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2 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
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IP Telephony is the technology of transmitting voice conversations over a data
network infrastructure using IP (Internet Protocol). IP Telephony is the ability
to make a phone call using an IP based device, optionally via gateways such
as the Business Communications Manager or using Internet Telephony
Service Providers (ITSPs). This convergence of voice, video, and data
enhances our ability to collaborate with tools such as video conferencing and
other data related facilities.
Business Communications Manager (BCM) with Voice over IP (VoIP) provides
several business critical advantages:
Cost Savings. IP networks can be significantly less expensive to
operate and maintain than traditional networks. The simplified network
infrastructure of an Internet Telephony solution cuts costs by
connecting IP telephones over your LAN and eliminates the need for
dual cabling. IP Telephony can also provide “internal” dialling capability
on site-to-site calls via global four-digit dialling plans.
Portability and flexibility. Employees can be more productive
because they are no longer confined by geographic location. IP
telephones work anywhere on the network, even over a remote
connection. Network deployments and reconfigurations are simplified,
and service can be extended to remote sites and home offices over
cost-effective IP links.
Simplicity and consistency. Customers can centrally manage the IP
Telephony infrastructure from a central point via the Element Manager
application. The ability to network existing PBXs using IP can bring new
benefits to a business. For example, the ability to consolidate voicemail
onto a single system, or to fewer systems, making it easier for voice
mail users to network.
Compatibility. IP Telephony is supported over a wide variety of
transport technologies. A user can gain access to just about any
business system through a Digital Line, a LAN, frame relay,
asynchronous transfer mode, SONET or wireless connection.
Scalability. A future-proof, flexible, and safe solution, combined with
high reliability, allows a company to focus on customer needs, not
network problems.
NN40011-028 Issue 1.2 BCM Rls 6.0 7
IP Telephony
Note: All IP Clients require licence seats enabling on the BCM to allow
registration and functionality. The 2050 IP Softphone requires additional per
seat licensing, as does the 1100 series VPN feature. The Remote Worker
Solution (NAT traversal) also requires licensing, on a system-wide rather than
per seat basis.
IP Telephones and VoIP Trunks
This guide describes two similar applications for IP telephony on the BCM
system: IP telephones and VoIP trunks. These applications can be used
separately or together as a network voice/data solution.
IP Telephones
IP telephones offer the functionality of regular telephones, but do not require a
hardwire connection to the BCM. Instead, they must be plugged into an IP
network that is connected to the LAN or WAN card (BCM50(b)e only) on the
BCM.
Calls made from IP telephones through the BCM can pass over VoIP trunks or
across a Public Switched Telephone Network (PSTN).
Avaya provides a range of IP telephones. The „i-series‟ telephones are
hardwired to the system, in the case of the i2001, i2002, i2004, i2007 as well
as the newer 1110, 1120E, 1140E, 1210, 1220, 1230 and the i2033 IP
conference phone, or are accessed through your desktop or laptop computer
as in the case of the IP Softphone 2050.
VoIP Trunks
VoIP trunks (Lines) allow voice signals to travel across IP networks. A
gateway within the BCM converts the voice signal into IP packets, which are
then transmitted through the IP network. The device at the other end
reassembles the packets into a voice signal. NetMeeting is one of the H.323
protocol trunk devices that the BCM system supports.
H.323 is a standard for packet based multimedia communications systems.
H.323 is widely used as the standard for IP telephony and allows for the voice
packets to traverse an IP network. It was designed for multimedia
communication over IP networks, including audio, video, and data
conferencing. The most widely deployed use of H.323 is "Voice over IP"
followed by "Videoconferencing".
SIP Session Initiation Protocol is text based application-layer control
(signaling) protocol for creating, modifying, and terminating sessions with one
or more participants. It can be used to create two or multiparty VoIP telephone
calls. Name Translation and User Location is utilised where SIP translates an
address to a name and thus reaches the called party at any location.
8 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
Note: VoIP trunks are enabled via keycodes. The number of licence seats
applied determines the maximum number of simultaneous calls via VoIP
trunks.
Supporting Information
The following sections contain information the might be useful when
considering network design and integration of BCM VoIP functionality into the
network.
SIP Trunk Authentication
Ensures that only gateways with valid credentials can place calls to the BCM
and that BCM can provide valid credentials on outgoing calls when challenges
take place.
DNS (Domain Name Service)
DNS can be used to locate SIP servers. This means that customers do not
need to know the IP addresses of remote servers and can use domain name
entries instead.
SIP Proxy Failover
Enables use of multiple SIP Proxies without relying on DNS query method
with multiple entries.
SIP REFER
Standards based method for handling incoming SIP REFER messages to
support Call Transfer requests in a SIP network environment.
G.711 Fax Support
Option to use G.711 when placing calls from fax machines.
IP Network
The network administrator should be able to advise you about the network
setup and how the BCM fits into the network.
WAN
A Wide Area Network (WAN) is a communications network that covers a wide
geographic area, such as a state or country. If you want to deploy IP
telephones that will be connected to a LAN outside of the LAN that the BCM is
installed on, you must ensure the BCM has access to a network device that
has a WAN connection. This includes ensuring that you obtain IP addresses
and routing information that allows the remote telephones to find the BCM,
and vice versa.
LAN
A Local Area Network (LAN) is a communications network that serves users
within a confined geographical area. For BCM, a LAN is any IP network
NN40011-028 Issue 1.2 BCM Rls 6.0 9
IP Telephony
connected to a LAN Interface on the BCM system. Often, the LAN can include
a router that forms a connection to the Internet.
Public Switched Telephone Network
The PSTN can play an important role in IP telephony communications. In
many installations, the PSTN forms a fallback route. If a call across a VoIP
trunk does not have adequate voice quality, the call can be routed across the
PSTN instead, either on public lines or on a dedicated ISDN connection
between the two systems. The BCM also serves as a gateway to the PSTN
for all voice traffic on the system.
Gatekeeper
A gatekeeper tracks IP addresses of specified devices, and provides
authorisation for making and accepting calls for these devices. A gatekeeper
is not required for the BCM system, but can be useful on networks with a
large number of devices.
A gatekeeper controls all H.323 clients (endpoints like MS Netmeeting) in its
zone. Its primary function is to address translation between alias addresses
and IP addresses. This way you can call "Fred" instead of knowing which IP
address he currently works on. VoIP gateways can register at the
gatekeeper and the gatekeeper finds the right gateway to use to call a
specific number.
For example in the diagram below digital telephone A wants to call IP
telephone B, which is attached to BCM B, over a network that is under the
control of a gatekeeper. Digital telephone A sends a request to the
gatekeeper. The gatekeeper provides Digital telephone A with the
information it needs to contact BCM B over the network. BCM B then passes
the call to IP telephone B.
10 NN40011-028 Issue 1.2 BCM Rls 6.0
Below is a diagram showing an example of a VoIP Network.
IP Telephony
Key IP Telephony Concepts
In traditional telephony, the voice path between two telephones is circuit
switched. This means that the digital connection between the two telephones
is dedicated to the call. The voice quality is usually excellent, since there is no
other signal to interfere.
In IP telephony, voice quality between IP telephones can vary significantly
from call to call and time of day. When two IP telephones are on a call, each
IP telephone encodes the speech at the handset microphone into small data
packets called frames. The system sends the frames across the IP network to
the other telephone, where the frames are decoded and played at the handset
receiver. If some of the frames get lost while in transit, or are delayed too
long, the receiving telephone experiences poor voice quality.
Codecs
The algorithm used to compress and decompress voice is embedded in a
software entity called a codec (COde-DECode). Two popular Codecs are
G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per second
(kbps) while G.729 samples at a far lower rate of 8 kbps. Voice quality is
NN40011-028 Issue 1.2 BCM Rls 6.0 11
IP Telephony
better when using a G.711 CODEC, but more network bandwidth is used to
exchange the voice frames between the telephones.
If you experience poor voice quality, and suspect it is due to heavy network
traffic, you can get better voice quality by configuring the IP telephone to use
a G.729 CODEC.
The BCM supports these codecs:
G.729
G.723
G.729 with VAD (Voice Activity Detection - the transmission of "silent
packets" over the network)
G.723 with VAD
G.711-uLaw
G.711-aLaw
BCM allows for CODEC renegotiation. This means that two sets and/or
trunks using dissimilar CODEC settings, when initiating the VoIP call, would
negotiate and decide which CODEC to use. In earlier BCM software levels,
differing CODECS would have meant that the call would be dropped.
Jitter Buffer
Voice frames are transmitted at a fixed rate, because the time interval
between frames is constant. If the frames arrive at the other end at the same
rate, voice quality is perceived as good. In many cases, however, some
frames can arrive slightly faster or slower than the other frames. This is called
jitter, and degrades the perceived voice quality. To minimize this problem,
configure the IP telephone with a jitter buffer for arriving frames.
This is how the jitter buffer works - Assume a jitter buffer setting of five
frames:
The IP telephone firmware places the first five arriving frames in the
jitter buffer.
When frame six arrives, the IP telephone firmware places it in the
buffer, and sends frame one to the handset speaker.
When frame seven arrives, the IP telephone buffers it, and sends frame
two to the handset speaker.
The net effect of using a jitter buffer is that the arriving packets are
delayed slightly in order to ensure a constant rate of arriving frames at
the handset speaker.
12 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
The below diagram shows a Jitter Buffer example assuming a jitter buffer
setting of five frames:
Possible jitter buffer settings and corresponding voice packet latency (delay)
for the BCM system IP telephones are:
None
Small (G.711/G.729: 0.05 seconds)
Medium (G.711/G.729: 0.09 seconds)
Large (G.711/G.729: 0.15 seconds)
NN40011-028 Issue 1.2 BCM Rls 6.0 13
IP Telephony
QoS Routing
The process of prioritizing data frames is referred to as Quality of Service
(QoS) routing.
The BCM system supports QoS routing, when it is integrated with other Avaya
routing solutions. The BCM system can also be configured to monitor QoS so
that the system reverts to a circuit-switched line if a suitable QoS cannot be
guaranteed.
VoIP packets can also be “marked” using DSCP, with the aim of prioritising
these packets through the network.
Remote Working Capability
The latest release of BCM offers the option of being able to use an IP
Telephone in remote locations, as it were a phone on the local system. The
Remote Worker solution only requires standard routers and networking
capability to perform this function. If necessary, the IP telephone can be
moved to various locations as required, as long as there is network access to
the BCM.
14 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
A typical example of the Remote Worker solution would be a home worker
who wishes to connect an IP telephone to the main office BCM, using their
standard home router and the internet. The office BCM would be connected to
the internet via a router which has a static public IP address, and forwards the
IP telephone‟s data/voice traffic to the BCM (and vice-versa).
Alternatively, if extra security is required for the data/voice traffic, a VPN
connection can be initiated via the 1120 and 1140 IP telephones. This
requires enhanced IP phone configuration, and a VPN router at the main
office hosting the BCM.
Required Information
Before configuring IP Telephony, the following information will need to be
confirmed:
Which interface will be used for the Published IP address?
Is there a Gatekeeper connected to the BCM, if so, what is the IP
address of the Gatekeeper and the Alias name for the BCM?
If there is no Gatekeeper, what are the IP addresses of the remote
Gateways and what are the telephony destination digits required to dial
those systems?
What password will be used for IP Phone registration?
Are there any routers that should be referenced as part of the VoIP
configuration? These may be used to provide WAN access for
example.
If using the Remote Worker or 1100 series VPN solutions, what is the
public IP address of the router connecting the BCM to the
Internet/WAN network.
What telephony configuration is required for IP Telephony?
Will DHCP be required for the IP Phones, and if so, will the BCM be set
up to provide IP Addresses to the phones?
NN40011-028 Issue 1.2 BCM Rls 6.0 15
IP Telephony
Will the BCM be used to issue IP
Addresses to the IP phones?
Ensure that the required keycodes are applied to the
BCM: refer to the Keycodes section of this guide.
Set the BCM‟s IP Address that the IP phones will
register against: refer to the Published IP Interface
section of this guide.
Refer to the DHCP Configuration section
of this guide.
Register the IP phones: refer to the Registering the IP Phones to the System section of this guide.
Will the 2050 IP Softphone be used?
Refer to the 2050 IP Softphone
section of this guide.
IP Phones have been
configured for use.
Yes
No
Yes
No
Set the BCM up to allow IP phones to register:: refer to
the Preparing Your system for IP Telephone
Registration section of this guide.
Flow Charts
Use the following flow charts to determine which sections of this guide to use.
IP Telephone Configuration
16 NN40011-028 Issue 1.2 BCM Rls 6.0
VoIP Gateway Configuration
Will SIP be used over the VoIP
trunks?
Determine how incoming and outgoing calls will be
handled: refer to the Configuring the Local Gateway Settings section of this guide.
Check the H323 and/or SIP Media Parameters: refer to
the H323 & SIP Media Parameters section of this
guide.
Refer to the Private SIP Specific Configuration
section of this guide.
If not using a Gatekeeper on the network, manually
configure the Remote Gateways: refer to the Remote Gateways (Routing Table) section of this guide.
Yes
No
Will the SIP trunks be private to
another system, or public to an ITSP?
Refer to the Public SIP Trunk Configuration
section of this guide.
Private
Public to ITSP
IP Telephony
NN40011-028 Issue 1.2 BCM Rls 6.0 17
IP Telephony
General Configuration
The BCM supports the following IP telephony protocols: UNISTIM, H.323 and
SIP.
The IP telephones use UNISTIM.
The Symbol NetVision and NetVision Data telephones use H.323+.
VoIP Trunks can use either H.323 or SIP (defined on a per gateway
basis)
The applications that control these protocols on the BCM provide an invisible
interface between the IP telephones and the digital voice processing controls
on the BCM.
Keycodes
The first part of configuration for IP Telephony is ensuring that the required
keycodes have been purchased and are entered.
1. In Element Manager, select the Configuration tab and then open the
System folder. Select the Keycodes link and the keycodes that have
been entered will be displayed.
2. Three keycode types are available, depending on your requirements:
VoIP (H.323) or SIP GW Trunks: two trunk protocols for networking
between compatible telephone systems. The number of trunk
licence seats enables determines the maximum number of VoIP
calls that can be placed over VoIP trunks. SIP GW trunks will be
required to connect to ITSPs.
IP Clients: The number of IP Client licence seats determines the
number of IP Phones and Software IP Phones that can be
registered against the BCM.
18 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
Note: The exception to this rule is when registering telephones to be used
Remote Worker sets. Please refer to the Remote Worker Solution section of
this guide for instructions on S1/S2 assignment for this feature.
Note: The Published IP Address is the address that LAN CTE should also
register against. For further information, refer to the LAN CTE Guide.
Remote Worker: A single keycode unlocks the Remote Worker
solution
Published IP Interface
The Published IP Interface is the IP Address that IP Telephones need to
register against as well as the address that VoIP gateways need to be
“pointed” to. You have the choice of selecting the Customer LAN (refer to the
Configuring the LAN IP Address section of the System Start Up Guide) or
any VLAN IP Addresses (refer to the VLANs Guide) that are configured on
the BCM in the IP Subsystem section of Element Manager.
The Published IP Address must be set as the S1 IP (or S2 IP if the BCM will
be used as a “backup” registration BCM) when configuring IP phones for
registration.
NN40011-028 Issue 1.2 BCM Rls 6.0 19
IP Telephony
Use the following procedure to check or set the Published IP Address.
1. From the Configuration tab, open the System folder and select IP Subsystem. Click on the General Settings tab.
2. If checking the existing Published IP Address for IP phone registration
purposes, view the read-only field.
3. If changing the setting, from the Published IP Interface drop-down list,
select the Customer LAN or any of the VLANs configured on the BCM.
20 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
4. A warning box will appear stating that all Voice over IP applications will
be restarted. This may result in VoIP calls being dropped. Click OK to
continue.
5. If changed, the new setting will be displayed,
6. Changing the Published IP Interface setting also has the effect of
changing the S1 Primary Terminal Proxy Server IP Addresses (S1 &
S2) in the DHCP Server IP Terminal DHCP Options screen (refer to the
DHCP Server - IP Terminal Options section of this guide for further
information).
NN40011-028 Issue 1.2 BCM Rls 6.0 21
IP Telephony
Attribute
Value
Description
Echo
cancellation
<drop-down
menu>
Enabled w/NLP
Enabled
Disabled
Enable or disable echo cancellation for your system.
Default: Enabled w/NLP (check with your internet system
administrator before changing this)
Echo Cancellation selects what type of echo cancellation is
used on calls that go through a Media Gateway. NLP refers to
Non-Linear Processing.
T.38 UDP
redundancy
<numeric
character string>
If T.38 fax is enabled on the system, this setting defines how
many times the message is resent during a transmission, to avoid
errors caused by lost T.38 messages.
Note: If any network hardware handling network traffic does not support
DSCP, the packets will not be prioritised by that hardware, and will be treated
on an equal basis to non–prioritised packets.
Media Gateways
Certain types of IP communications pass through Media Gateways on the
BCM. You can control the performance of these communications by adjusting
the parameters for echo-cancellation and UDP Redundancy.
The Media Gateways panel allows you to set basic parameters that control IP
telephony.
1. Open the Resources folder and highlight Media Gateways. The
Media Gateways screen will be displayed on the right. Configure the
Parameters as described in the following table.
Media Gateways Settings
Quality of Service (QoS) Settings
The BCM can be configured to mark voice related data packets using the
Differentiated Services Code Point (DSCP) feature, so that they have priority
over other packets on the network. Prioritised packets pass through network
hardware supporting the DSCP feature, ahead of lower priority packets. This
has obvious benefits for real time applications such as Voice over IP.
22 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
Note: Only configure BCM QoS if you have a plan of what types of packets
are prioritised on the network, and the levels (class of service) of priority for
those packet types.
The following types of data packets can be prioritised:
VoIP Signalling (SIP, H.323, and Unistim)
Voice Media
T.38 Fax Media (SIP or H.323)
DSCP Marking
Use the following procedure to set the QoS values for VoIP Signalling, Voice
Media, and Fax Media packets.
1. In Element Manager, select the Configuration tab. Open the Data Services folder, and click on QoS.
NN40011-028 Issue 1.2 BCM Rls 6.0 23
IP Telephony
Note: Avaya Automatic QoS should only be used if there are other devices on
the network that support this feature.
2. In the DSCP Marking tab, select either to use Avaya Automatic QoS
settings or select the values for each of VoIP Signalling, Voice Media,
or Fax Media.
3. A value of CUSTOM can also be selected from the drop-down lists,
which will enable a customisable ToS (Terms of Service value) to be
entered.
DSCP Mapping
In this area DSCP values are assigned to various service classes. The service
classes determine the priority level of the DSCP value.
The available Service Classes are (in order of priority):
Therefore, a packet carrying a DSCP value associated with the Critical class
will have the highest priority (assuming the default VLAN P Bit Mapping
settings are not changed).
1. Click on the DSCP Mapping tab. If you want to assign a different
service class to a DSCP value, double-click in the corresponding
Avaya Service Code field and select the class from the drop-down list.
NN40011-028 Issue 1.2 BCM Rls 6.0 25
IP Telephony
IP Telephones
IP telephones offer the functionality of regular telephones, but do not require a
hardwire connection to the BCM. Instead, they must be plugged into an IP
network which is connected to the BCM.
Calls made from IP telephones through the BCM can pass over VoIP (H.323
or SIP) trunks or across Public Switched Telephone Network (PSTN) lines.
Avaya provides two types of IP telephones. The IP telephones are wired to an
IP network using Ethernet in the case of the IP telephones, or are accessed
through your desktop or laptop computer, as in the case of the 2050 IP
Softphone.
IP telephones can be configured to the network by the end user or by the
administrator. If the end user is configuring the telephone, the administrator
must provide the user with the required parameters.
DHCP Configuration
Refer to the following sections if the BCM will be used as the DHCP server for
the IP phones.
DHCP Server - IP Terminal Options
If the BCM is configured to pass on DHCP details to IP phones using either
the “Enabled –IP Phones Only” or “Enabled –All Devices” options in DHCP
Server General Settings, then the BCM should be configured to supply the
Primary (S1) and Secondary (S2) Terminal Proxy Server IP Addresses that
the IP Phones should register against.
If the BCM will not be passing on DHCP information to IP Phones, then the IP
Terminal DHCP Options do not require configuring.
Again, if you have configured the Published IP Interface in the Published IP Interface section, the S1 and S2 will be already set to the Published IP
Address. However, you may wish to check these settings.
Use the following procedure to check or change the IP Terminal DHCP
Options.
26 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
1. From Configuration tab open the Data Services folder and select
DHCP Server. Click on the General Settings tab. Check to see if the
BCM is configured to provide DHCP information to IP Phones.
2. If either Enabled – IP Phones Only or Enabled – All Devices is
selected, then continue with configuring the IP Terminal DHCP
Options.
3. Click on the IP Terminal Options tab.
NN40011-028 Issue 1.2 BCM Rls 6.0 27
IP Telephony
Attribute
Value
Description
Primary Terminal Proxy Server (S1)
IP Address
<IP
address>
The IP address of the Proxy Server for IP phones. This should be set
to the BCMs Published IP Address.
Port
<drop-down
list>
Select the appropriate port:
BCM
SRG
Meridian 1/Succession 1000
Centrex/SL-100
Other
Port number
<readonly>
The port number on the terminal through which IP phones connect.
Action
<readonly>
The initial action code for the IP telephone.
Retry count
<number>
The delay before an IP phone retries connecting to the proxy server.
Secondary Terminal Proxy Server (S2)
IP address
<IP
address>
The IP address of the Proxy Server for IP phones. This should be set
to the BCMs Published IP Address, or a backup BCM to register
against.
Port
<drop-down
list>
Select the appropriate port:
BCM
SRG
Meridian 1/Succession 1000
Centrex/SL-100
Other
Port number
<readonly>
The port number on the terminal through which IP phones connect.
Action
<readonly>
The initial action code for the IP telephone
Retry count
<number>
The delay before an IP phone retries connecting to the proxy server.
VLAN
VLAN identifiers
(commadelimited)
Specify the Virtual LAN (VLAN) ID numbers that are given to the IP
telephones.
If you want DHCP to automatically assign VLAN IDs to the IP
telephones, enter the VLAN IDs in the following format:
VLAN-A:id1, id3,…,idn.
Where:
VLAN-A – is an identifier that tells the IP telephone that this message
is a VLAN discovery message.
Id1, id2,…idn – are the VLAN ID numbers that DHCP can assign to
the IP telephones. You can have up to 4 (BCM50) or 8 (BCM450)
VLAN ID numbers listed. The VLAN ID numbers must be a number
from 1 to 4094.
For example, if you wanted to use VLAN IDs 1100, 1200, 1300 and
1400, you would enter the following string in this box: VLAN-A:1100,
1200, 1300, 1400.
If you do not want DHCP to automatically assign VLAN IDs to the
telephones, enter VLAN-A:none, in this text box.
Note1: The Avaya IP Terminal VLAN ID string, must be terminated
with a period (.).
Note2: If you do not know the VLAN ID, contact your network
administrator.
Note3: For information about how to setup a VLAN, refer to the user
4. Ensure that the IP address is set correctly for the Primary and
Secondary Terminal Proxy Servers. Again, these addresses will be
used during the IP Phone registration process. Also, ensure that the
Port is set to BCM. This will automatically set the Port number field to
7000.
5. Configure all other fields as required.
IP Terminal DHCP Options
28 NN40011-028 Issue 1.2 BCM Rls 6.0
IP Telephony
Attribute
Value
Description
documentation that came with your VLAN compatible switch, as well
as the VLAN Guide..
Avaya WLAN Handset Settings
TFTP Server
IP Address
Enter the IP Address of the TFTP server that is used for providing
firmware to the WLAN handsets and the 2245 IP Telephony Manager
WLAN IP
Telephony
Manager 2245
IP Address
Enter the IP Address WLAN IP Telephony Manager 2245
Note: Consult with the network administrator to determine a suitable range of
addresses, co-ordinating with the existing network design. For example, it
may be necessary to set up an Address Range for VLANs that host the IP
telephones. For more information on configuring VLANs, please refer to the
VLANs Guide.
Configuring the DHCP Address Ranges
If the BCM is configured to pass on DHCP information to IP Phones, you
should configure a suitable range of addresses to assign to the IP Phones.
1. In the Configuration panel, open the Data Services folder and select
DHCP Server.
2. Click on the Address Ranges tab.
NN40011-028 Issue 1.2 BCM Rls 6.0 29
IP Telephony
3. If there aren‟t any Address Ranges configured, click on the Add button.
4. Enter the start IP address in the From IP Address field. Enter the end
IP address of the range in the To IP address field. In the Default Gateway field, enter the IP Address of the network default gateway.
This may be the BCM S1 address in some situations. Click OK to
submit the settings.
5. The new address range will be displayed.
30 NN40011-028 Issue 1.2 BCM Rls 6.0
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