Additional Settings; Change The Root Password; Set Up Direct Sip (P2P) - Axis A82 Series User Manual

AXIS A82 Network Video Door Station Series

Additional settings

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

Change the root password

1. Log in to the product's webpage and go to System > Users.
2. For the root user, click Edit.
3. Enter a new password and save.

Set up direct SIP (P2P)

VoIP (Voice over IP) is a group of technologies that enables voice and multimedia communication over IP networks. For more,
see Voice over IP (VoIP) on page 13.
In this product VoIP is enabled through the SIP protocol. For more information about SIP, see Session Initiation Protocol (SIP) on
page 14
There are two types of setups for SIP. Peer-to-peer 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) on page 14.
1. Go to VoIP > SIP settings and select Enable SIP.
2. In Call timeout, set the number of seconds that a call will last before it ends if there is no answer.
3. To allow the device to receive incoming calls, select Allow incoming calls.
NOTICE NO NO TICE TICE
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 to disable incoming calls.
4. Set the number of seconds before timeout for incoming calls.
5. Click Save.
6. In Port settings, enter the SIP port number and SIP TLS port number.
Note
• SIP port – for SIP sessions. Signalling traffic through this port is non-encrypted. The default port number is 5060.
• SIP 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 – Enter the port used for the first RTP media stream in a SIP call. The default start port for media transport is
4000. Some firewalls might block RTP traffic on certain port numbers. A port number must be between 1024 and 65535.
7. Click Save.
8. In NAT traversal, select the protocols you want to enable for NAT traversal.
Note
Use NAT traversal when the device is connected to the network from behind a NAT router or a firewall. For more information
see .
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