AXIS C8210 Network Audio Amplifier
The device interface
•
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
•
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.
Additional
•
UDP-to-TCP switching
: Select to allow calls to switch transport protocols from UDP (User Datagram Protocol) to TCP
(Transmission Control Protocol) temporarily. The reason for switching is to avoid fragmentation, and the switch can
take place if a request is within 200 bytes of the maximum transmission unit (MTU) or larger than 1300 bytes.
•
Allow via rewrite
: Select to send the local IP address instead of the router's public IP address.
•
Allow contact rewrite
: Select to send the local IP address instead of the router's public IP address.
•
Register with server every
: Set how often you want the device to register with the SIP server for the existing
SIP accounts.
•
DTMF payload type
: Changes the default payload type for DTMF.
SIP accounts
All current SIP accounts are listed under
SIP accounts
. For registered accounts, the colored circle lets you know the status.
The account is successfully registered with the SIP server.
There is a problem with the account. Possible reasons can be authorization failure, that the account credentials are wrong,
or that the SIP server can’t find the account.
The
peer to peer (default)
account is an automatically created account. You can delete it if you create at least one other account
and set that account as default. The default account is always used when a VAPIX
®
Application Programming Interface (API) call
is made without specifying which SIP account to call from.
Account
: Click to create a new SIP account.
•
Active
: Select to be able to use the account.
•
Make default
: Select to make this the default account. There must be a default account, and there can only
be one default account.
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