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SIP setup assistant: Click to set up and configure SIP step by step.
Enable SIP: Check this option to make it possible to initiate and receive SIP calls.
Allow incoming calls: Check this option to allow incoming calls from other SIP devices.
Call handling
• Calling timeout: Set the maximum duration of an attempted call if no one answers.
• Incoming call duration: Set the maximum time an incoming call can last (max 10 min).
• End calls after: Set the maximum time that a call can last (max 60 minutes). Select Infinite call
duration if you don’t want to limit the length of a call.
Ports
A port number must be between 1024 and 65535.
• SIP port: The network port used for SIP communication. The signaling traffic through this port is non-
encrypted. The default port number is 5060. Enter a different port number if required.
• TLS port: The network port used for encrypted SIP communication. The signaling traffic through this
port is encrypted with Transport Layer Security (TLS). The default port number is 5061. Enter a
different port number if required.
• RTP start port: The network port used for the first RTP media stream in a SIP call. The default start
port number is 4000. Some firewalls block RTP traffic on certain port numbers.
NAT traversal
Use NAT (Network Address Translation) traversal when the device is located on an private network (LAN) and
you want to make it available from outside of that network.
Note
For NAT traversal to work, the router must support it. The router must also support UPnP®.
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
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 the TURN server address
and the login information.
Audio and video
• Audio codec priority: Select at least one audio codec with the desired audio quality for SIP calls.
Drag-and-drop to change the priority.
Note
The selected codecs must match the call recipient codec, since the recipient codec is decisive when a call
is made.
• Audio direction: Select allowed audio directions.
• H.264 packetization mode: Select which packetization mode to use.
– Auto: (Recommended) The device decides which packetization mode to use.
– None: No packetization mode is set. This mode is often interpreted as mode 0.
– 0: Non-interleaved mode.
– 1: Single NAL unit mode.
• Video direction: Select allowed video directions.
Additional
AXIS A8207-VE Mk II Network Video Door Station
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