Konftel Konftel 300IP Instruction Manual

Conference phones for every situation
Installation and Administration
of Konftel 300IP
ENGLISH
PRELIMINARY

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CONTENT
This document only includes setup, registration of accounts, and conguration. The use of
the conference phone is described in the Quick reference guide and the User guide. Latest
version of all documentation can be downloaded from www.konftel.com/300ip.
Description 2
Display information and web interface 3
Display screen................................................................3
Navigation and selection in menus ...................................4
Using the web interface...................................................6
Installation 7
Connecting.....................................................................7
Obtaining a network address ...........................................8
Login.............................................................................8
Software upgrade and basic settings ................................9
Registering an account..................................................12
Settings 14
Login...........................................................................14
Basic........................................................................... 14
SIP.............................................................................. 17
Network.......................................................................23
Media ..........................................................................24
Web interface...............................................................27
Time & Region .............................................................28
Provisioning .................................................................29
System ........................................................................30
Provisioning – upgrade and conguration 31
Firmware upgrade ......................................................... 31
Using a conguration le...............................................33
Managing phone book and conference guide 38
Importing and exporting contacts ...................................38
Importing and exporting conference groups .....................39
Checking status and logs 40
Device .........................................................................40
Network.......................................................................40
Time & Region .............................................................41
SIP.............................................................................. 41
Media ..........................................................................42
Log .............................................................................43
SIP Trace.....................................................................44
Trouble shooting 45
Hard reset to factory settings .........................................45
Fault codes ..................................................................45
Technical data 46
Service and guarantee 48
ABOUT THIS DOCUMENT

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DESCRIPTION
DISPLAY SCREEN
On Hook
Press to display this screen.
Off Hook
Press to display this screen.
Line status:
Line free (Before account name – telephone not registered)
Line connected (Before account name – telephone registered)
Line on hold (“HOLD” displayed on the screen – all calls on hold)
Line (called party) busy
Own line put on hold by other party
Recording call
Secure connection
Information text displays one of the following:
• Number or name of each phone line
(The name will be displayed if a number is in the phonebook)
• Explanation of what you should do (For example ENTER NUMBER)
• Status (For example HOLD when you place all calls on hold)
Line menu
Press LINE to switch to and from this menu.
SD memory card port
Expansion
microphone
port
Display screenSpeaker Microphone Keypad
Power supply port
Security lock port
AUX port
Network cable port
Expansion microphone port
LEDs
Flashing blue Incoming call
Steady blue light Call in progress
Flashing red On hold,
microphone and
speakers turned
off
Steady red light Mute,
microphone
turned off
Alphanumeric keys, phone book
MENU
settings menu
UP ARROW
navigation in menus,
display of call list
C
no/end/back,
recording
Increase volume
Decrease volume
Mute
HOLD
OK
yes/conrm choice
Answer/connect
new line
Hang up/end line
CONFERENCE
Automatic dialling
of conference
groups
LINE
Line selection
DOWN ARROW
navigation in menus,
display of call list
DISPLAY INFORMATION AND WEB INTERFACE
Maintenance
Clean the equipment
with a soft, dry cloth.
Never use liquids.
Date
Time
Display text (can be changed)
Registered
Not registered
Account name (can be changed)
Call duration
Time
Information text (see below)
Phone lines (L1–L4)
Line status (see below)
Secure connection (see page 10)
Line/number/name
New line (two lines if two accounts are registered)
Option for creating or splitting conference calls

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Menu tree
Menu tree, advanced settings
The advanced settings is protected by administrators PIN. The default value is 1234.
Menu
Press MENU to switch to and from a menu.
NAVIGATION AND SELECTION IN MENUS
Press MENU.
Select the option you want from the menu using the arrow buttons.
Conrm by pressing OK to select the marked option.
Cancel the setting or go back one level in the menu by pressing C.
Quit the menu by pressing MENU again.
Note that after you have made changes to a setting, you must press OK to activate
the setting.
It is possible to open a menu option directly by pressing the number button that
corresponds to the position of the option in the menu (e.g. 2 to open PHONEBOOK
and then 3 to select EDIT CONTACT).
Writing style in instructions
In the instructions, MENU > SETTINGS (5) means you should:
Press MENU.
Mark the SETTINGS option using the arrow buttons and conrm by pressing OK to
open the menu (or press button number 5).
Correspondingly, Phonebook > Conference Guide in the web interface means you should
select Menu Phonebook and the Conference Guide tab.
DISPLAY INFORMATION AND WEB INTERFACE DISPLAY INFORMATION AND WEB INTERFACE
The simplest way to make settings and edit contacts
is using a PC and the Konftel 300IP web interface.
CONF GUIDE
BASIC
SETTINGS
SETTINGS SYSTEM STATUSRECORDING
DEFAULT
PROFILE 1
PROFILE 2
PROFILE 3
PROFILE 4
PROFILES PHONE BOOK
SEARCH
CONTACT
ADD
CONTACT
EDIT
CONTACT
ERASE
CONTACT
ERASE
ALL
STATUS
SEARCH
GROUP
ADD
GROUP
EDIT
GROUP
ERASE
GROUP
ERASE
ALL
STATUS
PLAYBACK
FILE
RENAME
FILE
ERASE
FILE
ERASE
ALL
SETTINGS
STATUS
BASIC
ADVANCED
RESTART
REBOOT
RESET
ACCOUNTS
NETWORK
NAT
TRAVERSAL
MEDIA
TIME
DEVICE
ACCOUNTS
LANGUAGE
KEY TONE
RING LEVEL
EQUALIZER
AUX
TIME
FORMAT
SCREEN
TEXT
DEFAULT
Current menu
Submenu
Marked option – open by pressing OK button
Scrolling list
(indication of where the marked option is in the list or menu)
List of setting options:
Existing settings
Marked option – select by pressing OK button
List of names:
Marked name – select by pressing OK button
ACCOUNTS
SETTINGS
NETWORK NAT
TRAVERSAL
BASIC
ADVANCED
STUN
OFFER ICE
ACCOUNT 1
ACCOUNT 2
TRANSPORT
MEDIA TIME REGION WEB
INTERFACE
NTP
TIME DATE
TIME ZONE
DAYLIGHT
SAVING
CODEC
VAD
DTMF
SIGNALLING
SECURITY
G722
PCMA
PCMU
G729
SRTP
SIGNALLING
PIN
ENABLE/
DISABLE
ACCOUNT
EDIT
ACCOUNT

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USING THE WEB INTERFACE
You can use the web browser of a PC connected to the same network to manage
contacts, conference groups and settings in the Konftel 300IP.
For security reasons, recordings can only be managed directly on the Konftel 300IP.
All other settings that can be made directly on the Konftel 30IP can also be made via
the web interface. It is also possible to import and export contacts and conference
groups, name user proles and change PIN codes, which can only be done via the web
interface. The administrator can also view logs, update software and create a congura-
tion le.
The default setting for the PIN code is 0000 for the user account (Default, Prole 1,
Prole 2, Prole 3 and Prole 4) and 1234 for the administrators account (Admin).
We recommend that you change the PIN codes in order to protect the settings. The
code may consist of eight digits. The administrator can always view and change the
PIN codes to the user accounts. The administrator’s PIN code can only be reset with a
complete reset to factory settings.
Checking IP address
Press MENU and select the sub menu STATUS > NETWORK (7,2).
Check the conference phone’s network address under the heading IP ADDRESS.
Use this address to log into the web server in the conference phone.
Login
 Log into the web server in Konftel 300IP by entering the phone’s network address
in your computer’s web browser.
Select Admin as Prole and enter your PIN.
DISPLAY INFORMATION AND WEB INTERFACE
A short and simplied installation description is found in the printed Installation guide.
The guide includes the basic settings for a quick start and works in most cases.
CONNECTING
 Connect Konftel 300IP to the network as illustrated below.
 Plug the Konftel 300 into the mains using the power adapter as illustrated below.
 Konftel 300IP can be driven directly from the network (Power over Ethernet) if the
network supports this.
Place the conference phone in the middle of the table.
Konftel 300IP must obtain a network address and be registered in a SIP PBX before
it can be used. To register an account and make the settings in Konftel 300IP, it is
the easiest to use a computer connected to the same network and use the built-in web
server.
INSTALLATION

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OBTAINING A NETWORK ADDRESS
Connecting to a network with DHCP
See “Check IP address” under “USING THE WEB INTERFACE” on page 6.
Connecting to a network with static IP addresses
You need information on IP address, host name, domain, netmask, gateway, DNS 1,
and DNS 2. Host name can be set freely and domain and secondary DNS can be left
blank.
Press MENU and select SETTINGS > ADVANCED (5,2).
 Enter the PIN code.
The default code is 1234.
Select NETWORK (2)
Select STATIC IP.
 Enter values for the IP ADDRESS.
 Enter three digits (begin with 0 if necessary), press OK, enter three digits, and so
on.
 Enter HOST NAME
Default is kt300ip.
 Enter DOMAIN
 Enter NETMASK
 Enter GATEWAY
 Enter DNS 1
 Enter DNS 2
The display shows DONE.
LOGIN
See “Login” under “USING THE WEB INTERFACE” on page 6.
INSTALLATION
SOFTWARE UPGRADE AND BASIC SETTINGS
The following settings should be done during installation.
Note that all settings on the Basic tab also affect the user prole Default. Other user
proles can be changed individually.
The settings on the Basic tab, except the name and PIN for Admin, can be modied by
any user. Other settings require a login as Admin.
Upgrade software
See the heading “PROVISIONING – UPGRADE AND CONFIGURATION” on page 31 for
a detailed description and upgrading options.
Select Settings > Provisioning.
Click on Check Now.
Compare the latest version, with the current version (shown on the web page).
If you want to upgrade, select the desired version in the list box and click on
Upgrade.
The browser window and the display on Konftel 300IP shows that the upgrade has begun.
The download and installation can take several minutes. Do not interrupt the
upgrade and do not disconnect any plug on Konftel 300IP during the upgrade. An
interrupted upgrade can make the conference phone unusable.
When installation is complete, the text “Upgrade Complete. The unit will be
rebooted.” is shown in your browser, and after a while you hear the Konftel music
signature, that shows that the conference phone has started.
INSTALLATION

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Setting time and region
Select Settings > Time & Region.
Select the time zone and if you want automatic correction for DST (Daylight saving).
It is also possible to set the time and date manually or choose a different time
server.
Select the region where you are.
This setting affects the signalling.
Save the setting.
Konftel 300IP reboots with the new settings.
INSTALLATION
Changing the language
Select Settings > Basic.
Select the desired language in the list box after Language and save the setting.
Changing the PIN
PIN code for Admin should be changed from the default setting to protect the settings.
Note the new PIN in a safe place. The administrator’s PIN code can only be reset by a
full factory reset!
Select Settings > Basic and click the Edit button on the Admin line.
 Enter a new PIN.
The PIN code can contain 8 digits.
Click on the Set and Save buttons.
INSTALLATION

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REGISTERING AN ACCOUNT
The conference phone can be registered in a company SIP PBX or with a public IP
telephony service provider. You can store settings for two accounts in Konftel 300IP. To
register your phone, you must have access to the account information and all necessary
settings that the SIP PBX or service provider requires.
See the heading “SIP” on page 17 for a detailed description of all settings.
Select Settings > SIP.
Click Yes at Enable account under Account 1.
 Enter the account information you have received.
 Account name can be chosen freely and is the name or phone number you want to
appear in the phone display.
Leave the default values if you have no other information.
 Select a method of NAT traversal if you got information of this.
INSTALLATION
Select a different transport protocol if you got information of this. See page 20
about using a secure transport protocol.
Save the settings by clicking the Save button.
Konftel 300IP responds by showing REGISTERING. Your selected account name appears at the
bottom of the display with a lled square in front, if the registration is successful.
Make media settings
Select a different codec priority, if you do not accept the default settings. See page
24.
Select a SRTP option if you need a secure media protocol. See page 25. Note that
this also requires a corresponding transport setting on the SIP tab.
INSTALLATION

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Almost all settings can be done directly on Konftel 300IP. See “NAVIGATING THE
MENUS AND MAKE SETTINGS” on page 5 for using the menu system. Because it is
easier to use the web interface, we describe the settings using this interface.
For safety reasons, recordings can only be managed directly on the Konftel 300IP. All
other settings can be changed via the web interface. The web interface also allows you
to import and export contacts and conference groups, rename user proles and change
PIN codes. As an administrator, you can also study logs, upgrade the software en create
an XML based conguration le for easier management of a set of phones.
LOGIN
See “USING THE WEB INTERFACE” on page 6 for a description of how to log in to the
web server in Konftel 300IP.
BASIC
Select Settings > Basic.
SETTINGS
These settings affect the Admin and Default proles. To change the basic settings of a
user prole, you need to log in with that prole.
Proles – edit name and PIN
PIN code can be changed from the default setting to protect the settings.
Select Settings > Basic and click the Edit button on the account you want to
change.
 Enter a new PIN code.
The PIN code may consist of 8 digits.
You can also choose to change the name of a user prole.
Click on the Set and Save buttons.
Make a note of the new PIN code and keep it in a safe place.
The administrator’s PIN code can only be reset with a complete reset to factory
settings!
Default account
This setting determines which account to be used as default. By pressing LINE before
dialing a number, you can choose the alternative account for the call.
 Select Account 1 or Account 2 and click on the Save button.
On phone: MENU > SETTINGS > BASIC > ACCOUNT (5,1,1).
Language
Select language using the list box and click on the Save button.
On phone: MENU > SETTINGS > BASIC > LANGUAGE (5,1,2).
Ring level
There are six volume levels plus a silent mode. You will hear the ring tone for each level
you select. If you select silent mode, only the blue LEDs on the phone ashes when an
incoming call is received.
Select level using the list box and click on the Save button.
On phone: MENU > SETTINGS > BASIC > RING LEVEL (5,1,4).
Key tone
You can select whether or not you want a tone to be heard when you press a button.
Select On or Off and click on the Save button.
On phone: MENU > SETTINGS > BASIC > KEY TONE (5,1,3).
SETTINGS

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SETTINGS
Recording tone
A short beep is heard every 20 seconds so that all the parties in the call know it is
being recorded. This feature can be turned off.
Select On or Off and click on the Save button.
On phone: MENU > RECORDING TONE > SETTINGS (4,5).
Time format
Select 12 hour or 24 hour and click on the Save button.
On phone: MENU > SETTINGS > BASIC > TIME FORMAT (5,1,7).
Equalizer
The sound reproduction can be adjusted to the required pitch (SOFT, NEUTRAL or
BRIGHT).
Select Soft, Neutral or Bright and click on the Save button.
On phone: MENU > SETTINGS > BASIC > EQUALIZER (5,1,5).
Screen text
The text on the display screen is shown when Konftel 300IP is in stand-by mode (on
hook).
 Enter your new text in the text box and click on the Save button.
On phone: MENU > SETTINGS > BASIC > SCREEN TEXT (5,1,8).
SIP
Select Settings > SIP
Konftel 300IP can store information for two accounts, e.g. one PBX and one public
service provider.
Why use two accounts?
It may be practicable to register a second account if the PBX or SIP server is located in
another country than the phone. Otherwise would local calls using the PSTN (telephone
network), connected through the ordinary PBX or SIP server, be connected as interna-
tional calls.
Account 1 and Account 2
Enable account It is possible to store account information for future use, but
temporarily make it disabled.
SETTINGS

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Account name This is the name showed on the display and can be set according
to company standards.
User The account (customer) name.
Registrar Shall contain the IP address or the public name of the SIP server
where the account is registered (e.g. 10.10.1.100 for a local SIP
server or sip.company.net for a public VoIP service provider)
Proxy Shall contain the proxy server used for Internet communication,
if any. Can be left blank.
Realm The protection domain where the SIP authentication (name and
password) is valid. This is usually the same as the registrar. If
left blank, or marked with a “*”, the information is taken from
the Registrar eld.
Authentication name The name used for the Realm authentication. This may be the
same as the user name, but must be lled in.
Password The password used for the Realm authentication.
Registration Interval This is a request to the SIP server for when the registration
should expire. Konftel 300IP automatically renews the
registration within the time interval if the phone still is on and
connected to the server. The default value is 1800 seconds.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > ACCOUNTS (5,2,1).
Nat traversal
NAT (Network Address Translation) is a rewall or router function that operates by
rewriting the IP addresses in the IP headers as packets pass from one interface to the
other. When a packet, for example, is sent from the inside, the source IP address and
port are rewritten from the private IP address space into the address space on the
outside (Internet).
NAT rewrites the addresses but leaves the packages themselves untouched. This kind of
translation works just ne for many protocols, but causes a lot of trouble for SIP pack-
ages that contain address information in its content (for example an INVITE request
from one IP address to another).
NAT traversal is a solution to this problem, providing a “view from the outside” that
makes it possible to replace the IP address in the SIP requests with the address shown
on the other side of the rewall.
Note that in some cases NAT traversal is not necessary. Some public service providers
of IP telephony keep track of the actual IP address used to register a phone, and the
one used in the SIP requests from the same phone, and then replaces the addresses in
the SIP messages.
SETTINGS
STUN STUN (Simple Traversal of UDP through NATs) is a protocol
for assisting devices behind a NAT rewall or router with their
packet routing. STUN is commonly used in applications of
real-time voice, video, messaging, and other interactive IP com-
munications.
The protocol allows applications operating through a NAT to
discover the presence and specic type of NAT, and obtain the
mapped (public) IP address (NAT address) and port number
that the NAT has allocated for the application’s User Datagram
Protocol (UDP) connections to remote hosts. The protocol re-
quires assistance from a 3rd-party network server (STUN server).
STUN should be activated if an external SIP server cannot
connect to the Konftel 300IP behind a rewall NAT function and
the SIP server supports STUN. A suitable STUN server is usually
provided by the VoIP service provider.
Note: STUN might also be referred to as Session Traversal
Utilities for NAT.
STUN host The IP address or public name of the STUN server.
Offer ICE ICE (Interactive Connectivity Establishment), is a STUN addition
that provides various techniques to allow SIP-based VoIP devices
to successfully traverse the variety of rewalls that may exist
between the devices. The protocol provides a mechanism for
both endpoints to discover the most optimized path to be used
for the media trafc.
TURN TURN (Traversal Using Relay NAT) TURN is an extension to the
STUN protocol to facilitate NAT traversal when both endpoints
are behind symmetric NAT. With TURN, media trafc for the ses-
sion will have to go to a relay server. Since relaying is expensive,
in terms of bandwidth that must be provided by the provider,
and also additional delay for the media trafc, TURN is normally
used as the last resort when endpoints cannot communicate
directly.
TURN User User authentication name on the TURN server.
TURN host The IP address or public name of the TURN server.
Password User authentication password on the TURN server.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > NAT TRAVERSAL (5,2,3).
SETTINGS

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Transport
The transport setting only concern which protocol to be used for SIP messages between
the devices involved. These settings do not include the media (the actual call). The
settings on the Media tab should be set accordingly.
Note that if you choose to use a secure connection, both units must support it.
Otherwise they cannot negotiate a connection. If an incoming call demands a secure
TLS or SIPS connection, Konftel 300IP uses the appropriate protocol even if you have
set the phone to use UDP.
Protocol UDP (User Datagram Protocol) is a protocol on the transport
layer in the Internet Protocol Suite. It is a stateless protocol for
short messages – datagrams. Stateless imply that it does not in
advance establish any connection between sender and receiver.
UDP does not guarantee reliability or ordering in the way that
TCP does. Datagrams may arrive out of order or go missing
without notice. The advantage is speed and efciency.
UDP is the default protocol for SIP.
TCP (Transmission Control Protocol) is a protocol on the transport
layer in the Internet Protocol Suite. TCP is the standard protocol
for Internet communication. TCP keeps track of all individual
packets of data, that they arrive to the receiver and are put to-
gether properly. TCP is not the default protocol for SIP, because
it is slower and uses more bandwidth than UDP.
With UDP and TCP, SIP packets travel in plain text. TLS
(Transport Layer Security) is a cryptographic protocol that provide
security and data integrity for communications over TCP/IP
networks. TLS encrypts the datagrams of the transport layer
protocol in use. The secure connection may be to the end device
or to the rst server (usually the SIP server where the phone is
registered). There is no guarantee that there is a secure channel
to the end point, but because the SIP server is the only part
receiving the user authentication, this is still a rather secure
solution.
SIPS (Secure SIP) is a security measure that uses TLS to provide
an encrypted end-to-end channel for the SIP messages. To use
SIPS, however, both VoIP devices and the SIP server must sup-
port it.
 Even if Transport is set to TLS or SIPS, Konftel 300IP still accepts incoming UDP
or TCP signalling.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > ACCOUNTS > TRANSPORT (5,2,1,3).
SETTINGS
TLS Settings
If you select TLS or SIPS under the transport setting, this additional settings appear on
the page.
It may be possible to use secure communication without using a certicate and making
any changes to this settings. In some cases the SIP server requires a certicate for
user/client verication, if you choose to use TLS or SIPS. This should be specied in
the account information.
You are also able to increase security furthermore by requiring verication of the server,
or the client when Konftel 300IP acts as a server for incoming calls.
Method The TLS include a variety of security measures. The methods is
dened in the versions of the standard (SSL, SSL v2, SSL v3,
TLS v1, TLS v2). The default method is SSLv23, which accepts
both SSL v2 and v3.
Negotiation timeout The TLS settings are negotiated during a call setup (both incom-
ing and outgoing). If this negotiation does not succeed within the
specied time (seconds) the negotiation is aborted. Timeout is
disabled with 0 (zero).
Verify client When set to On, the Konftel 300IP will activate peer verication
for incoming secure SIP connections (TLS or SIPS).
Require client certicate
When set to On, the Konftel 300IP rejects incoming secure SIP
connections (TLS or SIPS) if the clients doesn’t have a valid
certicate.
Verify server When Konftel 300IP is acting as a client (outgoing connections)
using secure SIP (TLS or SIPS) it will always receive a certicate
from the peer. If Verify server is set to On, Konftel 300IP closes
the connection if the server certicate is not valid.
SETTINGS

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23
Certicate Here you can upload a certicate to the Konftel 300IP to be
used for TLS or SIPS communication.
A certicate is a le that combines a public key with information
of the owner of the public key, all signed by a trusted third party.
If you trust the third party, then you can be sure that the public
key belongs to the person/organization named in that le. You
can also be sure that everything you decrypt with that public key
is encrypted by the person/organization named in the certicate.
Root certicate The public key in the root certicate is used to verify other
certicates. A root certicate is only needed if you have selected
client or server verication.
A root certicate is a certicate that is signed by the same
public key that is in the certicate, a so-called “self-signed”
certicate. A typical root certicate is a certicate received from
a Certicate Authority.
Private key Here you can upload a private key to the Konftel 300IP to be
used for TLS or SIPS communication.
A private key is one of the keys in a key-pair used in asymmetric
cryptography. Message that is encrypted with the public key can
only be decrypted using the private key.
Private key password The password used for encryption of the private key, if it is
encrypted.
SETTINGS SETTINGS
NETWORK
Select Settings > Network.
DHCP Dynamic Host Conguration Protocol is a protocol used by
network devices (clients) to obtain the parameters necessary
for operation in the IP network. This protocol reduces system
administration workload, allowing devices to be added to the
network with little or no manual conguration.
DHCP should be set to on if no other information is given. When
set to on, all other information on this page will be set automati-
cally.
IP address IP address of the device (Konftel 300IP). The address is pro-
vided by the network administrator or service provider if DHCP is
not in use.
Hostname Is set to kt300ip as default. Can be changed to suitable name.
Netmask Is usually set to 255.255.255.0 to limit network trafc to the
subnet.
Domain The domain where the device is located. May be left blank.
Gateway The device or server used for Internet communication.
Primary DNS The address to the primary DNS (Domain Name System) server -
a program or computer that maps a human-recognisable name to
its computer-recognisable identier (IP address).
Secondary DNS The address to an optional secondary DNS server.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > NETWORK (5,2,2).

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25
SETTINGSSETTINGS
MEDIA
Select Settings > Media.
The media settings affect how the audio is sent between the devices. The involved
devices perform a negotiation using SIP before a call is connected. All devices must
support the same media types, codecs and security settings.
Codec
Codecs are used to convert an analogue voice signal to a digitally encoded version and
vice versa. Codecs vary in the sound quality and the bandwidth required. Konftel 300IP
supports the most common codecs and each codec can be given a precedence depend-
ing of if you prefer high quality audio or low bandwidth use.
The priority can be set to from 4 (high) to 1 (low) or 0 (disabled)
G722 G.722 is an ITU-T standard codec that provides 7 kHz wideband
audio at a data rate within 64 kbit/s. It offers a signicant
improvement in speech quality over older narrowband codecs
such as G.711, but requires a high quality network connection
between the devices.
G711 Alaw G.711 is an ITU-T standard codec that uses audio companding.
Companding algorithms reduce the dynamic range of an audio
signal. In analogue systems, this can increase the signal-to-noise
ratio achieved during transmission, and in the digital domain, it
can reduce the quantization error.
There are two main compression algorithms dened in the
standard, the µ-law algorithm (used in North America & Japan)
and A-law algorithm (used in Europe and the rest of the world).
G711 Ulaw See G711 µ-law above.
G729 G.729 is an ITU-T standard codec that operates at 8 kbit/s. It
is mostly used in VoIP applications with low bandwidth require-
ment.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > MEDIA > CODEC (5,2,4,1).
Security
The media in VoIP calls is usually sent using the RTP protocol (Real-time Transport
Protocol). RTP is a standardized packet format for delivering audio and video over the
Internet.
SRTP (Secure Real-time Transport Protocol) is an extension to RTP to provide encryp-
tion, message authentication and integrity for the audio and video streams.
All devices must support SRTP to establish a connection, and therefore there is pos-
sible to set SRTP as disabled, optional or mandatory.
SRTP If set to disabled, the media is sent using RTP. Note that despite
this setting, Konftel 300IP will still use a secure channel if the
opposite device demands it.
If set to optional or mandatory, a padlock will be shown in the
right lower corner on the display. If the other devices support
SRTP, the padlock will be shown closed. If not it will be shown
open.
If set to mandatory, the call will not be connected if the other
devices do not support SRTP.
Secure signalling The SIP messages (signalling) and the SRTP cipher key are sent
on a different channel than the media and are not affected by
the RTP/SRTP setting. To ensure a secure connection, the signal-
ling must be secured using TLS or SIPS, see page 20. Note that
the SIP transport setting must be set accordingly.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > MEDIA > SECURITY (5,2,4,4).

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SETTINGS
WEB INTERFACE
Select Settings > Web interface.
The web server in the Konftel 300IP supports secure connections using HTTPS.
Enable HTTPS Set Enable HTTPS to On if you need a secure communication
between the PC used for setup and the phone.
Certicate To use HTTPS you need to upload a certicate to the phone.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > WEB INTERFACE (5,2,7).
SETTINGS
VAD
Voice Activity Detection (speech detection) is a technique used in speech processing
wherein the presence or absence of human speech is detected in regions of audio. In
VoIP applications, the main use of VAD is to avoid unnecessary coding and transmission
of silence packets, saving on computation and on network bandwidth.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > MEDIA > VAD (5,2,4,2).
DTMF
DTMF (Dual-tone multi-frequency) signalling is used for telephone signalling over the
line to the phone switch or PBX.
If the device itself generates the tones and it is sent in the voice-frequency band, the
method is called Inband. Using VoIP, this is not the best method. Low bit rate codecs
may corrupt the signalling tones and make them hard to recognize for the switch.
RFC 2833 is a method for carrying DTMF-signalling in RTP packets using a separate
RTP payload format. With this method a PSTN gateway reproduces the DTMF tones
sent from the end device.
With SIP Info the DTMF signalling is sent as SIP requests. The SIP switch creates the
tones if the call is transferred to the PSTN.
Use RFC 2833 or SIP Info as preferred methods. Switch to inband only if you encounter
problems using DTMF signalling with your PBX/SIP switch.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > MEDIA > DTMF SIGNALLING (5,2,4,3).

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TIME & REGION
Select Settings > Time & Region.
Time
Enable NTP NTP (Network Time Protocol) is a protocol for distributing the
Coordinated Universal Time (UTC) by means of synchronizing
the clocks of computer systems over packet-switched, variable-
latency data networks.
Time This eld shows the actual time if NTP is enabled. Otherwise
enter the correct time (hh:mm:ss) and save the setting.
Date This eld shows the actual date if NTP is enabled. Otherwise
enter the correct date (yyyy-mm-dd) and save the setting.
Timezone Select the UTC time zone in your country.
Daylight saving Select the Yes radio button if DST (Daylight Saving Time or
Summer Time) is currently used in your country. Note that this
setting only adjusts the time one hour, and do not change the
time automatically when the DST starts and ends.
NTP Server The NTP pool is a dynamic collection of networked computers
that volunteer to provide highly accurate time via NTP to clients
worldwide. The machines that are ”in the pool” are part of the
pool.ntp.org domain as well as of several subdomains divided by
geographical zone and are distributed to NTP clients via round
robin DNS.
On phone: MENU > SETTINGS > ADVANCED > (PIN) > TIME (5,2,5).
SETTINGS
Region
Select the region where you are. This setting affect the signalling (disconnect tone, busy
tone, etc).
On phone: MENU > SETTINGS > ADVANCED > (PIN) > REGION (5,2,6).
PROVISIONING
See “PROVISIONING – UPGRADE AND CONFIGURATION” on page 31.
SETTINGS

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31
SYSTEM
Select Settings > System.
Application restart
The Restart button restarts the phone application. This takes less than 30 seconds.
On phone: MENU > SYSTEM > RESTART (6,1).
System reboot
The Reboot button reboots the conference phone. The starting procedure may take
about two minutes.
On phone: MENU > SYSTEM > REBOOT (6,2).
System conguration
The Reset button resets Konftel 300IP to factory default settings. Every personal set-
ting including account information is erased.
On phone: MENU > SYSTEM > FACTORY RESET (6,3).
Hard reset to factory settings
See page 45 about resetting the phone if you have forgotten the Admin PIN code.
SETTINGS
Select Settings > Provisioning.
FIRMWARE UPGRADE
Konftel 300IP is easiest upgraded via a computer connected to the same network. Via
the web interface, you can check if a later version is available, and then automatically
install it.
It is also possible to download the latest version, via the Konftel website (www.konftel.
com/300ip), and then install the le via the web interface.
Click on the Check Now button.
Compare the latest software version with the current version (shown on the same
page).
If you choose to upgrade, select version in the list box and click on the Upgrade
button.
The browser window and the Konftel 300 iP display shows that the upgrading has begun.
The download and installation can take several minutes. Do not interrupt the
upgrade and do not disconnect any plug on Konftel 300IP during the upgrade. An
interrupted upgrade can make the conference phone unusable.
PROVISIONING – UPGRADE AND CONFIGURATION

32
33
When installation is complete, the text “Upgrade Complete. The unit will be
rebooted.” is shown in your browser, and after a while you hear the Konftel music
signature, which indicates that the conference phone has started.
Upgrading from downloaded le
It is possible to download a rmware le from www.konftel.com/300ip, and install it
into the Konftel 300IP from the local hard disk.
Download the rmware le from www.konftel.com/300ip.
Click on the Browse… button and locate and choose the downloaded le.
Click on the Check Now button.
After upgrading
If DHCP is used in the network, the IP address may have been changed. If the web
browser looses contact with Konftel 300IP, check the IP address on the conference
phone (see “USING THE WEB INTERFACE” on page 6).
USING A CONFIGURATION FILE
It is possible to save a conguration xml le to be used as:
• Backup (i.e. if the system has been reset to factory default)
• Conguration interface (there are some settings that are not congurable via the
web interface)
• Management tool (export, edit, and import settings to a set of phones instead of
doing the settings on each phone)
The structure of the xml le is as follows:
<locale>
<region>
<logging>
<level> The phone application logs messages to log facility LOCAL0.
Log level 1–5 (equivalent to Fatal–Trace)
<log_sip> Log sip messages to log facility LOCAL1. Default is true.
<remote_log> Log messages to a remote log server. Default is false.
<remote_host /> Remote log server.
<network>
<net>
<dhcp> Specify if DHCP should be used to obtain network settings. If so, the
other network settings won’t be used.
<ip> Specify the IP address of the KT300IP.
<netmask> The netmask of the IP address.
<gateway> Specify the default gateway to be used.
<dns /> Specify at most two Domain Name Servers to be used.
<dns />
<hostname> Specify host name.
<domain /> Specify domain name.
<time>
<ntp>
<timezone>
<daylight_save>
<ntps>
PROVISIONING – UPGRADE AND CONFIGURATIONPROVISIONING – UPGRADE AND CONFIGURATION

34
35
<sip>
<udp_transport> Specify if UDP shall be used as transport.
<udp_port> Specify the UDP port to listen to.
<tcp_transport> Specify if TCP shall be used as transport.
<tcp_port> Specify the TCP port to listen to.
<tls_transport> Specify if TLS shall be used as transport.
<sips_transport> Specify if SIPS shall be used as transport.
<tls_port> Specify the TLS port to listen to.
<rtp_port> Specify the start port for RTP trafc.
<outbound_proxy /> Specify the URL of outbound proxies to visit for all outgoing requests.
The outbound proxies will be used for all accounts, and it will be used to
build the route set for outgoing requests. The nal route set for outgoing
requests will consists of the outbound proxies and the proxy congured
in the account.
<use_stun> Use Simple Traversal of UDP through NATs (STUN) for NAT traversal.
Default is no.
<stun_domain /> Specify domain name to be resolved with DNS SRV resolution to get
the address of the STUN servers. Alternatively application may specify
stun_host and stun_relay_host instead.
<stun_host /> Specify STUN server to be used, in ”HOST[:PORT]” format. If port is not
specied, default port 3478 will be used.
<use_turn> Use Traversal Using Relay NAT (TURN) for NAT traversal. Default is no.
<turn_host /> Specify TURN relay server to be used.
<turn_tcp> Use TCP connection to TURN server. Default is false.
<turn_user /> TURN username.
<turn_passwd /> TURN password.
<nat_type_in_sdp> Support for adding and parsing NAT type in the SDP to assist trouble-
shooting. The valid values are:
0: no information will be added in SDP, and parsing is disabled
1: only the NAT type number is added
2: add both NAT type number and name
<require_100rel> Specify whether support for reliable provisional response (100rel and
PRACK) should be required by default. Note that this setting can be
further customized in account conguration.
<use_srtp> Specify default value of secure media transport usage. Note that this
setting can be further customized in account conguration.
0: SRTP will be disabled, and the transport will reject RTP/SAVP offer.
1: SRTP will be advertised as optional and incoming SRTP offer will be
accepted.
2: The transport will require that RTP/SAVP media shall be used.
PROVISIONING – UPGRADE AND CONFIGURATION
<srtp_secure_signaling>Specify whether SRTP requires secure signalling to be used. This option
is only used when use_srtp option above is non-zero. Note that this
setting can be further customized in account conguration.
0: SRTP does not require secure signalling
1: SRTP requires secure transport such as TLS
2: SRTP requires secure end-to-end transport (SIPS)
<codec>
<type> Codec type
<name> Codec name
<prio> Codec priority (0-4)
<dtmf> DTMF signalling. Default is 2.
0: In-band
1: SIP message
2: RTP message
<no_vad> Disable VAD? Default is VAD enabled.
<ec_tail> Echo canceller tail length, in milliseconds.
<enable_ice> Enable ICE?
<enable_relay> Enable ICE relay?
<enable_presence> Enable the use of presence signalling.
<tls>
<tls_password /> Password for the private key
<tls_method> TLS protocol method from pjsip_ssl_method, which can be:
0: Default (SSLv23)
1: TLSv1
2: SSLv2
3: SSLv3
23: SSLv23
<tls_verify_server> Verify server certicate.
<tls_verify_client> Verify client certicate.
<tls_require_client_cert> Require client certicate.
<tls_neg_timeout> TLS negotiation timeout in seconds to be applied for both outgoing and
incoming connections. If zero, no timeout is used.
<account>
<valid> If this account information is valid or not.
<name> User dened name of the account
<id> The full SIP URL for the account.
<registrar> This is the URL to be put in the request URI for the registration.
PROVISIONING – UPGRADE AND CONFIGURATION

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<publish_enabled> If this ag is set, the presence information of this account will be
Published to the server where the account belongs.
<initial_auth> If this ag is set, the authentication client framework will send an empty
Authorization header in each initial request.
<initial_algo /> Specify the algorithm to use when empty Authorization header is to be
sent for each initial request (see above).
<pidf_tuple_id /> Optional PIDF tuple ID for outgoing PUBLISH and NOTIFY. If this value
is not specied, a random string will be used.
<force_contact /> Optional URI to be put as Contact for this account. It is recommended
that this eld is left empty, so that the value will be calculated automati-
cally based on the transport address.
<require_100rel> Specify whether support for reliable provisional response (100rel and
PRACK) should be required for all sessions of this account.
<proxy_uri /> Optional URI of the proxies to be visited for all outgoing requests that are
using this account (REGISTER, INVITE, etc).
<reg_timeout> Optional interval for registration, in seconds. If the value is zero, default
interval will be used.
<cred> Array of credentials. If registration is desired, normally there should be
at least one credential specied, to successfully authenticate against the
service provider. More credentials can be specied, for example when
the requests are expected to be challenged by the proxies in the route
set.
<realm> Realm. Use ”*” to make a credential that can be used to authenticate
against any challenges.
<scheme /> Scheme (e.g. ”digest”).
<username> Authentication name
<cred_data_type> Type of data (0 for plaintext password).
<cred_data> The data, which can be a plaintext password or a hashed digest.
<auto_update_nat> This option is useful for keeping the UDP transport address up to date
with the NAT public mapped address. When this option is enabled and
STUN is congured, the library will keep track of the public IP address
from the response of REGISTER request. Once it detects that the
address has changed, it will unregister current Contact, update the UDP
transport address, and register a new Contact to the registrar.
<ka_interval> Set the interval for periodic keep-alive transmission for this account. If
this value is zero, keep-alive will be disabled for this account. The keep-
alive transmission will be sent to the registrar’s address, after successful
registration.
<ka_data /> Specify the data to be transmitted as keep-alive packets. Default: CR-LF.
<use_srtp> Specify whether secure media transport should be used for this account.
0: SRTP will be disabled, and the transport will reject RTP/SAVP offer.
PROVISIONING – UPGRADE AND CONFIGURATION
1: SRTP will be advertised as optional and incoming SRTP offer will be
accepted.
2: The transport will require that RTP/SAVP media shall be used.
<srtp_secure_signaling> Specify whether SRTP requires secure signalling to be used. This
option is only used when use_srtp option above is non-zero.
0: SRTP does not require secure signalling
1: SRTP requires secure transport such as TLS
2: SRTP requires secure end-to-end transport (SIPS)
<account>
Same as above for account 2
<provisioning>
<upgrade>
<url> Place to nd software upgrades. The supported URL types are: HTTP,
FTP, and TFTP.
<www>
<enable_https> Secure communication to the 300IP web server. Default is false.
Export conguration
Click on the Export button.
The conguration le is shown in the web browser.
Choose to save the page as an xml le.
The xml le is as default saved in your folder for downloaded les.
If needed, edit the xml le in a suitable editor.
Import conguration
Click on the Browse… button.
Select the xml le and choose to open it.
Click on the Import button.
PROVISIONING – UPGRADE AND CONFIGURATION