SpectraLink 8440 - Dot1 X><Eapollogoff

Spectralink 84-Series Series Wireless Telephones Administration Guide
1725-86984-000_P.docx
September 2016 222
Parameter
Permitted values
Defaults
sec.srtp.mki.enabled
1
0 or 1
0
The master key identifier (MKI) is an optional parameter for the cryptographic parameter in the SDP that uniquely
identifies the SRTP stream within an SRTP session. MKI is expressed as a pair of decimal numbers in the form:
|mki:mki_length| where mki is the MKI value and mki_length its length in bytes. If 1, a four-byte MKI
parameter is sent within the SDP message of the SIP INVITE / 200 OK. If 0, the MKI parameter is not sent.
sec.srtp.mki.length
1
1 to 4
4
The length of the master key identifier (MKI), in bytes. Microsoft Skype for Business offers 1-byte MKIs.
sec.srtp.mki.startSessionAtOne
0 or 1
0
If set to 1, use an MKI value of 1 at the start of an SDP session. If set to 0, the MKI value will increment for each
new crypto key.
sec.srtp.resumeWithNewKey
1
0 or 1
1
If 0, a key is not provided when resuming a call. If 1, a key is provided when resuming a call.
sec.srtp.simplifiedBestEffort
0 or 1
0
If 0, no SRTP is supported. If 1, negotiation of SRTP compliant with Microsoft Session Description Protocol
Version 2.0 Extensions is supported.
1
Change causes handset to restart or reboot.
<dot1x><eapollogoff/>
Table 10-17: 802.1X EAP over LAN (EAPOL) Logoff Parameters
Parameter
Permitted Values
Default
sec.dot1x.eapollogoff.enabled
1
0 or 1
0
If 0, the handset will not send an EAPOL Logoff message on behalf of the disconnected supplicant. If 1, the
feature is enabled and the handset will send an EAPOL Logoff message on behalf of the disconnected supplicant
connected to the handset’s secondary (PC) port.
sec.dot1x.eapollogoff.lanlinkreset
1
0 or 1
0
If 0, the handset software will not reset (recycle) the LAN port link in the application initiation stage. If 1, the LAN
port link will be reset in the application initiation stage.
1
Change causes handset to restart or reboot.
Secure Real-Time Transport Protocol
Secure Real-Time Transport Protocol (SRTP) provides a way of encrypting audio stream(s) to
avoid interception and eavesdropping on phone calls. As described in RFC 3711, both RTP and
RTCP signaling may be encrypted using an AES (advanced encryption standard) algorithm. The
parameters used to configure SRTP are shown in Table 10-18: Secure Real Time Transport
Protocol. When this feature is enabled, handsets will negotiate with the other end-point the type
of encryption and authentication to use for the session. This negotiation process is compliant
with RFC4568 —Session Description Protocol (SDP) Security Descriptions for Media Streams.