AudioCodes Mediant 800B - Page 1099

Version 7.2 1099 Mediant 800B Gateway & E-SBC
User's Manual 72. Configuration Parameters Reference
Parameter Description
configure voip > sbc settings
> sbc-dtls-mtu
[SbcDtlsMtu]
Defines the maximum transmission unit (MTU) size for the DTLS
handshake. The device does not attempt to send handshake packets
that are larger than the configured value. Adjusting the MTU is useful
when there are network constraints on the size of packets that can be
sent.
The valid value range is 228 to 1500. The default is 1500.
Note: The parameter is applicable only to the SBC application.
Aria Protocol Support
configure voip > media
security > ARIA-protocol-
support
[AriaProtocolSupport]
Enables ARIA algorithm cipher encryption for SRTP. This is an
alternative option to the existing support for the AES algorithm. ARIA
is a symmetric key block cipher algorithm standard developed by the
Korean National Security Research Institute.
 [0] Disable (default)
 [1] Enable
Note:
 To configure the ARIA bit-key encryption size (128 or 192 bit) with
HMAC SHA-1 cryptographic hash function, use the
SRTPofferedSuites parameter.
 The ARIA feature is available only if the device is installed with a
License Key that includes this feature. For installing a License
Key, see License Key on page 857.
Disable Authentication On
Transmitted RTP Packets
configure voip > media
security > RTP-
authentication-disable-tx
[RTPAuthenticationDisableTx]
Enables authentication on transmitted RTP packets in a secured RTP
session.
 [0] Enable (default)
 [1] Disable
Disable Encryption On
Transmitted RTP Packets
configure voip > media
security > RTP-encryption-
disable-tx
[RTPEncryptionDisableTx]
Enables encryption on transmitted RTP packets in a secured RTP
session.
 [0] Enable (default)
 [1] Disable
Disable Encryption On
Transmitted RTCP Packets
configure voip > media
security > RTCP-encryption-
disable-tx
[RTCPEncryptionDisableTx]
Enables encryption on transmitted RTCP packets in a secured RTP
session.
 [0] Enable (default)
 [1] Disable
configure voip > sip-definition
settings > srtp-state-behavior-
mode
[ResetSRTPStateUponRekey]
Global parameter that enables synchronization of the SRTP state
between the device and a server when a new SRTP key is generated
upon a SIP session expire. You can also configure this functionality
per specific calls, using IP Profiles
(IpProfile_ResetSRTPStateUponRekey). For a detailed description of
the parameter and for configuring this functionality in the IP Profiles
table, see 'Configuring IP Profiles' on page 436.
Note: If this functionality is configured for a specific IP Profile, the
settings of this global parameter is ignored for calls associated with
the IP Profile.