Axis C1004-E User Manual page 27

AXIS C1004-E Network Cabinet Speaker
The device interface
• 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.
• Name: Enter a descriptive name. This can, for example, be a first and last name, a role, or a location. The name is
not unique.
• User ID: Enter the unique extension or phone number assigned to the device.
• Peer-to-peer: Use for direct calls to another SIP device on the local network.
• Registered: Use for calls to SIP devices outside the local network, through a SIP server.
• Domain: If available, enter the public domain name. It will be shown as part of the SIP address when calling other
accounts.
• Password: Enter the password associated with the SIP account for authenticating against the SIP server.
• Authentication ID: Enter the authentication ID used for authenticating against the SIP server. If it is the same as the
user ID, you don't need to enter the authentication ID.
• Caller ID: The name which is presented to the recipient of calls from the device.
• Registrar: Enter the IP address for the registrar.
• Transport mode: Select the SIP transport mode for the account: UPD, TCP, or TLS. When you select TLS, you get the
option to use media encryption.
• Media encryption (only with transport mode TLS): Select the type of encryption for media (audio and video) in SIP calls.
• Certificate (only with transport mode TLS): Select a certificate.
• Verify server certificate (only with transport mode TLS): Check to verify the server certificate.
27