP communications through the PBX. Therefore, the PBX
must be assigned static IP addresses, which must be programmed to each IP telephone on the network.
Note
When a DHCP server (which automates IP addressing of devices on the network) is not used, static IP
addressing must also be enabled for all IP telephones.
Does only a single router provide access to the IP network?
In a dual network, 2 routers provide access to the IP network as shown in the diagram below. However, only
one router can be used as an access point to the network.
Therefore, in the diagram below, if router A, whose IP address is assigned as the default gateway IP address
of the PBX and the IP telephones, fails, VoIP communications are no longer possible; they are not able to
switch their default gateway from router A to router B to access the IP network.
IP Network
Router A Router B
Default gateway of
the PBX: Router A
Default gateway of
the IP Telephone:
Router A
Document Version 2009-12 IP Networking Guide 17
2.1.2 Network Configuration

Does the router not use network address translation (NAT/NAPT)?
If the router uses address translation techniques (e.g., NAT/NAPT) to convert between global and local IP
addresses, VoIP communications cannot be carried out effectively. Therefore, the routers used to access the
IP network must not use NAT/NAPT. Generally, NAT and NAPT are features that are available with routers.
Router Router
Router Router
Global IP Address
Domain
Local IP Address
Domain
Local IP Address
Domain
Local IP Address
Domain
Using NAT/NAPT NOT Using NAT/NAPT
IP Network IP Network
18 IP Networking Guide Document Version 2009-12
2.1.2 Network Configuration

Is there only a single IP network between 2 ends of a call?
A huge degradation in speech quality will be produced when calls are made through multiple IP networks as
shown below; therefore, it is recommended that you avoid establishing a VoIP network in this fashion.
IP Network 1 IP Network 2
PSTN/
PRI QSIG, etc.
IP Network 1 IP Network 2
Document Version 2009-12 IP Networking Guide 19
2.1.2 Network Configuration

Are the network devices located appropriately for effective VoIP
communications?
Transmission delays can cause pauses and loss in VoIP communications. The more network devices (e.g.,
routers and switching hubs) there are between the PBX and IP telephones or the IP network interface, the
longer the transmission delays. This is because a certain amount of delay is inevitable when packets go through
each network device.
To prevent unnecessary delays, it is recommended to connect the PBX as close to the IP telephones and the
IP network interface as possible so that the number of the network devices is kept to a minimum.
IP Network
RouterRouter
Switching
Hub
Switching
Hub
Switching
Hub
2.1.3 Network Devices
You must evaluate the network devices that are used in the existing network to see if a VoIP network can be
implemented. Below are the points that should be evaluated.
Can the firewall pass packets appropriately?
If the VoIP network contains a firewall, the firewall must be configured appropriately to allow VoIP packets,
listed in the table below, to pass through the network without being blocked by filtering.
For more information, consult your network administrator.
[IP Packets from IPCMPR Card and IP Telephones]
Protocol Description TCP/UDP Default Port No.
RTP (IP telephone) Real-time Transport Protocol.
Used for voice data transmission.
UDP 12000 to 12255
Maintenance (IPCMPR) Panasonic proprietary protocol.
Used for communication parameter
negotiation with the PBX, download of
country/area data, confirmation of
connection with the PBX, and
notification of error messages and
statistical information to the PBX.
TCP 35300
Maintenance (IP
telephone)
UDP 9301
MGCP (IPCMPR) Media Gateway Control Protocol.
Used for call control command data and
LCD/LED data transmission.
UDP 2727
MGCP (IP telephone) UDP 2427
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2.1.3 Network Devices