Akuvox SPD-R25 User manual

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Akuvox SDP-R25 User Manual
12/05/2014

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Table of Contents
Production Overview................................................................... 3
1. Production Description.................................................................................... 3
2. Features............................................................................................................3
Configuration............................................................................... 5
1. Web Login.........................................................................................................5
1.1. Obtaining the IP address....................................................................... 5
1.2. Login the Web........................................................................................5
2. Status................................................................................................................6
3. Language.......................................................................................................... 8
4. Network configuration..................................................................................... 8
5. Account.......................................................................................................... 10
6. Push Button.................................................................................................... 14
7. Phone............................................................................................................. 15
7.1. Call Feature.......................................................................................... 15
7.2. Voice.................................................................................................... 16
7.3. Country Ringtone.................................................................................18
8. PhoneBook..................................................................................................... 19
8.1. Call Log.................................................................................................19
9. Security...........................................................................................................20
9.1. Web Password Modify.........................................................................20
9.2. Web Server Certificate.........................................................................20
10. Upgrade...................................................................................................... 22
10.1. Basic upgrade................................................................................... 22
10.2. Advanced Upgrade........................................................................... 23

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Production Overview
1. Production Description
The Akuvox SPD-R25 is the video door phone, that you can connect with your Akuvox IP
Phones for remote unlock control and monitoring. You can operate the indoor handset to
communicate with visitors via voice and video, and unlock the door if you wish. It’s
applicable in apartment, villas, Office, building and so on.
2. Features
Key Features
HD Voice
Compatible with Asterisk and Broadsoft platforms
Physical Features
Camera: 2.0 Mega Pixels
White balance:Auto

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Lens: 4.0mm/F2.8
Viewing Angle (Diagonal): 50
Minimum Illumination: 1 LUX (without LED Illumination)
LED: 6 LEDs
Power Requirement: DC12V
Operating Temperature; -30C ~ 40C
Weight: 180g
Size (WXHXD)185 x 68 x 50 mm
Phone Features
Video resolution: 320 x 240, with 20pics per second
Wide angle lens and IR LEDs for night vision Crystal sound quality
Remote door opening
Integrated microphone and speaker
Water-proof outdoor unit: IP55
Support all the VoIP Phones
IP-PBX Features
Video Codec: H.264
Audio Codec: PCMU
VAD, CNG , Echo Canceller
Network Features
SIP v1(RFC2543), V2(RFC3261)
Static IP/DHCP for IP conguration
3 DTMF modes: In-Band, RFC2833, SIP INFO
HTTP/HTTPS Web Server for Management
NTP for Auto Time Setting
TFTP/FTP/HTTP/HTTPS client API
Administration Features
Auto provisioning using FTP/TFTP/HTTP/HTTPS/PnP
Dial through IP PBX Using Phone Number
Dial through IP PBX Using URL Address
Conguration Managements with Web, keypad on the phone, and Auto Provisioning
Security Features
Support HTTPS (SSL)
Support SRTP for Voice Data Encryption
Support Login for Administration
Sip Over TLS

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Configuration
1. Web Login
1.1. Obtaining the IP address
The Akuvox R25 uses Static IP by default, and the default IP address is 192.168.1.100.
If the IP address is unknown, press the call button when the door phone is initialing, after a
short period of time, the phone will announce its IP.
1.2. Login the Web
Open a Web Browser, enter the corresponding IP address. Then, type the default user name
and password to log in. The default User Name and Password are as below,
User name: admin
Password: admin

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2. Status
Status, including product information, network information and Account information, can be
viewed from, Status > Basic.

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Sections Description
Product Information To display the device’s information such as Model name,
MAC address (IP device’s physical address), Firmware
version and Hardware firmware.
Network Information To display the device’s Networking status(LAN Port),such
as Port Type(which could be DHCP/Static/PPPoE), Link
Status, IP Address, Subnet Mask, Gateway, Primary DNS
server, Secondary DNS server, Primary NTP server
and Secondary NTP server(NTP server is used to
synchronize time from INTERNET automatically).
Account Information To display device’s Account information and Registration
status (account username, registered server’s address,
Register result).

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3. Language
Web Language can be configured from, Phone > Time/Lang.
Select the desire language from the pull-down list of Type. The default language is English.
4. Network configuration
To configure the basic network settings, Go to Network > Basic.
The static IP is set as default, and its IP address is 192.168.1.100.

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Sections Description
LAN Port To display and configure LAN Port settings.
DHCP: If selected, IP phone will get IP address, Subnet
Mask, Default Gateway and DNS server address from
DHCP server automatically.
Static IP: If selected, you have to set IP address,
Subnet Mask, Default Gateway and DNS server
manually.
PPPoE: Use PPPoE username/password to connect to
PPPoE server.
For advanced settings, go to Network > Advanced
Sections Description
Local RTP To display and configure Local RTP settings.
Max RTP Port: Determine the maximum port that RTP
stream can use.
Min RTP Port: Determine the minimum port that RTP
stream can use.
TR069 To display and configure TR069 settings.
Active: To enable or disable TR069 feature.

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Version: To select supported TR069 version (version
1.0 or 1.1).
ACS/CPE: ACS is short for Auto configuration servers
as server side, CPE is short for Customer-premise
equipment as client side devices.
URL: To configure URL address for ACS or CPE.
User name: To configure username for ACS or CPE.
Password: To configure Password for ACS or CPE.
Periodic Inform: To enable periodically inform.
Periodic Interval: To configure interval for periodic
inform.
Note: TR-069(Technical Report 069) is a technical
specification entitled CPE WAN Management Protocol
(CWMP).It defines an application layer protocol for
remote management of end-user devices.
5. Account
To configure your SIP account, go to Account > Basic.

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Sections Description
SIP Account To display and configure the specific Account settings.
Status: To display register result.
Display Name: Which is sent to the other call party for
displaying.
Register Name: Allocated by SIP server provider, used
for authentication.
User Name: Allocated by your SIP server provide,
used for authentication.
Password: Used for authorization.
SIP Server 1 To display and configure Primary SIP server settings.
Server IP: SIP server address, it could be an URL or IP
address.

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Registration Period: The registration will expire after
Registration period, the IP phone will re-register
automatically within registration period.
SIP Server 2 To display and configure Secondary SIP server settings.
This is for redundancy, if registering to Primary SIP server
fails, the IP phone will go to Secondary SIP server for
registering.
Note: Secondary SIP server is used for redundancy, it can
be left blank if there is not redundancy SIP server in user’s
environment.
Outbound Proxy Server To display and configure Outbound Proxy server settings.
An outbound proxy server is used to receive all initiating
request messages and route them to the designated SIP
server.
Note: If configured, all SIP request messages from the IP
phone will be sent to the outbound proxy server
forcefully.
Transport Type To display and configure Transport type for SIP message
UDP: UDP is an unreliable but very efficient transport
layer protocol.
TCP: Reliable but less-efficient transport layer
protocol.
TLS: Secured and Reliable transport layer protocol.
DNS-SRV: A DNS RR for specifying the location of
services.
NAT To display and configure NAT(Net Address Translator)
settings.
STUN: Short for Simple Traversal of UDP over NATS, a
solution to solve NAT issues.
Note: By default, NAT is disabled.
For advance account settings, go to Account > Advanced.

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Sections Description
Codecs To display and configure available/unavailable codecs list.
Codec means coder-decoder which is used to transfer
analog signal to digital signal or vice versa.
Familiar codecs are PCMU(G711U), PCMA(G711A), G722
(wid-bandth codecs), G723,G726,G729 and so on.
Call To display and configure call-related features.
Max Local SIP Port: To configure maximum local sip
port for designated account.
Min Local SIP Port: To configure minimum local sip
port for designated account.
Encryption To enable or disabled SRTP feature.
Voice Encryption (SRTP): If enabled, all audio signal
(technically speaking it’s RTP streams) will be
encrypted for more security.
NAT To display NAT-related settings.
UDP Keep Alive message: If enabled, IP phone will

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send UDP keep-alive message periodically to router to
keep NAT port alive.
UDP Alive Msg Interval: Keepalive message interval.
Rport: Remote Port, if enabled, it will add Remote
Port into outgoing SIP message for designated
account.
6. Push Button
To configure Push Button, go to Push Button.
Sections Description
Push Botton To configure the destination number you want to contact
with.
DTMF Code To select the desired DTMF Code
Lock Reset To set the lock reset time

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Max Call Time To configure the max call time
Push to Hang up To enable or disable the Push to Hang up function
7. Phone
7.1. Call Feature
Call feature can be configured from, Phone > Call Feature.
Sections Description
Call Waiting To enable or disable Call Waiting.
Call Waiting Enable: If enabled, it allows IP phones to
receive a new incoming call when there is already an
active call.
Call Waiting Tone: If enabled, it allows IP phones to
play the call waiting tone to the waiting callee.

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Auto Redial Auto redial allows IP phones to redial an unsuccessful call
for designated times within designated interval.
Auto Redial: To enable or disable auto redial feature.
Auto Redial Interval: Determine the interval between
two consecutive attempts.
Auto Redial Times: Determine how many times to
redial.
DND
DND(Do Not Disturb) allows IP phones to ignore any
incoming calls.
Return Code when DND: Determine what response
code should be sent back to server when there is an
incoming call if DND on.
DND On Code: The Code used to turn on DND on
server’s side, if configured, IP phone will send a SIP
message to server to turn on DND on server side if
you press DND when DND is off.
DND Off Code: The Code used to turn off DND on
server’s side, if configured,IP phone will send a SIP
message to server to turn off DND on server side if
you press DND when DND is on.
Remote Control Remote Control allows specific host to interact with IP
phone by sending HTTP or HTTPS requests. The specific
action could be answering an incoming call, hangup an
ongoing call and so on.
Allowed Access IP List: To configure the allowed host
address.
Note: For now, IP phone can only support IP address, IP
address list and IP address pattern as allowed hosts
Others
Return Code When Refuse: Allows user to assign
specific code as return code to SIP server when an
incoming call is rejected.
Auto Answer Delay: To configure delay time before an
incoming call is automatically answered.
7.2. Voice
Voice can be configured from, Phone > Voice

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Sections Description
Echo Canceller Echo Canceller: To remove acoustic echo from a voice
communication in order to improve the voice quality .
VAD (Voice Activity Detection): Allow IP phone to
detect the presence or absence of human speech
during a call. When detecting period of “silence”, VAD
replaces that silence efficiently with special packets
that indicate silence is occurring. It can facilitate
speech processing, and deactivate some processes
during non-speech section of an audio session. It can
avoid unnecessary coding or transmission of silence
packets in VoIP applications, saving on computation
and network bandwidth.
CNG (Comfort Noise Generation): Allow IP phone to
generate comfortable background noise for voice
communications during periods of silence in a
conversation. It is a part of the silence suppression or
VAD handling for VoIP technology. CNG, in
conjunction with VAD algorithms, quickly responds
when periods of silence occur and inserts artificial
noise until voice activity resumes. The insertion of
artificial noise gives the illusion of a constant
transmission stream, so that background sound is
consistent throughout the call and the listener does
not think the line has released.

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Jitter Buffer Jitter buffer is a shared data area where voice packets can
be collected, stored, and sent to the voice processor in
even intervals. Jitter is a term indicating variations in
packet arrival time, which can occur because of network
congestion, timing drift or route changes. The jitter buffer,
located at the receiving end of the voice connection,
intentionally delays the arriving packets so that the end
user experiences a clear connection with very little sound
distortion.
IP phones support two types of jitter buffers: fixed and
adaptive.
Fixed: Add the fixed delay to voice packets. You can
configure the delay time for the static jitter buffer on IP
phones.
Adaptive: Capable of adapting the changes in the
network's delay. The range of the delay time for the
dynamic jitter buffer added to packets can be also
configured on IP phones.
Mic Volume To configure Microphone volume
7.3. Country Ringtone
Country Ringtone can be configured from, Phone > Tone.
Select the desired country ringtone from the pull-down list of Select Country.

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8. PhoneBook
8.1. Call Log
Sections Description
Call History To display call history records.
Available call history type are All calls, Dialed calls,
Received calls, Missed calls, Forwarded calls.
HangUp: To click to hangup ongoing call on the IP phone.
Note: For “HangUp” feature, you need to have the remote
control privilege to control IP phone via Web UI. Please
refer to section “Remote Control” in the Web
UI->Phone->Call Feature page.

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9. Security
9.1. Web Password Modify
To modify web passoword, go to Security > Basic
Sections Description
Web Password Modify
To modify user’s password.
Current Password: The current password you used.
New Password: Input new password you intend to
use.
Confirm Password: Repeat the new password.
Note: For now, IP phone can only support user admin.
9.2. Web Server Certificate
To check or upload your web server certificate, go to Security > Advanced