SIP Speaker iW30 - Speaker Fanvil - Free user manual and instructions

USER MANUAL SIP Speaker iW30 Fanvil

Black Fanvil audio player with visible grille and speaker grille (no text or symbols on device body)

SIP Speaker iW30

USER MANUAL

Document Version: <1.0>

Software Version: <2.1.1>

Release Date: <2018-5-3>

1 Directory

1 Directory....I
2 Picture.... III
3 Form......V
4 Safety Notices....1
5 Product introduction ....2
6 Start Using....3

6.1 Connecting the power supply and the network....3

6.1.1 Connecting network ....3
6.1.2 Port description....4

6.2 Quick Setting .... 5
6.3 Basic operation .... 5
6.4 Answer a call ....5
6.5 Volume 5
6.6 Video linkage....6

7 Page settings....7

7.1 Browser configuration .... 7
7.2 Password Configuration....7
7.3 Configuration via Web....8

7.3.1 System 8

7.3.1.1 Information 8
7.3.1.2 Account....9
7.3.1.3 Configurations....9
7.3.1.4 Upgrade....10
7.3.1.5 Auto Provision....11
7.3.1.6 FDMS....14
7.3.1.7 Tools 15

7.3.2 Network....18

7.3.2.1 Basic....18
7.3.2.2 VPN 17

7.3.3 Line 19

7.3.3.1 SIP 19
7.3.3.2 Basic setting....24

7.3.4 Intercom settings 26

7.3.4.1 Features....26

Fanvil

7.3.4.2 Audio 27
7.3.4.3 Video 29
7.3.4.4 MCAST 32
7.3.4.5 Action URL 35
7.3.4.6 Time/Date 36

8 Appendix....37

8.1 Technical parameters ...... 37
8.2 Basic functions....37
8.3 Schematic diagram....38
8.4 The radio terminal configuration notice....38

2 Picture

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Figure 5 8

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Figure 7 9

Figure 8 ......10

Figure 9 ......11

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Figure 22 24

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Figure 25 27

Figure 26 29

Figure 27 29

Figure 28 29

Figure 29 ....30

Figure 30 ......32

Figure 31 ......33

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Figure 33 35

Figure 34 36

Figure 35 38

Figure 36 ....38

Figure 37 39

3 Form

Diagram 1 ....4

Diagram 2 ....8

Diagram 3 9

Diagram 4 9

Diagram 5 ......10

Diagram 6 ....12

Diagram 7 ....14

Diagram 8 ......15

Diagram 9 ......16

Diagram 10 ......18

Diagram 11 ......21

Diagram 12 ......24

Diagram 13 ......26

Diagram 14 ......27

Diagram 15 ....30

Diagram 16 ....35

Diagram 17 ....36

Diagram 18 ......37

Safety Notices

1. Please use the specified power adapter. If special circumstances need to use the power adapter provided by other manufacturers, please make sure the voltage and current provided in accordance with the requirements of this product, meanwhile, please use the safety certificated products, otherwise may cause fire or get an electric shock.
2. When using this product, please do not damage the power cord, or forcefully twist it, Stretch pull or banding, and not to be under heavy pressure or between items, Otherwise may cause the power cord damage, thus lead to fire or get an electric shock.
3. Before use, please confirm the temperature and environment humidity suitable for the product work. (Move the product from air conditioning room to natural temperature, which may cause this product surface or internal components produce condense water vapor, please open power use it after waiting for this product is natural drying).
4. Non-technical staff not remove or repair, improper repair or may cause electric shock, fire or malfunction, etc., Which can lead to injury accident, and also can cause your product damage.
5. Do not use fingers, pins, wire and other metal objects, foreign body into the vents and gaps. It may cause current through the metal or foreign body, which even cause electric shock and injury accident. If any foreign body or objection falls into the product please stop usage.
6. Please do not discard the packing bags or stored in places where children could reach, if children trap his head with it, may cause nose and mouth blocked, and even lead to suffocation.
7. Please use this product with normal usage and operating, in bad posture for a long time to use this product may affect your health.
8. Please read the above safety notices before installing or using this phone. They are crucial for the safe and reliable operation of the device.

5 Product introduction

This product is a complete digital network intercom equipment, its core part adopts mature VOIP solutions (Broadcom 1190), the performance is stable and reliable; Paging system can use g.711 and g.722 with loud and clear voice; Besides, simple installation, low standby power consumption.

Black electronic device with visible grille and speaker grille, no text or symbols on body

6 S t a r t U s i n g

Before you start to use equipment, please make the following installation:

6.1 Connecting the power supply and the network

6.1.1 Connecting network

In prior to this step, please check if your network can work normally and have capacity of broadband internet access.

- Broadband Router

Connect one end of the network cable to the intercom WAN port, the other end is connected to your broadband router's LAN port, so that the completion of the network hardware connections. In most cases, you must configure your network settings to DHCP mode.

graph LR
    A["Internet"] --> B["ADSL / Cable Modem"]
    B --> C["Broadband Router"]
    C --> D["Refriger"]

Figure1

- No Broadband Router

Connect one end of the network cable to the intercom WAN port, the other end is connected to the broadband modem to your LAN port, so that the completion of the network hardware connections. In most cases, if you are using the cable broadband, you must configure your network settings to DHCP mode; if you are using the ADSL, you must configure your network settings to PPPoE mode.

graph LR
    A["Internet"] --> B["ADSL / Cable Modem"]
    B --> C["Server"]

Figure2

6.1.2 Port description

Diagram1

Icons	Description	Feature Picture
Power	DC Power Input port	Input Range:+12~+24V DC(Notice: do not connect the incorrect polarity)	[YYH]
WANWAN port		10M/100M Adaptive Ethernet port, connected to the network
LAN LAN Port		10M/100M Adaptive Ethernet port, connected to the computer (which can be configured to routing mode, or to bridge mode) or IPC camera.
NET	The Network Light	1, The network gets through, and the light put out2, The network cannot get through, and the light blink fast within 0.5s3, The network gets through but registration fail, and the light blink slowly with 1s
VOLUME /RST	button	1、Press and hold volume down button for 3 seconds; the door phone would report the IP address by voice, and the voice volume will go down by single press the button.2、Long press the volume up button for 10 seconds, the speaker issued a rapid beep, and then quickly press the “volume up” button three times, beep stopped. Wait 10 seconds, successfully switch to dynamic IP after the system automatically voice broadcast IP address. Switching again will become a fixed IP address, and the voice volume will go up by single press the button.3、Press the reset button for 3 seconds, the device automatically restarts and restores the factory configuration.
AUDIO	Audio output	Connect audio port to output the audio headphones or external speakers.

6.2 Quick Setting

The product provides a rich and complete function and parameter setting; users may need to have a network with SIP protocol in order to understand the related knowledge on behalf of all the significance of the parameters. In order to high quality voice service and low-cost advantage, allowing users to enjoy the facility brought fast, especially in the listed in this section the basic and necessary to set options users can quickly get started, no without understanding the complicated SIP protocol.

In this step, please confirm the Internet access can be normal operation and complete the connection to the network hardware. The intercom default for DHCP mode.

Long press # key for 3 seconds, device's IP address will be played on voice, or use the "iDoorPhoneNetworkScanner.exe" software to scan the IP address of the device.
✨ Log on to the WEB device configuration.
In a SIP page configuration service account, user name, parameters that are required for server address register.
You can set function parameters in the Webpage (Intercom->feature).
In the intercom Settings -> voice page setup the volume

#	IP Address	Serial Number	MAC Address	SW Version	Description
1	172.18.3.240	iW30	0c:38:3e:1f:c2:26	2.1.1.3488	SIP Speaker

Figure3

6.3 Basic operation

6.4 Answer a call

When call coming, the device will automatically answer, if the “Auto Answer Timeout” was set, user will hear the bell in the set time, automatic answer after a timeout.

6.5 Volume

If you are not satisfied with the default volume, please logon the web page of the device, go to Intercom Setting -> Audio page, to set the volume.

6.6 Video linkage

Use other manufacturers camera please connect to the switch, the device LAN Port interface can only connect the original camera.
◇ Landing page configuration camera user name, password, port number and other information. For more information, please refer to the Video settings

7 Page settings

7.1 Browser configuration

When the device and your computer successfully connected to the network, enter the IP address of the device on browsers. You can see the Webpage management interface the login screen.

Enter the user name and password and click [logon] button to enter the settings screen.

User: Password: Language: English Logon

Figure4

After configuring the equipment, remember to click “Apply” to save the configuration. If this is not done, the equipment will lose the modifications when it rebooted.

7.2 Password Configuration

There are two levels of access: root level and general level. A user with root level access can browse and set all configuration parameters, while a user with general level can set all configuration parameters except server parameters for SIP.

✨ Default user with general level:

■ Username: guest
■ Password: guest

✨ Default user with root level:

■ Username: admin
■ Password: admin

7.3 Configuration via Web

7.3.1 System

7.3.1.1 Information

Information	Account	Configurations	Upgrade	Auto Provision	FDMS	Tools
System Information
Model:		IW30
Hardware:		2.1
Software:		2.1.1.3488
Uptime:		00 : 44 : 55
Last uptime:		13:26:22
MEMInfo:		ROM: 0.8/8(M) RAM: 2.3/16(M)
System Time:		2018-05-31 10:34
Network
Network mode:		DHCP
MAC:		0c:38:3e:1f:c2:26
IP:		172.18.3.240
Subnet mask:		255.255.0.0
Default gateway:		172.18.1.1
SIP Accounts
Line 1	1001	Registered
Line 2	N/A	Inactive

Figure5

Diagram2

Information
Field Name Explanation
System Information	Display equipment model, hardware version, software version, uptime, Last uptime and MEMinfo.
Network	Shows the configuration information for WAN port, including connection mode of WAN port (Static, DHCP, PPPoE), MAC address, IP address of WAN port.
SIP Accounts	Shows the phone numbers and registration status for the 2 SIP LINES.

7.3.1.2 Account

Information Account Configurations Upgrade Auto Provision FDMS Tools System Network Line Intercom settings Change Web Authentication Password Old Password: New Password: Confirm Password: Apply Add New User Username Web Authentication Password Confirm Password Privilege Administrators Add User Accounts User Privilege admin Administrators Delete

Figure6

Diagram3

Account
Field Name	Explanation
Change Web Authentication Password
You Can modify the login password to the account
Add New User
You can add new user
User Accounts
Show the existing user information

7.3.1.3 Configurations

Information Account Configurations Upgrade Auto Provision FDMS Tools System Network Line Intercom settings Export Configurations Right click here to SAVE configurations in 'txt' format. Right click here to SAVE configurations in 'xml' format. Import Configurations Configuration file: Select Import Reset to factory defaults Click the [Reset] button to reset the phone to factory defaults. ALL USER'S DATA WILL BE LOST AFTER RESET! Reset

Figure7

Diagram4

Configurations
Field Name Explanation
Export Configurations	Save the equipment configuration to a txt or xml file. Please Right click on the choice and then choose “Save Link As.”
Import Configurations	Browse to the config file and press Update to load it to the equipment.
Reset to factory defaults	This will reset factory default settings and remove all configuration information.

7.3.1.4 Upgrade

Information Account Configurations Upgrade Auto Provision FDMS Tools System Software upgrade Current Software Version: 2.1.1.3488 System Image File Select Upgrade Network Line Intercom settings

Figure8

Diagram5

Upgrade
Field Name	Explanation
Software upgrade
Browse to the firmware and press Update to load it to the equipment.

7.3.1.5 Auto Provision

Information Account Configurations Upgrade Auto Provision FDMS Tools System Network Line Intercom settings Common Settings Current Configuration Version General Configuration Version CPE Serial Number 00100400FV02001000000c383e1fc226 Authentication Name Authentication Password Configuration File Encryption Key General Configuration File Encryption Key Save Auto Provision Information □ DHCP Option >> SIP Plug and Play (PnP) >> Stotic Provisioning Server >> TR069 >> Apply

Figure9

DHCP Option >> Option Value Option 66 Custom Option Value 66 (128~254)

Figure10

SIP Plug and Play (PnP) >> Enable SIP PnP Server Address 224.0.1.75 Server Port 5060 Transportation Protocol UDP Update Interval 1 Hour

Figure11

Static Provisioning Server >> Server Address 0.0.0.0 Configuration File Name Protocol Type FTP ✓ Update Interval 1 Hour Update Mode Disabled ✓

Figure12

TR069 >>

Enable TR069

Enable TR069 Warning Tone

ACS Server Type

ACS Server URL

ACS User

ACS Password

TLS Version:

INFORM Sending Period

STUN Server Addr

STUN Enable

Figure13

Diagram6

Auto Provision
Field Name Explanation
Common Settings
Current Configuration Version	Show the current config file's version. If the version of configuration downloaded is higher than this, the configuration will be upgraded. If the endpoints confirm the configuration by the Digest method, the configuration will not be upgraded unless it differs from the current configuration
General Configuration Version	Show the common config file's version. If the configuration downloaded and this configuration is the same, the auto provision will stop. If the endpoints confirm the configuration by the Digest method, the configuration will not be upgraded unless it differs from the current configuration.
CPE Serial Number Serial number of the equipment
Authentication Name	Username for configuration server. Used for FTP/HTTP/HTTPS. If this is blank the phone will use anonymous
Authentication Password	Password for configuration server. Used for FTP/HTTP/HTTPS.
Configuration File Encryption Key	Encryption key for the configuration file
General Configuration File Encryption Key	Encryption key for common configuration file
Save Auto Provision Information	Save the auto provision username and password in the phone until the server URL changes
DHCP Option
Option Value	The equipment supports configuration from Option 43, Option 66, or a Custom DHCP option. It may also be disabled.
Custom Option Value C	Custom option number. Must be from 128 to 254.
SIP Plug and Play (PnP)
Enable SIP PnP	If this is enabled, the equipment will send SIP SUBSCRIBE messages to a multicast address when it boots up. Any SIP server understand that message will reply with a SIP NOTIFY message containing the Auto Provisioning Server URL where the phones can request their configuration.
Server Address PnP Server	Address
Server Port PnP Server	Port
Transportation Protocol	PnP Transfer protocol - UDP or TCP
Update Interval Interval	time for querying PnP server. Default is 1 hour.
Static Provisioning Server
Server Address	Set FTP/TFTP/HTTP server IP address for auto update. The address can be an IP address or Domain name with subdirectory.
Configuration File Name	Specify configuration file name. The equipment will use its MAC ID as the config file name if this is blank.
Protocol Type Specify the Protocol type FTP, TFTP or HTTP.
Update Interval Specify	the update interval time. Default is 1 hour.
Update Mode	1. Disable - no update2. Update after reboot - update only after reboot.3. Update at time interval - update at periodic update interval
TR069
Enable TR069 Enable/	Disable TR069 configuration
ACS Server Type Select	Common or CTC ACS Server Type.
ACS Server URL ACS	Server URL.
ACS User User name for	ACS.
ACS Password ACS Password.
TR069 Auto Login	Enable/Disable TR069 Auto Login.
INFORM Sending Period	Time between transmissions of “Inform” is 3600 seconds.

7.3.1.6 FDMS

Information Account Configurations Upgrade Auto Provision FDMS Tools System Network Line Intercom settings FDMS Settings Enable FDMS FDMS Interval 3600 Doorphone Info Settings Community Name Building Number Room Number Apply

Figure14

Diagram7

FDMS Settings
Enable FDMS Enable/Disable FDMS configuration
FDMS Interval	The time to send sip Subscribe information to the FDMS server on a regular basis. Unit is in second.
Doorphone Info Settings
Community Name	The name of the community where the device is installed
Building Number	The name of the building where the equipment is installed
Room Number The name of the room where the equipment is installed

Information Account Configurations Upgrade Auto Provision FDMS Tools Syslog Enable Syslog Server Address 0.0.0.0 Server Port 514 SIP Log Level None SIP Log Level None Apply Network Packets Capture Start Auto Reboot Setting Reboot Mode Disable Fixed Time 2 (0=23) Uptime 72 (H) Sip Rag Fail Reboot Welling Time 180 (x) Network Fail Reboot Waiting Time 500 (x) Apply Reboot Phone Click [Reboot] button to restart the phone! Reboot

Figure15

Syslog provide a client/server mechanism for the log messages which is recorded by the system. The Syslog server receives the messages from clients and classifies them based on priority and type. Then these messages will be written into a log by rules which the administrator has configured.

There are 8 levels of debug information.

Level 0: emergency; System is unusable. This is the highest debug info level.

Level 1: alert; Action must be taken immediately.

Level 2: critical; System is probably working incorrectly.

Level 3: error; System may work incorrectly.

Level 4: warning; System may work correctly but needs attention.

Level 5: notice; It is the normal but significant condition.

Level 6: Informational; It is the normal daily messages.

Level 7: debug; Debug messages normally used by system designer. This level can only be displayed via telnet.

7.3.1.7 Tools

Diagram8

Tools

Field Name Explanation

Syslog

Enable Syslog Enable or disable system log.

Server Address System log server IP address.

Server Port System log server port.

APP Log Level Set the level of APP log.

SIP Log Level Set the level of SIP log.

Network Packets Capture

Capture a packet stream from the equipment. This is normally used to troubleshoot problems.

Reboot Phone

Some configuration modifications require a reboot to become effective. Clicking the Reboot button will lead to reboot immediately.Note: Be sure to save the configuration before rebooting.

7.3.2 Network

7.3.2.1 Basic

Basic VPN Network Status IP: 172.18.3.240 Subnet mask: 255.255.0.0 Default gateway: 172.10.1.1 MAC: 0c:38:3c:1fc:2:26 MAC Timestamp: 20180403 Settings Static IP ○ DHCP ● PPPoE ○ DNS Server Configured by DHCP Primary DNS Server Secondary DNS Server Apply Service Port Settings ? Web Server Type HTTP HTTP Port 80 HTTPS Port 443 Apply HTTPS Certification File: https.pem 4501 Bytes Upload Delete

Figure16

Diagram9

Field Name Explanation
Network Status
IP The current IP address of the equipment
Subnet mask The current Subnet Mask
Default gateway The current Gateway IP address
MAC The MAC address of the equipment
MAC Timestamp Get the MAC address of time.
Settings
Select the appropriate network mode. The equipment supports three network modes:
Static IP	Network parameters must be entered manually and will not change. All parameters are provided by the ISP.
DHCP	Network parameters are provided automatically by a DHCP server.
PPPoE	Account and Password must be input manually. These are provided by your ISP.
If Static IP is chosen, the screen below will appear. Enter values provided by the ISP.
DNS Server	Select the Configured mode of the DNS Server.
Configured by
Primary DNS Server	Enter the server address of the Primary DNS.
Secondary DNS Server	Enter the server address of the Secondary DNS.
Click the APPLY button after entering the new settings. The equipment will save the new settings and apply them. If a new IP address was entered for the equipment, it must be used to login to the phone after clicking the APPLY button.
Service Port Settings
Web Server Type S	Specify Web Server Type - HTTP or HTTPS
HTTP Port	Port for web browser access. Default value is 80. Change this from the default to enhance security. Setting this port to 0 will disable HTTP access.Example: The IP address is 192.168.1.70 and the port value is 8090, The accessing address is http://192.168.1.70:8090.
HTTPS Port	Port for HTTPS access. An https authentication certification must be downloaded into the equipment before using https.Default value is 443. Change this from the default to enhance security.
Note:1) Any changes made on this page require a reboot to become active.2) It is suggested that the make the values bigger than 1024 if users change the port to HTTPS. Values less than 1024 are reserved.3) If the HTTP port is set to 0, HTTP service will be disabled.

7.3.2.2 VPN

The device supports remote connection via VPN. It supports both Layer 2 Tunneling Protocol (L2TP) and Open VPN protocol. This allows users securely connect from public network to local network remotely.

graph TD
    A["Ethernet"] --> B["Member B"]
    A --> C["Member A"]
    B --> D["Modem"]
    C --> E["Modem"]
    D --> F["ADSL"]
    E --> G["ISPI"]
    F --> H["ISP2"]
    G --> I["ISP3"]
    H --> J["Internet"]
    I --> J
    J --> K["Router"]
    K --> L["Firewall"]
    L --> M["Switch"]
    M --> N["Member D"]
    M --> O["Member C"]
    N --> P["ISPI"]
    O --> Q["ISPI"]
    style J fill:#FFD700,stroke:#333

Through the VPN implementation logic line

Member A Member B Member C Member D

Figure17

Basic VPN Virtual Private Network (VPN) Status VPN IP Address: 0.0,0.0 VPN Mode Enable VPN □ L2TP ○ OpenVPN ● Layer 2 Tunneling Protocol (L2TP) L2TP Server Address Authentication Name Authentication Password Apply OpenVPN Files OpenVPN Configuration file: client.ovpn N/A CA Root Certification: ca.crt N/A Client Certification: client.crt N/A Client Key: client.key N/A Upload Delete Upload Delete Upload Delete Upload Delete

Figure18
Diagram10

Field Name Explanation
VPN IP Address Shows the current VPN IP address.
VPN Mode
Enable VPN Enable	/Disable VPN.
L2TP Select Layer	2 Tunneling Protocol
OpenVPN	Select OpenVPN Protocol. (Only one protocol may be activated. After the selection is made, the configuration should be saved and the phone be rebooted.)
Layer 2 Tunneling Protocol (L2TP)
L2TP Server Address	Set VPN L2TP Server IP address.
Authentication Name	Set User Name access to VPN L2TP Server.
Authentication Password	Set Password access to VPN L2TP Server.
Open VPN Files
Upload or delete Open VPN Certification Files

7.3.3 Line

7.3.3.1 SIP

Configure a SIP server on this page.

SIP Basic Settings System Network Line Intercom settings Line SIP 1 Basic Settings >> Line Status Registered SIP Proxy Server Address 172.18.2.103 Phone number 1001 SIP Proxy Server Port 5060 Display name 1001 Backup Proxy Server Address Authentication Name 1001 Backup Proxy Server Port 5060 Authentication Password ********** Activate ✓ Outbound proxy address Outbound proxy port Realm Codecs Settings >> Advanced Settings >> Apply

Figure19

Codes Settings >> Disabled Codes Enabled Codes G.722 G.711U G.711A G.729AB

Figure20

Advanced Settings >> Subscribe For Voice Message Voice Message Number Voice Message Subscribe Period 3600 Second(s) Enable DND Blocking Anonymous Call Use 182 Response for Call waiting Anonymous Call Standard None Dial Without Registered Click To Talk User Agent Response Single Codec Use Feature Code Enable DND Enable Blocking Anonymous Call Ring Type Default Conference Type Local Server Conference Number Transfer Timeout 0 Second(s) Enable Long Contact Enable Use Inactive Hold Use Quote in Display Name DND Disabled Disable Blocking Anonymous Call Specific Server Type COMMON Registration Expiration 3600 Second(s) Use VPN ✓ Use STUN Convert URI ✓ DTMF Type RFC2833 ✓ DTMF SIP INFO Mode Send */# ✓ Transportation Protocol UDP ✓ Local Port 5060 SIP Version RFC3261 ✓ Caller ID Header PAI-RPID- ✓ Enable Strict Proxy Enable user=phone ✓ Enable SCA RTP Encryption ✓ Apply Enable DNS SRV Keep Alive Type SIP Optioi ✓ Keep Alive Interval 60 Second(s) Sync Clock Time Enable Session Timer Session Timeout 0 Second(s) Enable Rport ✓ Enable PRACK ✓ Auto Change Port ✓ Keep Authentication Auto TCP Enable Feature Sync Enable GRUU RTP Encryption Key

Figure21

Diagram11

SIP
Field Name Explanation
Basic Settings (Choose the SIP line to configured)
Line Status	Display the current line status at page loading. To get the up to date line status, user has to refresh the page manually. There are some status here:1) Inactive, indicates that this line is not activated yet, user can activate the line by selecting the option “activate”.2) Timeout, indicates the SIP registration status timeout. It means that there’s no response from SIP server. User may need to checkthe network or SIP server IP address and port.3) Registered, indicates the SIP account is registered to SIP server successfully, and is able to send or receive calls.4) 403 forbidden, indicates the SIP error code 403, means SIP server rejected the SIP registration because the username and password are incorrect. User will need to check the username and password, they must be matched with the username and password which were provided by SIP server.5) Other SIP error code, check SIP protocol standard, or contact support.
Username	Enter the username of the service account, assigned by IPPBX administrator, or provided by ISP provider.
Display name Enter the display name to be sent in a call request.
Authentication Name	Enter the authentication name of the service account, which is assigned by IPPBX administrator, or provided by ISP provider.
Authentication Password	Enter the authentication password of the service account, which is assigned by IPPBX administrator, or provided by ISP provider.
Activate Whether the service of the line should be activated
SIP Proxy Server Address	Enter the IP or FQDN address of the SIP proxy server
SIP Proxy Server Port	Enter the SIP proxy server port, default is 5060
Outbound proxy address	Enter the IP or FQDN address of outbound proxy server provided by the service provider
Outbound proxy port	Enter the outbound proxy port, default is 5060
Realm Enter the SIP domain if requested by the service provider
Codecs Settings
Set the priority and availability of the codecs by adding or remove them from the list.
Advanced Settings
Subscribe For Voice Message	Enable the device to subscribe a voice message waiting notification, if enabled, the device will receive notification from the server if there is voice message waiting on the server
Voice Message Number	Set the number for retrieving voice message
Voice Message Subscribe Period	Set the interval of voice message notification subscription
Enable DND Enable	Do-not-disturb, any incoming call to this line will berejected automatically
Blocking Anonymous Call	Reject any incoming call without presenting caller ID
Use 182 Response for Call waiting	Set the device to use 182 response code at call waiting response
Anonymous Call Standard	Set the standard to be used for anonymous
Dial Without Registered	Set call out by proxy without registration
Click To Talk Set Click To Talk
User Agent Set the user agent, the default is Model with Software Version.
Response Single Codec	If setting enabled, the device will use single codec in response to an incoming call request
Ring Type Set the ring tone type for the line
Conference Type	Set the type of call conference, Local=set up call conference by the device itself, maximum supports two remote parties, Server=set up call conference by dialing to a conference room on the server
Server Conference Number	Set the conference room number when conference type is set to be Server
Transfer Timeout Set the timeout of call transfer process
Enable Long Contact	Allow more parameters in contact field per RFC 3840
Use Quote in Display Name	Whether to add quote in display name
Use Feature Code	When this setting is enabled, the features in this section will not be handled by the device itself but by the server instead. In order to control the enabling of the features, the device will send feature code to the server by dialing the number specified in each feature code field.
Specific Server Type	Set the line to collaborate with specific server type
Registration Expiration	Set the SIP expiration interval
Use VPN Set the line to use VPN restrict route
Use STUN Set the line to use STUN for NAT traversal
Convert URI Convert not digit and alphabet characters to %hh hex code
DTMF Type Set the DTMF type to be used for the line
DTMF SIP INFO Mode	Set the SIP INFO mode to send ‘*’ and ‘#’ or ‘10’ and ‘11’
Transportation Protocol	Set the line to use TCP or UDP for SIP transmission
Local Port Set the Local Port
SIP Version Set the SIP version
Caller ID Header Set	the Caller ID Header
Enable Strict Proxy	Enables the use of strict routing. When the phone receives packets from the server, it will use the source IP address, not the address in via field.
Enable user=phone Sets user=phone in SIP messages.
Enable SCA Enable/Disable SCA (Shared Call Appearance)
Enable BLF List Enable/Disable BLF List
Enable DNS SRV	Set the line to use DNS SRV which will resolve the FQDN in proxy server into a service list
Keep Alive Type	Set the line to use dummy UDP or SIP OPTION packet to keep NAT pinhole opened
Keep Alive Interval Set the keep alive packet transmitting interval
Enable Session Timer	Set the line to enable call ending by session timer refreshment. The call session will be ended if there is not new session timer event update received after the timeout period
Session Timeout Set the session timer timeout period
Enable rport Set the line to add rport in SIP headers
Enable PRACK Set the line to support PRACK SIP message
Enable DNS SRV	Set the line to use DNS SRV which will resolve the FQDN in proxy server into a service list
Auto Change Port Enable/Disable Auto Change Port
Keep Authentication	Keep the authentication parameters from previous authentication
Auto TCP	Using TCP protocol to guarantee usability of transport for SIP messages above 1500 bytes
Enable Feature Sync	Feature Sycn with server
Enable GRUU	Support Globally Routable User-Agent URI (GRUU)
RTP Encryption	Enable RTP encryption such that RTP transmission will be encrypted
RTP Encryption Key	Set the pass phrase for RTP encryption

7.3.3.2 Basic setting

STUN – Simple Traversal of UDP through NAT – A STUN server allows a phone in a private network to know its public IP and port as well as the type of NAT being used. The equipment can then use this information to register itself to a SIP server so that it can make and receive calls while in a private network.

graph LR
    A["Gateway"] -->|Wants to receive data on Port 5060| B["Private Network"]
    B -->|Send request to STUN Server from Port 5060| C["NAT"]
    C -->|NAT Mapping Port 12345| D["Public Network"]
    D --> E["STUN Server"]
    C -->|What’s my ip ?| F["Gateway"]
    E --> F

Figure22

SIP Basic Settings System Network Line Intercom settings SIP Settings Local SIP Port 5060 Registration Failure Retry Interval 32 Second(s) Enable Strict UA Match Enable DHCP Option 120 Strict Branch Apply STUN Settings STUN NAT Traversal FALSE Server Address Server Port 3478 Binding Period 50 Second(s) SIP Waiting Time 800 millisecond Apply TLS Certification File: sips.pem N/A Upload Delete

Figure23

Diagram12

Basic Settings
Field Name Explanation
SIP Settings
Local SIP Port Set	the local SIP port used to send/receive SIP messages.
RegistrationFailure RetryInterval	Set the retry interval of SIP REGISTRATION when registration failed.
Enable Strict UA Match	Enable or disable Strict UA Match
Enable DHCP Option 120	DHCP Server would respond an OPTION message to the request from DHCP client. To working with the terminal device, Access device and DHCP policy server would be able to implement the zero configuration and auto provisioning. OPTION 120 is one of the OPTIONS in which the device could obtain the SIP server address from the ACK response sent back by the DHCP server.Then the SIP Agent of terminal device starts register with the SIP server address.
Strict Branch The value determined whether it's exactly matched the Branch
STUN Settings
Server Address STUN Server IP address
Server Port STUN	Server Port – Default is 3478.
Binding Period	STUN blinding period – STUN packets are sent at this interval to keep the NAT mapping active.
SIP Waiting Time	Waiting time for SIP. This will vary depending on the network.
TLS Certification File
Upload or delete the TLS certification file used for encrypted SIP transmission.
Note: the SIP STUN is used to achieve the SIP penetration of NAT, and the realization of a service, when the equipment configuration of the STUN server IP and port (usually the default is 3478), and select the Use Stun SIP server, the use of NAT equipment to achieve penetration.

7.3.4 Intercom settings

7.3.4.1 Features

Features Audio Video MCAST Action URL Time/Date System Network Line Intercom settings Enable DND Enable Intercom Mute Enable Auto Answer No Answer Auto Hangup Voice Read IP Enable Delay Start Description SIP Speaker Apply Ban Outgoing Enable Intercom Ringling Auto Answer Timeout Auto Hangup Timeout Voice Play Language Delay Start Time 1 (0~60)Second(s) (1~60)Second(s) English (1~180)Second(s)

Figure24

Diagram13

Features
Field Name Explanation
Basic Settings
Enable DND	DND might be disabled phone for all SIP lines, or line for SIP individually. But the outgoing calls will not be affected
Ban Outgoing If enabled, no outgoing calls can be made.
Enable Intercom Mute	If enabled, mutes incoming calls during an intercom call.
Enable Intercom Ringing	If enabled, plays intercom ring tone to alert to an intercom call.
Enable Auto Answer	Enable Auto Answer function
Auto Answer Timeout	Set Auto Answer Timeout
No Answer Auto Hangup	Enable automatically hang up when no answer
Auto Hangup Timeout	Configuration in a set time, automatically hang up when no answer
Voice Read IP Enable or disable voice broadcast IP address
Voice Play Language	Set language of the voice prompt
Enable Delay Start Enable or disable the start delay
Delay Start Time Set start delay time
Description	Device description displayed on IP scanning tool software. Initial

Value is "SIP Speaker".

7.3.4.2 Audio

This page configures audio parameters such as voice codec; speak volume, MIC volume and ringer volume.

Features Audio Video NCAST Action URL Time/Date Audio Settings First Codec G.722 Second Codec G.711A Third Codec G.711U Fourth Codec G.725AB Fifth Codec None Sixth Codec None DTMF Payload Type 101 (96~127) Default Ring Type Type 1 G.729AB Payload Length 20ms Tone Standard United Str G.722 Timestamps 160/20ms G.723.1 Bit Rate 6.3kb/s Speakerphone Volume 5 (1~8) MIC Input Volume 5 (1~8) Broadcast Output Volume 5 (1~9) Signal Tone Volume 4 (0~9) Enable VAD Apply Speaker Settings# Speaker Panel See External Speaker Power 10 W Apply AEC Settings# Speaker Limit in Double Talk 12 Local Noise Inhibition in No Talking 18 Speaker Inhibition in Double Talk 6 Mic Inhibition in Double Talk 6 Apply Reset

Figure25

Diagram14

Audio Setting
Field Name Explanation
First Codec	The first codec choice: G.711A/u, G.722, G.723.1, G.729AB, G.726-32
Second Codec	The second codec choice: G.711A/u, G.722, G.723.1, G.729AB, G.726-32, None
Third Codec	The third codec choice: G.711A/u, G.722, G.723.1, G.729AB, G.726-32, None
Fourth Codec	The forth codec choice: G.711A/u, G.722, G.723.1, G.729AB, G.726-32, None
DTMF Payload Type	The RTP Payload type that indicates DTMF. Default is 101
Default Ring Type	Ring Sound – There are 9 standard types and 3 User types.
G.729AB G.729AB	Payload Length – Adjusts from 10 – 60 ms.
Payload Length
Tone Standard Configure tone standard area.
G.722 Timestamps	Choices are 160/20ms or 320/20ms.
G.723.1 Bit Rate	Choices are 5.3kb/s or 6.3kb/s.
Speakerphone Volume	Set the speaker calls the volume level.
MIC Input Volume	Set the MIC calls the volume level.
Broadcast Output Volume	Set the broadcast the output volume level.
Signal Tone Volume	Set the audio signal the output volume level.
Enable VAD	Enable or disable Voice Activity Detection (VAD). If VAD is enabled, G729 Payload length cannot be set greater than 20 ms.
Speaker Settings
These settings are only for the devices which support multiple output power. Be aware of that, the selected output power must be less than the real output power of the external speaker, otherwise the external speaker might be damaged.
Speaker	The embedded speaker can be set to use static output power mode, and the external speak can be set as 10W, 20W, 30W output power. NOTE: this device support embedded speaker
External Speaker Power	Set the external speaker power, it must be lower than the real power of the external speaker, otherwise the external speaker might be damaged.
AEC Settings
Speaker Limit in Double Talk	Limit maximum volume of the speaker while it's in the two-way conversation, the bigger the value, the loader the volume allowed.
Local Noise Inhibition in No Talking	While there's no talking on the conversation, the background noise will be inhibited, this value determined how much it's inhibited. The higher the value, the more background noise will be inhibited. It's not recommended to set it too big, because there will be more background noise while talking in the conversation.
Speaker Inhibition in Double Talk	Set the speaker inhibition while it's in the two-way conversation, the higher of the inhibition value, the smaller of the volume.
Mic Inhibition in Double Talk	Set the MIC inhibition while it's in the two-way conversation, the higher of the inhibition value, the smaller of the volume.

7.3.4.3 Video

Features Audio Video MCAST Action URL Time/Date Camera Status Active Max Access Num 5 Max M Num 2 Use 0 Max S Num 5 Use 0 Authentication Setting Mac 00:12:17:21:f6:99 Auth Code 6fc3938128e9b4f500053 Apply Connection mode setting Connect Mode: Local Apply

Figure26

Video Capture>> IRCUT Mode Automatic Day/Night Mode Automatic White Balance Automatic Horizon Flip Enable Anti Flicker Disable Vertical Flip Enable IR Swap Disable DNC Threshold 29 (10~50) Backlight Compensation Disable AutoFill Sensitivity 5 (1~10) wide dynamic Enable Wide dynamic upper limit 30 (0~100) Fill Light Enable Default Apply

Figure27

Video Encode>> Main Stream Sub Stream Encode Format H264 H264 Resolution 720P CIF Frame Rate 20 20 VBR VBR Quality General General Bitrate 1700 318 I Frame Interval 2 [1~12]S 2 (1~12)S Activate ✓ ✓ Default Apply Encode Static config Base line Apply

Figure28

Advanced Settings >>

Video Direction

H.264 Payload Type

Sendonly

117 (96\~127)

Default

Apply

RTSP Information

Main StreamUrl:

rtsp://172.18.3.240/user=admin&password=tlJwpbo6&channel=1&stream=0.sdp?real_stream

Sub StreamUrl:

rtsp://172.18.3.240/user=admin&password=tlJwpbo6&channel=1&stream=1.sdp?real_stream

Preview

Preview

Figure29
Diagram15

Video
Field Name Explanation
Camera Status: Display the relevant information of the camera, including maximum access, maximum stream, maximum sub stream, and the status.
Authentication Setting
MAC MAC address
Auth Code Enter	authentication code to activate use
Connection mode setting
Local Connect the original camera
External Connect to another manufacturers camera
Video Capture
IRCUT Mode	Auto: IRCUT switches according to the actual ambient light level of the cameraSynchronization: The switching of the IRCUT is determined by the actual brightness of the IR lamp.
Day/Night Mode	Automatic: automatically switches according to the DNC Threshold and the brightness of the actual environment where the camera is locatedDay Mode: The camera's video screen is always colored, if there is IR-cut will be synchronized to switch.Night Mode: the camera's video screen is always black and white, if there is IR-cut will be synchronized switch.
White Balance	Automatic: Automatically adjusts according to the actual environment in which the camera is located.Outdoor: installed in the outdoor preferred.Indoor: installed in the room preferred.
Horizon Flip The	video is flipped horizontally
Anti Flicker	Enable the option. In a fluorescent environment can eliminate the video horizontal scroll
Vertical Flip The	video is flipped horizontally
IR Swap IR-cut filter switch
DNC Threshold	In the Day / Night mode Auto option, the color switching black and white threshold is set
Backlight Compensation	In front of a very strong background light can see people or objects clearly
AutoFill Sensitivity	In the environment changes in light and shade, the higher the sensitivity the faster the video changes
wide Dynamic Set wide dynamic
Wide Dynamic Upper Limit	Change the brightness of the background image, the higher the brighter.
Fill Light Enable	or disable Fill Light
Video Encode
Encode Format Only H.264 encoding format is supported
Resolution	Main stream: support 720PSub-stream: you can select CIF (352 * 288), D1 (720 * 576)
Frame Rate	The larger the value is, the more coherent the video would be got; not recommend adjusted.
Bitrate Control	CBR: If the code rate (bandwidth) is insufficient, it is preferred.VBR: Image quality is preferred, not recommended.
Quality	Video quality adjustment, the better the quality needs to transfer faster
Bit rate It is proportional to video file size, not recommend adjusted.
I Frame Interval	The greater the value is, the worse the video quality would be, otherwise the better video quality would be; not recommend adjusted.
Activate When you selected it, the stream is enabled, otherwise disabled
Encode Static config
Select the video codec type, it's recommended to use “Base Line” to stay the same as the video output or stream receiver.
Advanced Settings
Video Direction	Select the transport type of the video stream
H.264 Payload Type	Set the payload type of H.264
RTSP Information
Main StreamUrl	Access the main address of RTSP
Sub Stream URL	Access the child address of RTSP

7.3.4.4 MCAST

Features Audio Video MCAST Action URL Time/Date System Network Line Intercom settings MCAST Settings Priority 1 Enable Page Priority Index/Priority Name Host:port 1 2 3 4 5 6 7 8 9 10 Apply

Figure30

It is easy and convenient to use multicast function to send notice to each member of the multicast via setting the multicast key on the device and sending multicast RTP stream to pre-configured multicast address. By configuring monitoring multicast address on the device, monitor and play the RTP stream which sent by the multicast address.

- MCAST Settings

Equipment can be set up to monitor up to 10 different multicast address, used to receive the multicast RTP stream sent by the multicast address.

Here are the ways to change equipment receiving multicast RTP stream processing mode in the Web interface: set the ordinary priority and enable page priority.

Priority:

In the drop-down box to choose priority of ordinary calls the priority, if the priority of the incoming flows of multicast RTP, lower precedence than the current common calls, device will automatically ignore the group RTP stream. If the priority of the incoming flow of multicast RTP is higher than the current common calls priority, device will automatically receive the group RTP stream, and keep the

current common calls in state. You can also choose to disable in the receiving threshold drop-down box, the device will automatically ignore all local network multicast RTP stream.

■ The options are as follows:

✿ 1-10: To definite the priority of the common calls, 1 is the top level while 10 is the lowest

✨ Disable: ignore all incoming multicast RTP stream

✨ Enable the page priority:

Page priority determines the device how to deal with the new receiving multicast RTP stream when it is in multicast session currently. When Page priority switch is enabled, the device will automatically ignore the low priority multicast RTP stream but receive top-level priority multicast RTP stream and keep the current multicast session in state; If it is not enabled, the device will automatically ignore all receiving multicast RTP stream.

■ Web Settings:

MCAST Settings Priority 1 Enable Page Priority ✓ Index/Priority Name Host:port 1 ss 239.1.1.1:1366 2 ee 239.1.1.1:1367

Figure31

The multicast SS priority is higher than that of EE, which is the highest priority.

Note: when pressing the multicast key for multicast session, both multicast sender and receiver will beep.

- Listener configuration

MCAST Settings Priority 3 Enable Page Priority Index/Priority Name Host:port 1 group 1 224.0.0.2:2366 2 group 2 224.0.0.2:1366 3 group 3 224.0.0.6:3366 4 5 6 7 8 9 10

Figure32

■ Blue part (name)

"Group 1", "Group 2" and "Group 3" are your setting monitoring multicast name. The group name will be displayed on the screen when you answer the multicast. If you have not set, the screen will display the IP: port directly.

■ Purple part (host: port)

It is a set of addresses and ports to listen, separated by a colon.

■ Pink part (index / priority)

Multicast is a sign of listening, but also the monitoring multicast priority. The smaller number refers to higher priority.

■ Red part (priority)

It is the general call, non multicast call priority. The smaller number refers to high priority. The followings will explain how to use this option:

The purpose of setting monitoring multicast "Group 1" or "Group 2" or "Group 3" launched a multicast call.
All equipment has one or more common non multicast communication.
When you set the Priority for the disable, multicast any level will not answer, multicast call is rejected.
when you set the Priority to a value, only higher than the priority of multicast can come in, if you set the Priority is 3, group 2 and group 3 for priority level equal to 3 and less than 3 were rejected, 1 priority is 2 higher than ordinary call priority device can answer the multicast message at the same time, keep the hold the other call.

■ Green part (Enable Page priority)

Set whether to open more priority is the priority of multicast, multicast is pink part number. Explain how to use:

The purpose of setting monitoring multicast "group 1" or "3" set up listening "group of 1" or "3" multicast address multicast call.
All equipment has been a path or multi-path multicast phone, such as listening to "multicast information group 2".
If multicast is a new "group of 1", because "the priority group 1" is 2, higher than the current call "priority group 2" 3, so multicast call will can come in.
If multicast is a new "group of 3", because "the priority group 3" is 4, lower than the current call "priority group 2" 3, "1" will listen to the equipment and maintain the "group of 2".

- Multicast service

Send: when configured ok, our key press shell on the corresponding equipment, equipment directly into the Talking interface, the premise is to ensure no current multicast call and 3-way of the case, the multicast can be established.

Lmonitor: IP port and priority configuration monitoring device, when the

call is initiated and incoming multicast, directly into the Talking interface equipment.

7.3.4.5 Action URL

System Network Line Intercom settings Features Audio Video MCAST Action URL Time/Date Action URL Event Settings Active URL Limit IP Setup Completed Registration Succeeded Registration Disabled Registration Failed Off Hooked On Hooked Incoming Call Outgoing calls Call Established Call Terminated DND Enabled DND Disabled Mute Unmute Missed calls IP Changed Idle To Busy Busy To Idle Input1 Output1 Tamper Apply

Figure33

Diagram16

Action URL Settings

URL for various actions is performed by the phone. These actions are recorded and sent as xml files to the server. Sample format is http://InternalServer /FileName.xml

7.3.4.6 Time/Date

Features Audio Video MCAST Action URL Time/Date Network Time Server Settings Time Synchronized via SNTP ✓ Time Synchronized via DHCP □ Primary Time Server time.nlst.gov Secondary Time Server pool.ntp.org Time zone (UTC+B) China,Singapore,Australia Resync Period 60 (1~5000)Second(s) Date Format Date Format: ↓ JAN MON Apply Daylight Saving Time Settings Location China(Beijing) DST Set Type Disabled Apply Manual Time Settings ? 2018-05-31 13 34 Apply System Time: 2018-05-31 13:34

Figure34
Diagram17

Time/Date
Field Name Explanation
Network Time Server Settings
Time Synchronized via SNTP	Enable time-sync through SNTP protocol
Time Synchronized via DHCP	Enable time-sync through DHCP protocol
Primary Time Server	Set primary time server address
Secondary Time Server	Set secondary time server address, when primary server is not reachable, the device will try to connect to secondary time server to get time synchronization.
Time zone	Select the time zone
Resync Period	Time of re-synchronization with time server
Date Format
Date Format	Select the time/date display format
Daylight Saving Time Settings
Location	Select the user's time zone specific area
DST Set Type	Select automatic DST according to the preset rules of DST, or the manually input rules
Manual Time Settings
The time set by hand, need to disable SNTP service first.

8.1 Technical parameters

Diagram18

Communication protocol		SIP 2.0(RFC-3261)
Main chipset		Broadcom
Button	Reset	One
	Volume	Two
Speech flow	Protocols	RTP/SRTP
	Decoding	G.729、G.723、G.711、G.722、G.726
	Audio amplifier	Max 30W
	Volume control	Adjustable
LED Indicating lamp		One
Port	Power	One
	WAN	10/100BASE-TX s Auto-MDIX, RJ-45
	LAN	10/100BASE-TX s Auto-MDIX, RJ-45
power supply mode		12V 2A DC~24V 2A DC or POE
Cables		CAT5 or better
working temperature		-10°C to 50°C
working humidity		20% - 80%
storage temperature		-10°C to 50°C
overall dimension		165x240x185mm (W x H x L)
Package dimensions		260x315x305mm (W x H x L)
Package weight		3.1KG

8.2 Basic functions

• 2 SIP lines
  ● POE enabled (Power over Ethernet)
  ● Support for dc power supply
• Support VLAN
  ● Support camera linkage
  ● Wall-mount installation
• Multicast

8.3 Schematic diagram

On the back of the interface diagram

POWER LAN WAN NET VOLUME RST AUDIO + - 12V-24V 2A

Black studio microphone unit with visible sound waves and buttons (no text or symbols)

Figure35

8.4 The radio terminal configuration notice

How to avoid an incoherency sound when the broadcast playing?

When the terminal use as broadcast, the speaker is loud, if not set mute for microphone, the AEC (echo cancellation) of equipment will be activated, which leads the sound incoherence. In order to avoid such circumstance, when the equipment turn to use as radio should be set as intercom mode, and activate the intercom mute, so as to ensure the broadcast quality.

Features Audio Video MCAST Action URL Time/Date System Network Line Intercom settings Audio Settings First Codec G.722 Second Codec G.711A Third Codec G.711U Fourth Codec G.729AB Fifth Codec None Sixth Codec None DTMF Payload Type 101 (96~127) Default Ring Type Type 1 G.729AB Payload Length 20ms Tone Standard United Sts G.722 Timestamps 160/20ms 0.723.1 Bit Rate 6.3kb/s Speakerphone Volume 5 (1~9) MIC Input Volume 5 (1~9) Broadcast Output Volume 5 (1~9) Signal Tone Volume 4 (0~9) Enable VAD Apply Speaker Settings① Speaker Panel Spe External Speaker Power 10 W Apply AEC Settings② Speaker Limit in Double Talk 12 Local Noise Inhibition in No Talking 18 Speaker Inhibition in Double Talk 8 Nic Inhibition in Double Talk 6

Figure36

◇ How to improve broadcasting tone quality?

In order to obtain better broadcast quality, recommend the use of the HD (G.722) mode for broadcast.

Voice bandwidth will be by the narrow width (G.711) of 4 KHz, is extended to broadband (G.722)7 KHz, when combined with the active speaker, the effect will be better.

Features Audio Video MCAST Action URL Time/Date System Network Line Intercom settings Audio Settings First Codec G.722 Second Codec G.711A Third Codec G.711U Fourth Codec G.729AB Fifth Codec None Sixth Codec None DTMF Payload Type 101 (98~127) Default Ring Type Type 1 G.729AB Payload Length 20ms Tone Standard United Sb G.722 Timestamps 160/20ms G.723.1 Bit Rate 6.3kb/s Speakerphone Volume 5 (1~9) MIC Input Volume 5 (1~9) Broadcast Output Volume 5 (1~9) Signal Tone Volume 4 (0~9) Enable VAD Apply

Figure37