AXIS I8016-LVE Network Video Intercom

To find Axis devices on the network and assign them IP addresses in Windows®, use AXIS IP Utility or AXIS Device Manager. Both applications are free and can be downloaded from axis.com/support.

For more information about how to find and assign IP addresses, go to How to assign an IP address and access your device.

You can use the device with the following browsers:

✓: Recommended
*: Supported with limitations

1. Open a browser and type the IP address or host name of the Axis device.

   If you don’t know the IP address, use AXIS IP Utility or AXIS Device Manager to find the device on the network.

2. Type the username and password. If you access the device for the first time, you must create an administrator account. See Create an administrator account.

For descriptions of all features and settings in the web interface of devices with AXIS OS, see AXIS OS web interface help.

1. To make sure that the device has its original AXIS OS, or to take full control of the device after a security attack:

2. Reset to factory default settings. See Reset to factory default settings.

   After the reset, secure boot guarantees the state of the device.

3. Configure and install the device.

The first time you log in to your device, you must create an administrator account.

1. Enter a username.

2. Enter a password. See Secure passwords.

3. Re-enter the password.

4. Accept the license agreement.

5. Click Add account.

The device password is the primary protection for your data and services. Axis devices do not impose a password policy as they may be used in various types of installations.

To protect your data we strongly recommend that you:

• Use a password with at least 8 characters, preferably created by a password generator.

• Don’t expose the password.

• Change the password at a recurring interval, at least once a year.

This section covers all the important configurations that an installer needs to do to get the product up and running after the hardware installation has been completed.

You can run a speaker test to verify from a remote location that a speaker is working as intended. The speaker performs the test by playing a series of test tones that are registered by the built-in microphone. Every time you run the test, the registered values are compared with the values that were registered during the calibration.

1. Go to the device interface > Audio > Speaker test.

2. To calibrate the audio device, click Calibrate.

3. Note

   Once the Axis product is calibrated, the speaker test can be run at any time.

4. To run the speaker test, click Run the test.

VoIP (Voice over IP) is a group of technologies that enables voice and multimedia communication over IP networks. For more information, see Voice over IP (VoIP).

In this device VoIP is enabled through the SIP protocol. For more information about SIP, see Session Initiation Protocol (SIP)

There are two types of setups for SIP. Direct or peer-to-peer (P2P) is one of them. Use peer-to-peer when the communication is between a few user agents within the same IP network and there is no need for extra features that a PBX-server could provide. For information on how to set it up, see Peer-to-peer SIP (P2PSIP).

1. Go to Communication > SIP > Settings and select Enable SIP.

2. To allow the device to receive incoming calls, select Allow incoming calls.

3. NOTICE

   When you allow incoming calls, the device accepts calls from any device connected to the network. If the device is accessible from a public network or the internet, we recommend you not to allow incoming calls.

4. Click Call handling.

5. In Calling timeout, set the number of seconds that a call will last before it ends if there is no answer.

6. If you have allowed incoming calls, set the number of seconds before timeout for incoming calls in Incoming call timeout.

7. Click Ports.

8. Enter the SIP port number and TLS port number.

9. Note

   • SIP port – for SIP sessions. Signalling traffic through this port is non-encrypted. The default port number is 5060.
   • TLS port – for SIPS and TLS secured SIP sessions. Signalling traffic through this port is encrypted with Transport Layer Security (TLS). The default port number is 5061.
   • RTP start port – the port used for the first RTP media stream in a SIP call. The default start port is 4000. Some firewalls can block RTP traffic on certain port numbers. The port number must be between 1024 and 65535.

10. Click NAT traversal.

11. Select the protocols you want to enable for NAT traversal.

12. Note

    Use NAT traversal when the device is connected to the network from behind a NAT router or a firewall. For more information see NAT traversal.

13. Click Save.

VoIP (Voice over IP) is a group of technologies that enables voice and multimedia communication over IP networks. For more information, see Voice over IP (VoIP).

In this device, VoIP is enabled through the SIP protocol. For more information about SIP, see Session Initiation Protocol (SIP)

There are two types of setups for SIP. A PBX server is one of them. Use a PBX server when the communication should be between an infinite number of user agents within and outside the IP network. Additional features could be added to the setup depending on the PBX provider. For more information, see Private Branch Exchange (PBX).

1. Request the following information from your PBX provider:

2. • User ID

   • Domain

   • Password

   • Authentication ID

   • Caller ID

   • Registrar

   • RTP start port

3. Go to Communication > SIP > Accounts and click + Add account.

4. Enter a Name for the account.

5. Select Registered.

6. Select a transport mode.

7. Add the account information from the PBX provider.

8. Click Save.

9. Set up the SIP settings in the same way as for peer-to-peer, see Set up direct SIP (P2P). Use the RTP start port from the PBX provider.

This example explains how to create a new contact in the contact list. Before you start, enable SIP in Communication > SIP.

To create a new contact:

1. Go to Communication > Contact list.

2. Click + Add contact.

3. Enter the first and last name of the contact.

4. Enter the contact’s SIP address.

5. Note

   For information about SIP addresses, see Session Initiation Protocol (SIP).

6. Select the SIP account to call from.

7. Note

   Availability options are defined in System > Events > Schedules.

8. Choose the contact’s Availability. If there’s a call when the contact isn’t available, the call gets canceled unless a there’s a fallback contact.

9. Note

   A fallback is a contact, to whom the call gets forwarded if the original contact doesn’t reply or isn’t available.

10. In Fallback, select None.

11. Click Save.

By default, the call button is configured to make VMS (video management software) calls. If you want to keep this configuration, you just need to add the Axis intercom to the VMS.

This example explains how to set up the system to call a contact in the contact list when a visitor presses the call button.

1. Go to Communication > Calls > Call button.

2. Under Recipients, remove VMS.

3. Under Recipients, select an existing or create a new contact.

To disable the call button, turn off Enable call button.

When a visitor makes a call from the intercom, the person who answers can use the Dual-Tone Multi-Frequency signaling (DTMF) of his SIP device to unlock the door. The door controller unlocks and locks the door.

• This example explains how to:

• define the DTMF signal in the intercom

• set up the intercom to:

• • request the door controller to unlock the door, or

  • unlock the door using the internal relay.

• You make all settings in the intercom’s webpage.

• Before you start

• Allow SIP calls from the device and create a SIP account. See Set up direct SIP (P2P) and Set up SIP through a server (PBX).

1. Define the DTMF signal in the intercom

2. Go to Communication > SIP > DTMF.

3. Click + Add sequence.

4. In Sequence, enter 1.

5. In Description, enter Unlock door.

6. In Accounts, select the SIP account.

7. Click Save.

8. Set up the intercom to unlock the door using the internal relay

9. Go to System > Events > Rules and add a rule.

10. In the Name field, enter DTMF unlock door.

11. From the list of conditions, under Call, select DTMF and Unlock door.

12. From the list of actions, under I/O, select Toggle I/O once.

13. From the list of ports, select Relay 1.

14. Change Duration to 00:00:07, which means that the door is open for 7 seconds.

15. Click Save.

Your camera uses visible light to deliver color images during the day. But as the visible light diminishes, color images become less bright and clear. If you switch to night mode when this happens, the camera uses both visible and near-infrared light to deliver bright and detailed black-and-white images instead. You can set the camera to switch to night mode automatically.

1. Go to Video > Image > Day-night mode, and make sure that the IR-cut filter is set to Auto.

To read about all the features and settings available in the web interface of devices with AXIS OS, go to AXIS OS web interface help.

Voice over IP (VoIP) is a group of technologies that enables voice communication and multimedia sessions over IP networks, such as the internet. In traditional phone calls, analog signals are sent through circuit transmissions over the Public Switched Telephone Network (PSTN). In a VoIP call, analog signals are turned into digital signals to make it possible to send them in data packets across local IP networks or the internet.

In the Axis product, VoIP is enabled through the Session Initiation Protocol (SIP) and Dual-Tone Multi-Frequency (DTMF) signaling.

Example

When you press the call button on an Axis intercom, a call is initiated to one or more predefined recipients. When a recipient replies, a call is established. The voice and video is transferred through VoIP technologies.

The Session Initiation Protocol (SIP) is used to set up, maintain and terminate VoIP calls. You can make calls between two or more parties, called SIP user agents. To make a SIP call you can use, for example, SIP phones, softphones or SIP-enabled Axis devices.

The actual audio or video is exchanged between the SIP user agents with a transport protocol, for example RTP (Real-Time Transport Protocol).

You can make calls on local networks using a peer-to-peer setup, or across networks using a PBX.

The most basic type of SIP communication takes place directly between two or more SIP user agents. This is called peer-to-peer SIP (P2PSIP). If it takes place on a local network, all that’s needed are the SIP addresses of the user agents. A typical SIP address in this case would be sip:<local-ip>.

Example

1. User agent A - intercom. SIP address: sip:192.168.1.101
2. User agent B - SIP-enabled phone. SIP address: sip:192.168.1.100

You can set up the Axis intercom to call for example a SIP-enabled phone on the same network using a peer-to-peer SIP setup.

When you make SIP calls outside your local IP network, a Private Branch Exchange (PBX) can act as a central hub. The main component of a PBX is a SIP server, which is also referred to as a SIP proxy or a registrar. A PBX works like a traditional switchboard, showing the client's current status and allowing for example call transfers, voicemail, and redirections.

The PBX SIP server can be set up as a local entity or offsite. It can be hosted on an intranet or by a third party provider. When you make SIP calls between networks, calls are routed through a set of PBXs, that query the location of the SIP address to be reached.

Each SIP user agent registers with the PBX, and can then reach the others by dialing the correct extension. A typical SIP address in this case would be sip:<user>@<domain> or sip:<user>@<registrar-ip>. The SIP address is independent of its IP address and the PBX makes the device accessible as long as it is registered to the PBX.

Example

1. sip:mydoor@company.com
2. sip:myspeaker@company.com
3. PBX sip.company.com
4. sip:office@company.com

When you press the call button on an Axis intercom, the call is forwarded through one or more PBXs to a SIP address either on the local IP network or over the internet.

You can create rules to make your device perform an action when certain events occur. A rule consists of conditions and actions. The conditions can be used to trigger the actions. For example, the device can start a recording or send an email when it detects motion, or show an overlay text while the device is recording.

To learn more, see Get started with rules for events.

Use NAT (Network Address Translation) traversal when the Axis device is located on an private network (LAN) and you want to access it from outside of that network.

Each NAT traversal protocol can be used separately or in different combinations depending on the network environment.

• ICE The ICE Interactive Connectivity Establishment) protocol increases the chances of finding the most efficient path to successful communication between peer devices. If you also enable STUN and TURN, you improve the ICE protocol’s chances.

• STUN - STUN (Session Traversal Utilities for NAT) is a client-server network protocol that lets the Axis device determine if it is located behind a NAT or firewall, and if so obtain the mapped public IP address and port number allocated for connections to remote hosts. Enter the STUN server address, for example, an IP address.

• TURN - TURN (Traversal Using Relays around NAT) is a protocol that lets a device behind a NAT router or firewall receive incoming data from other hosts over TCP or UDP. Enter TURN server address and the login information.

With analytics and apps you can get more out of your Axis device. AXIS Camera Application Platform (ACAP) is an open platform that makes it possible for third parties to develop analytics and other apps for Axis devices. Apps can be preinstalled on the device, available for download for free, or for a license fee.

To find the user manuals for Axis analytics and apps, go to help.axis.com.

With this application you can make calls between SIP-enabled Axis devices and linked Microsoft® Teams accounts. To find out more, see the user manual for AXIS Client for Unified Communication Systems.

To reset the product to the factory default settings:

1. Disconnect power from the product.

2. Press and hold the control button while reconnecting power. See Product overview.

3. Keep the control button pressed for 15–30 seconds until the status LED indicator flashes amber.

4. Release the control button. The process is complete when the status LED indicator turns green. If no DHCP server is available on the network, the device IP address will default to one of the following:

5. • Devices with AXIS OS 12.0 and later: Obtained from the link-local address subnet (169.254.0.0/16)

   • Devices with AXIS OS 11.11 and earlier: 192.168.0.90/24

6. Use the installation and management software tools to assign an IP address, set the password, and access the device.

   The installation and management software tools are available from the support pages on axis.com/support.

You can also reset parameters to factory default through the device’s web interface. Go to Maintenance > Factory default and click Default.

AXIS OS determines the functionality of our devices. When you troubleshoot a problem, we recommend that you to start by checking the current AXIS OS version. The latest version might contain a correction that fixes your particular problem.

To check the current AXIS OS version:

1. Go to the device’s web interface > Status.

2. Under Device info, see the AXIS OS version.

1. Download the AXIS OS file to your computer, available free of charge at axis.com/support/device-software.

2. Log in to the device as an administrator.

3. Go to Maintenance > AXIS OS upgrade and click Upgrade.

4. When the upgrade has finished, the product restarts automatically.

Problems upgrading AXIS OS

Problems setting the IP address

Problems accessing the device

Problems with MQTT

Problems with operating the device

If you can’t find what you’re looking for here, try the troubleshooting section at axis.com/support.

When you set up your system, it’s important to consider how different settings and situations affect performance. Some factors affect bandwidth (bitrate), others affect frame rate, and some affect both.

The most important factors to consider:

• High image resolution or lower compression levels result in images containing more data which in turn affects the bandwidth.

• Access by large numbers of Motion JPEG clients or unicast H.264/H.265/AV1 clients affects the bandwidth.

• Simultaneous viewing of different streams (resolution, compression) by different clients affects both frame rate and bandwidth.

  Use identical streams wherever possible to maintain a high frame rate. Stream profiles can be used to ensure that streams are identical.

• Accessing video streams with different codecs simultaneously affects both frame rate and bandwidth. For optimal performance, use streams with the same codec.

• Heavy usage of event settings affects the product’s CPU load which in turn affects the frame rate.

• Using HTTPS may reduce frame rate, in particular if streaming Motion JPEG.

• Heavy network utilization due to poor infrastructure affects the bandwidth.

• Viewing on poorly performing client computers lowers perceived performance and affects frame rate.

• Running multiple AXIS Camera Application Platform (ACAP) applications simultaneously may affect the frame rate and the general performance.

If you need more help, go to axis.com/support.

1. To check relay state, go to System > Accessories and find the relay port.

2. Set Normal state to:

3. • for a fail-secure lock.

   • for a fail-safe lock.

1. To check relay state, go to System > Accessories and find the relay port.

2. Set Normal state to:

3. • for a fail-secure lock.

   • for a fail-safe lock.

1. To check relay state, go to System > Accessories and find the relay port.

2. Set Normal state to:

3. • for a fail-secure lock.

   • for a fail-safe lock.

1. To check relay state, go to System > Accessories and find the relay port.

2. Set Normal state to:

3. • for a fail-secure lock.

   • for a fail-safe lock.

1. To check relay state, go to System > Accessories and find the relay port.

2. Set Normal state to:

3. • for a fail-secure lock.

   • for a fail-safe lock.

When you connect the product to power, the front panel indicators light up for a few seconds.

This device supports microSD/microSDHC/microSDXC cards.

For SD card recommendations, see axis.com.

microSD, microSDHC, and microSDXC Logos are trademarks of SD-3C LLC. microSD, microSDHC, microSDXC are trademarks or registered trademarks of SD-3C, LLC in the United States, other countries or both.

• The control button is used for:

• Resetting the product to factory default settings. See Reset to factory default settings.

RJ45 Ethernet connector with Power over Ethernet (PoE).

4-pin terminal block for audio input and output.

Use the I/O connector with external devices in combination with, for example, motion detection, event triggering, and alarm notifications. In addition to the 0 VDC reference point and power (12 V DC output), the I/O connector provides the interface to:

Digital input

For connecting devices that can toggle between an open and closed circuit, for example PIR sensors, door/window contacts, and glass break detectors.

Digital output

For connecting external devices such as relays and LEDs. Connected devices can be activated by the VAPIX® Application Programming Interface, through an event or from the device’s web interface.