Configure Your Device; Set Up Direct Sip (P2P); Set Up Sip Through A Server (Pbx) - Axis I8016-LVE User Manual

AXIS I8016-LVE Network Video Intercom

Configure your device

Configure your device
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.

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 11 .
In this product VoIP is enabled through the SIP protocol. For more information about SIP, see Session Initiation Protocol (SIP) on
page 11
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 11 .
1. Go to System > SIP > SIP settings and select Enable SIP.
2. 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 not to allow incoming calls.
3. Click Call handling.
4. In Calling timeout, set the number of seconds that a call will last before it ends if there is no answer.
5. If you have allowed incoming calls, set the number of seconds before timeout for incoming calls in Incoming call timeout.
6. Click Ports.
7. Enter the SIP port number and TLS port number.
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 – 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.
8. Click NAT traversal.
9. 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 NAT traversal on page 13.
10. Click Save.

Set up SIP through a server (PBX)

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) on page 11 .
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