Tone Commander 7210, 7220 Ip Centrex Telephone Network Administration Guide

							Tone Commander 7210/7220 Network Administration Guide
13-102865  Rev. KPage 11 of 32
  3.  Enter the start and end of the IP address
range that can be assigned to telephones.
  4.  Enter the appropriate subnet mask.
  5.  Enter a name for the scope in the Name field,
and an optional description in the Comment
field.
  6.  Exclude addresses from the range and
change the lease duration if needed. Lease
duration should be set to 7 days or longer.
  7. Click OK.
  8. Click Yes to activate the scope.
When the lease expires the phone
shows a diagnostic display if idle,
while attempting to negotiate a new IP
address lease at preset intervals. If
the phone is active, the call will be
unaffected and the diagnostic display
will be shown when the call is cleared.
If the same IP address is offered by
the DHCP server, the phone returns to
operation without restarting, otherwise
the phone will restart after receiving a
new IP address. 
						

							Page 12 of 3213-102865  Rev. K
Scope Options
  1. Select 002 Time Offset from the Unused options
list and click Add to add it to the Active Options
list.
  2. Click Edit Array.
  3.  If your network time server is set to UTC time,
enter the hex value for your locations offset from
UTC time in seconds. If your network time server
is set to local time, enter 0. Click Add.
  4. Click OK.
Time ZoneOffset
Pacific Standard Time 0xffff8f80
Pacific Daylight Time 0xffff9d90
Mountain Standard Time 0 xffff9d90
Mountain Daylight Time 0 xffffaba0
Central Standard Time 0xffffaba0
Central Daylight Time 0xffffb9b0
Eastern Standard Time 0xffffb9b0
Eastern Daylight Time 0xffffc7c0
  5. Add 003 Router to the Active Options list.
  6. Click Edit Array.
  7.  Enter the router IP address, then click Add.
  8. Click OK. 
						

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  9. Add 004 Time Server to the Active Options list.
  10. Click Edit Array.
  11.  Enter the IP address of the SNTP time server for
your network, then click Add.
  12. Click OK.
  13. Add 066 Boot Server Host Name to the Active
Options list.
  14.  Enter the TFTP boot server’s IP address in the
String field.
  15. Click OK.
  16.  002 Time Offset, “003 Router”, 004 Time
Server, and “066 Boot Server Host Name”
should now appear in the Scope Options
pane.
To preclude the issuance of a new IP
address each time the phone reboots, it
is recommended that ICMP ping prior
address assignment be disabled at the
DHCP server. 
						

							Page 14 of 3213-102865  Rev. K
TFTP Server Configuration  ______________________
Telephones download configuration information and software upgrades from a TFTP server. The TFTP
server’s IP address must be identified during DHCP server setup, or entered manually in the phone if
DHCP is disabled.
Configuration packages distributed by Tone Commander include a phone boot ROM image file,
application software image file, two configuration files, and a Readme text file.
All files must be located in the TFTP server’s root folder .
The root folder should contain the following files:
filename.binBoot program image.
filename.zCompressed application image.
tcs7200a.txtConfiguration options not included in the standard DHCP options.
tcs7200b.txtNames the application file and boot program to be downloaded to the phone.
These files need to be in the TFTP server root folder.
The phone will attempt to download parameters or programs only if the
information in the above .txt files indicates that the phone is not at the desired
current configuration.
readme.txtinformation file
xxxxxxxxxxxxxx.txtOptional files allow customers to set up phones with specific configuration
parameters or program versions based on the phones’ alias (xxxxxxxxxxxxxx
in the file name, e.g., 30947701840101). Any information that is different in
this file overrides the information in the tcs7220a,b files. A separate file is
required for each phone that differs from the standard configuration or
programming as defined in tcs7200a.txt and tcs7200b.txt files.
Configuration files can be modified with any text editor. See page 29 for file format descriptions. 
						

							Tone Commander 7210/7220 Network Administration Guide
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Telephone Configuration Update  _________________
Telephone operating software and configuration parameters can be automatically updated daily at a
preset time, or manually updated.
Automatic Update
1. Select AUTO from the TFTP CONFIG
UPDATE menu.
(Setup → INSTL → UPDATE→ TFTP→ AUTO)
2. Enable TFTP Update if necessary.
3.  Enter the time you want the daily update
to occur.
The minutes value is calculated based on the
MAC address of the phone, to minimize the
possibility of multiple phones simultaneously
requesting updates. You may manually enter
the minutes digits, but it is recommended that
you use the default calculated value.
4. Press the Done key.
NOTE – For special instructions and information,
please refer to the Upgrading Notes
associated with a specific upgrade.
IP CONFIGURATION UPDATE
DHCP  TFTP  FLASH TELNET
TFTP UPDATE= DISABLED
ENABLE
TFTP CONFIG UPDATE
START AUTO
       SETUP MENU
INSTL ADMIN  USER
INSTALLATION OPTIONS   \
  IP  ALIAS  SPID UPDATE
TFTP UPDATE TIME=03:26AM
DISABLE     |BKSP  AM/PM 
						

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Manual Update
To perform a manual update, the phone must have the TFTP server’s IP address entered manually or
downloaded via DHCP.
Manual TFTP Server Entry
Starting Manual Update
Use the TFTP CONFIG UPDATE - START option
to initiate the manual update.  1. Select TFTP from the IP ADDRESS/CONFIG
menu.
  2.  Enter the TFTP server address with the dial
pad.
  3. Press the Done key.
       SETUP MENU
INSTL ADMIN  USER
INSTALLATION OPTIONS   \
  IP  ALIAS  SPID UPDATE
IP ADDRESS/CONFIG      \
PHONE GATEWY TFTP iMERGE
TFTP=192.168.72.195
      |BKSP CLEAR  PING
TFTP CONFIG UPDATE
START  AUTO
IP CONFIGURATION UPDATE
 DHCP  TFTP FLASH TELNET
       SETUP MENU
INSTL ADMIN  USER
INSTALLATION OPTIONS   \
  IP  ALIAS  SPID UPDATE 
						

							Tone Commander 7210/7220 Network Administration Guide
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Quality of Service  ______________________________
Quality of Service (QoS) settings can improve voice performance over a network by prioritizing voice
packets, and adjusting packet buffering and packetization rate.
These settings affect network traffic, and should not be changed unless required to correct audio
problems.
To view or change Quality of Service settings,
select QoS from the Installation Options menu.
(Setup → INSTL → More
4 → More
4 → QoS)
Layer 2, Layer 3, and Codecs can be selected
from the Quality of Service menu.
When finished viewing or changing any setting,
press the Done key to return to the previous
menu or the Setup key to exit Setup Mode.
INSTALLATION OPTIONS   \
 KEYS  MODE   VA   PARAM
QUALITY of SERVICE
  L2    L3        CODECS
INSTALLATION OPTIONS   \
  QoS  PASSWD RESET
       SETUP MENU
INSTL ADMIN  USER
INSTALLATION OPTIONS   \
  IP  ALIAS  SPID UPDATE 
						

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Ethernet Layer 2 802.1Q Options
IEEE 802.1Q allows packets to be assigned one of eight priority levels. Voice traffic with less than 10 ms
of latency is normally assigned a priority level of 6 (phone default).  Network switches must support
802.1Q for this setting to have an effect. If the LAN does not support 802.1Q, this parameter should be
set to OFF.
NOTE – The Ethernet card in a PC connected to the phone must support 802.1Q, since the phone does
not add 802.1Q tagging to packet headers not generated in the phone. PC packets are passed
through without modification.
1. Select L2.
2.  Select ON or OFF to enable/disable Layer 2
802.1Q support.
3.  To change the priority of voice or signaling
packets, select PRIORITY, select VOICE or
SIGNAL, then enter the priority level (0-7) with
the dial pad.
4.  If the network uses virtual LANs, select ID, then
enter the appropriate VLAN ID with the dial
pad.
802.1Q VLAN ID=0000
|BKSP CLEAR
QUALITY of SERVICE
  L2    L3        CODECS
VOICE 802.1Q PRIORITY=6
LAYER 2 802.1Q PRIORITY
VOICE SIGNAL
SIGNAL 802.1Q PRIORITY=6
LAYER 2 802.1Q=ON
 OFF  PRIORITY ID 
						

							Tone Commander 7210/7220 Network Administration Guide
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IP Layer 3 Differentiated Services (DiffServ)
The Quality of Service for voice and signaling packets is determined by each service type’s Differentiated
Services Code Point (DSCP) setting. This value must be matched to network router settings. The default
setting for both voice and signaling is 46.
1. Select L3.
2.  Select VOICE or SIGNAL to view or set the
DSCP value for voice or signaling packets.
3.  Use the dial pad to enter a new DSCP value
(0-63).
Codecs
Five codec selections are available. Any
codec/packet rate combination can be
disabled.
1. Select CODEC.
2.  Cycle through the codecs with the More
4
key. The display will show the codec type
(G.711, G.729A, or G.723), the packet
rate (10, 20, or 30ms), and the average
and maximum jitter delay for each enabled
codec.
3.  Enable or disable codecs as required.
If more than one codec is enabled, the
selected codec is negotiated on a per-call
basis, between the list of codecs enabled
on iMerge and codecs enabled on the
phone. iMerge codecs take precedence.
You can view the negotiated codec
selection through the Packet Diagnostics
menu (see below).
CAUTION – Make sure at least one
codec that is supported by the iMerge
CFG is enabled, otherwise a voice
channel cannot be established.
QUALITY of SERVICE
  L2    L3        CODECS
VOICE DIFFSERV DSCP=46
|BKSP CLEAR
LAYER 3 DIFFSERV
VOICE SIGNAL
SIGNAL DIFFSERV DSCP=46
|BKSP CLEAR
G.711/20ms JTR=35/80ms \
DISABLE             JTR
G.711/30ms JTR=45/90ms \
DISABLE             JTR
G729A/20ms DISABLED    \
ENABLE
G.723/30ms JTR=45/90 ms\
DISABLE             JTR
JTR DELAY AVE=45/MAX=90
 AVE+  AVE-  MAX+  MAX-
QUALITY of SERVICE
  L2    L3        CODECS
G.711/10ms JTR=20/40ms \
DISABLE             JTR 
						

							Page 20 of 3213-102865  Rev. K
Codec Selections
G.711 – Uncompressed, 64Kbps data rate  (10, 20, or 30 ms packet rate)
G.729A – Compressed, 8Kbps data rate  (20 ms packet rate)
G.723 – Compressed, 6.3Kbps data rate  (30 ms packet rate)
Uncompressed codecs with higher packet rates (e.g. G.711/10ms) may provide better voice
performance with lower audio delay, but increase network traffic.
4.  Average and maximum jitter delay can be set for each enabled codec. Select JTR, then change the
delay values with AVE+, AVE-, MAX+, and MAX-. Note that the maximum jitter delay cannot be set to
less than twice the average jitter delay setting.
The average jitter delay is the average amount of time that packets are received before they are played.
Since IP networks have variable packet transmission delays, yet packets must be played at a constant
rate, a local jitter buffer is required to “smooth out” the variations in packet arrival times. The larger the
variance in packet delay through the network, the larger the average jitter delay setting must be to
compensate. Audio dropouts may occur (due to delayed packets) if the average jitter delay is too small.
Unacceptable audio delay may result if the jitter delay is too long.
Use the Packet Diagnostics menu (see below) to review packet statistics and jitter performance. If a
significant number of packets (>5%) are concealed and a significant portion of packets are delayed longer
than the average jitter buffer delay setting, the average jitter buffer setting should be increased to
“capture” these dropped packets. If there are consistently very few concealed packets and almost all
received packets are delayed less than the average jitter delay setting, this setting can likely be
decreased without increasing the number of concealed packets to improve audio delay.
The maximum jitter delay is the longest delay allowed until playback for a packet that arrives early. This
setting also affects dropped packets due to clock slips on long-duration calls. If the iMerge packet
transmission rate is slightly faster than the phone playback rate, then the jitter buffer will gradually fill up
until it reaches the maximum capacity and overflows (after an hour or two). When this happens, an audio
“skip” will occur as the jitter buffer is reset to the average delay setting. Immediately before the overflow
correction occurs, the additional packet delay due to the jitter buffer is as long as the maximum jitter
delay. Therefore, the maximum jitter delay should be set long enough so that overflow events do not
happen very often, but not so long that excessive audio delay occurs before a correction is made.
Diagnostic tools built into the 7210/7220 can assist you in determining the optimum jitter delay settings for
your network. The ping test provides a quick method of measuring single packet network delays. For a
more detailed picture of packet delay and loss during actual calls, use the Packet Diagnostics menu.
Ping Test
You can test network delays by pinging the iMerge
server (Setup → ADMIN → DIAG → PING →
iMERGE). The ping delay will be shown in the
display. Select PING several times to perform
multiple ping tests and note the difference between
delay measurements.
The difference between readings (not absolute delay
values) gives an “order of magnitude” indication of
the average jitter buffer setting needed to prevent
dropped packets that result in audio interruptions.
DIAGNOSTIC OPTIONS
 LINK PACKET PING
iMERGE PING DELAY=20ms
                   PING
SELECT DEVICE TO PING  \
PHONE GATEWY TFTP iMERGE
       SETUP MENU
INSTL ADMIN  USER
ADMINISTRATION OPTIONS \
INSPCT VERS  TEST  DIAG