Axis C1111-E User Manual page 34

Cabinet speaker
Hide thumbs Also See for C1111-E:
Table of Contents

Advertisement

AXIS C1111-E Cabinet Speaker
The web interface
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.
• Max retransmissions: Set the maximum number of times the device tries to connect to the SIP server before it
stops trying.
• Seconds until failback: Set the number of seconds until the device tries to reconnect to the primary SIP server after it
has failed over to a secondary SIP server.
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.
Add 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.
• Answer automatically: Select to automatically answer an incoming call.
• Prioritize IPv6 over IPv4
connect to peer-to-peer accounts or domain names that resolve in both IPv4 and IPv6 addresses. You can only
prioritize IPv6 for domain names that are mapped to IPv6 addresses.
• 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.
• TLS version (only with transport mode TLS): Select the version of TLS to use. Versions v1.2 and v1.3 are the most
secure. Automatic selects the most secure version that the system can handle.
• Media encryption (only with transport mode TLS): Select the type of encryption for media (audio and video) in SIP calls.
: Select to prioritize IPv6 addresses over IPv4 addresses. This is useful when you
34

Advertisement

Table of Contents
loading

Table of Contents