Akuvox R25 User Manual
Contents
1. Sip Over TLS 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 Login Status User Name Password Remember Co Username Password 2 Status Status including product information network information and Account information can be viewed from Status gt Basic Product Information Model MAC Address Firmware Version Hardware Version Network Information LAN Port Type LAN Link Status LAN IP Address LAN Subnet Mask LAN Gateway LAN DNS1 LAN DNS2 Primary NTP Secondary NTP Account Information Sections Product Information Note Max length of characters for input box 255 Brosdsoft Phonebook server address 127 Remote Phonebook URL amp DHCP Auto AUTOP Manual Update Server Connected URL 192 168 1 40 63 The rest of input boxes 255 255 255 0 192 168 1 1 Warning 102 168 1 1 SP R25 Dc 11 05 00 17 5d 25 0 1 27 25 0 0 0 0 0 0 0 Field Description D pool ntp org 1 pool ntp org 101 192 168 1 126 Registration Fsiled2. Password Field Description Warning Sections Description LAN Port To display and configure LAN Port settings e DHCP If selected IP phone will get IP address Subnet Mask Default Gateway and DNS server address from DHCP server automatically O 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 gt 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 TRO69 To display and configure TRO69 settings O Active To enable or disable TRO69 feature O Version To select supported TRO69 version version 1 0 or 1 1 O 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 pro
3. Speaker Sensor Light Camera LED Light Push Button Call Button Microphone 2 Features gt Key Features HD Voice gt gt Crystal Sound Quality Wide Angle Lens and IR LEDS for Night Vision Remote Door Opening Water Proof Physical Features Camera 2 0 Mega Pixels White balance Auto 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 W x H x D 185x68 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 gt IP PBX Features Video Codec H 264 Audio Codec PCMU VAD CNG Echo Canceller gt Network Features gt SIP V1 RFC2543 V2 RFC3261 Static IP DHCP for IP configuration 3 DTMF Modes In Band RFC2833 SIP INFO HTTP HTTPS Web Server for Management NTP for Auto Time Setting TFTP FTP HTTP HTTPS Protocols Administration Features Auto Provisioning Using FTP TFTP HTTP HTTPS PnP Dial through IP PBX Using Phone Number Dial through IP PBX Using URL Address Configuration Managements with Web Keypad on the Phone and Auto Provisioning gt Security Features Support HTTPS SSL Support Login for Administration
4. stop packets capturing or to export captured Packet file O Start To start capturing all the packets file sent or received from IP phone O Stop To stop capturing packets Note IP phone will save captured packets file to a temporary file this file maximum size is 1M mega bytes and will top capturing once reaching this maximum size 21 Others To display or configure others features from this page O Config file To export or import configure file for IP phone 10 Security 10 1 Web Password Modify To modify web password go to Security gt Basic Web Password Modify User Name Note Max length of characters for input Current Password box New Password 255 Broadsoft Phonebook server address 127 Remote Phonebook URL amp AUTOP Manual Update Server URL 63 The rest of input boxes Confirm Password Warning Field 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 10 2 Web Server Certificate To check or upload your web server certificate go to Security gt Advanced 22 Web Server Certificate Note dd s S E E Max length of characters for input 1 Ringslink Ringslink Sun Jun 27 07 14 32 2037 be
5. Code 0 M 127 Remote Phonebook URL amp AUTOP Manual Update Server Lock Reset URL g Relay action time 500 Y 63 The rest of input boxes Max Call Time Warning Max Call Time 5 2 30Minutes Field Description Push to Hang up Push to Hangup Enabled v L Submit cancel Sections Description Push Button To configure the destination number or IP you want to contact with DTMF Code To select the desired DTMF Code Lock Reset To set the lock reset time 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 Voice Voice can be configured from Phone gt Voice 15 Echo Canceller Echo Canceller VAD CNG Jitter Buffer Jitter Type Min Delay Nominal Delay Max Delay Mic Volume Hand Free Volume Sections Submit Note Enabled x Max length of characters for input Disabled M box Enabled v 255 Broadsoft Phonebook server address 127 Remote Phonebook URL amp Fed Y AUTOP Manual Update Server URL 0 1000ms 63 The rest of input boxes 120 0 1000ms Warning 0 1000ms g Field Description 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
6. None None UnRegistered None None UnRegistered Description 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 DNSserver Secondary DNS server Primary NTP server and Secondary NTP server NTP server is used to synchronize timefrom INTERNET automatically Account Information To display device s Account information and Registration status account username registered servers address Register result 3 Language Web Language can be configured from Phone gt Time Lang Note Max length of characters for input eee 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 gt Basic The static IP is set as default and its IP address is 192 168 1 100 LAN Port Note DHCP Max length of characters for input Static IP box IP Address l192 168 1 100 255 Broadsoft Phonebook server a address 127 Remote Phonebook URL amp Default Gateway AUTOP Manual Update Server LAN DNS1 URL LAN DNS2 po 63 The rest of input boxes Subnet Mask 255 255 255 0 PPPoE User Name
7. go to Account gt Advanced 13 Call Max Local SIP Port 5062 1024 65535 Min Local SIP Port s062 _ 1024 65535 Encryption Voice Encryption NAT UDP Keep Alive Messages UDP Alive Msg Interval RPort Sections Description Codecs To display and configure available unavailable codecs list O Codec means coder decoder which is used to transfer analog signal to digital signal or vice versa O Familiar codecs is PCMU G711U Call To display and configure call related features O Max Local SIP Port To configure maximum local sip port for designated account O Min Local SIP Port To configure minimum local sip port for designated account Encryption To enable or disabled SRTP feature O 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 send UDP keep alive message periodically to router to keep NAT port alive UDP Alive Msg Interval Keepalive message interval O Rport Remote Port if enabled it will add Remote Port into outgoing SIP message for designated account 14 6 Push Button To configure Push Button go to Push Button Push Button Note o Key Number Te poesie Push Button 192 168 1 101 box gt Account DTMF Code 255 Broadsoft Phonebook server address gt Network DTMF
8. 1 126 Unknown 192 168 1 126 Warning Dialed 2014 08 19 02 17 49 101 192 168 1 126 Unknown 1000192 168 1 126 Dialed 2014 08 19 02 15 33 101 192 168 1 126 Unknown 100 192 168 1 126 Field Description Dialed 2014 08 19 02 14 24 101 192 168 1 126 Unknown 1000192 168 1 126 Dialed 2014 08 19 02 13 39 101 192 168 1 126 Unknown 100 192 168 1 126 Dialed 2014 08 19 02 11 52 101 192 168 1 126 Unknown 100 192 168 1 126 Received 2014 08 19 02 11 40 101 192 168 1 126 100 100 192 168 1 126 Dialed 2014 08 19 02 10 35 101 192 168 1 126 Unknown 100 192 168 1 126 SR ED CT ES CI 2 3 4 5 6 Z 8 9 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 Ul gt Phone gt Call Feature page 18 9 Upgrade 9 1 Basic upgrade To upgrade your device go to Upgrade gt Basic Upgrade Firmware Version Hardware Version Reboot Sections Reset To Factory Setting EXA RAIN Submit Cancel Note Max length of characters for input 25 0 1 27 box 25 0 0 0 0 0 0 0 255 Broadsoft Phoneb
9. Akuvox SDP R25 User Manual 05 06 2015 Content Production OVELVIEW c cccccccccccccccccccccccccccccccccccccccccccccccccccccccccccsecs 4 1 Production Description ss 4 DAROCA TUTO AAA AA AA ee ns en M ne dt ie 4 Configurations dd L Web IO A A ceded A tee ado LN 7 1 1 Obtaining the IP address 7 1 2 Login the Web tc theses ce eae eles terse Teed Pie eee 7 Zatanna As 8 3 langy geca eraa ne ibn Lens 10 4 Network configuration ss 10 5 ACCOUNT rer sn Eea eE sr ea tree den EEEE is 12 6 PUSH BUON sisi mnt annuaires 16 A ES see 17 7 1 Call Reature ss id ET R Ta A at 17 LD NOICR nr A ees NT coed eH esd en ee RU 18 7 3 Country Ringtone intimiste e descend sedeaasdhaedeedeaededans saeco 20 8 PIONS BOOK RS SES ne ne de eee 21 B21 Call Os nn en ire a fine ds 21 J Uperadess a 22 9 1 Basicupgrade iii cress tou Wedel aa aah x das aves EA 22 9 2 Advanced Upgrade 22 10 S CUrITVE ssh ten sement errant add tent seed 25 10 1 Web Password Modify a 25 10 2 Web Server Certificate sisi 25 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
10. CP option DHCP option If configured IP Phone will use designated DHCP option to get Auto Provisioning server s address via DHCP This setting require DHCP server to support corresponding option 20 Manual Update Server To display and configure manual update server s settings URL Auto provisioning server address O User name Configure if server needs an username to access otherwise left blank O Password Configure if server needs a password to access otherwise left blank Common AES Key Used for IP phone to decipher common Auto Provisioning configuration file O AES Key MAC Used for IP phone to decipher MAC oriented auto provisioning configuration file for example file name could be 0c1105888888 conf if IP phone s MAC address is 0c1105888888 Note AES is one of many encryption it should be configure only configure filed is ciphered with AES otherwise left blank AutoP To display and configure Auto Provisioning mode settings This Auto Provisioning mode is actually self explanatory For example mode Power on means IP phone will go to do Provisioning every time it powers on System Log To display system log level and export system log file O System log level From level 0 7 The higher level means the more specific system log is saved to a temporary file By default it s level 3 O Export Log Click to export temporary system log file to local PC PCAP To start
11. IP SIP server address it could be an URL or IP address 12 O 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 e UDP UDP is an unreliable but very efficient transport layer protocol e TCP Reliable but less efficient transport layer protocol e TLS Secured and Reliable transport layer protocol O 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
12. a clear connection with very little sound distortion O P 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 O 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 2 Country Ringtone Country Ringtone can be configured from Phone gt Tone Select the desired country ringtone from the pull down list of Select Country gt Push Button Select Country China v 17 8 PhoneBook 8 1 Call Log Call History v Hand Up Note eas CTT Max length of characters for input Dialed 2014 08 25 05 40 55 101 192 168 1 126 Unknown 100 192 168 1 126 bee Dialed 2014 08 19 03 56 53 101 192 168 1 126 Unknown 100 1921681126 255 Broadsoft Phonebook server Dialed 2014 08 19 02 57 48 101 192 168 1 126 Unknown 100 192 168 1 126 address Dialed 2014 08 19 02 57 25 101 192 168 1 126 Unknown 100 192 168 1 126 127 Remote Phonebook URL amp Dialed 2014 08 19 02 51 02 101 192 168 1 126 Unknown 100 192 168 1 126 e aan RS es Dialed 2014 08 19 02 30 33 101 192 168 1126 Unknown 100 192 168 1 126 G3 Tha vet bobos Dialed 2014 08 19 02 24 35 101 192 168 1 126 Unknown 100 192 168 1 126 Dialed 2014 08 19 02 19 38 101 192 168
13. 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 VolP 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 Jitter Buffer Jitter buffer is a shared data area where voice packets can be collected stored and sent to the voice processor in even 16 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
14. ook server Submit address Submit 127 Remote Phonebook URL amp AUTOP Manual Update Server URL 63 The rest of input boxes Warning Field Description Description Upgrade To select upgrading rom file from local or a remote server automatically Note Please make sure it s right file format for right model Firmware version To display firmware version firmware version starts with MODEL name Hardware Version To display Hardware version Reset to Factory Setting To enable you to reset IP phone s setting to factory settings Reboot To reboot IP phone remotely from Web UI 9 2 Advanced Upgrade To do the advanced upgrade for your device go to Upgrade gt Advanced 19 PNP Option PNP Config i Max length of characters for input DHCP Option Custom Option k128 254 Manual Update Server ttp 192 168 1 24 Schedule AutoP Immedistely Clear MD5 Submit Cancel System Log LogLevel Export Log PCAP PCAP Others Config File tgz PNP Option To display and configure PNP setting for Auto Provisioning PNP Plug and Play once PNP is enabled the phone will send SIP subscription message to PNP server automatically to get Auto Provisioning server s address By default this SIP message is sent to multicast address 224 0 1 75 PNP server address by standard DHCP Option To display and configure custom DH
15. s Web Server Certificate Upload 255 Broadsoft Phonebook server address BEN RE EAT Cancel 127 Remote Phonebook URL amp AUTOP Manual Update Server Client Certificate URC 63 The rest of input boxes Index IssueTo Issuer Expira Time 1 Warning Field Description 2 3 4 5 6 7 8 9 o Client Certificate Upload Index Auto v RO HR AE ar Sections Description Web Server Certificate To display or delete Certificate which is used when IP phone is connected from any incoming HTTPs request Note The default certificate could not be deleted Web Server Certificate To upload a certificate file which will be used as server certificate Upload Client Certificate To display or delete certificates which is used when IP phone is connecting to any HTTPs server Client Certificate Upload To upload certificate files which is used as client certificate 23
16. tocol for remote management of end user devices 5 Account To configure your SIP account go to Account gt Basic 11 SIP Account Status Registration Failed Account Account 1 M Account Active Enabled M Display Label 101 Display Name 101 Note Register Name 101 be Jongh ef characters for input User Name 101 255 Broadsoft Phonebook server Password eovcceee address 127 Remote Phonebook URL amp SIP Server 1 AUTOP Manual Update Server Server IP 192 168 1 126 Port 5060 URE 63 The rest of input boxes Registration Period 1800 30 65535s SIP Server 2 DAME Server IP Por _5060 Field Description Registration Period 1800 30 65535s Outbound Proxy Server Enable Outbound Disabled Nd Server IP Port 5060 Backup Server IP Port 5060 Transport Type Transport Type UDP M NAT NAT Disabled v Stun Server Address Port 3478 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 O 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 O Server
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