Spectralink 84-Series Series Wireless Telephones Administration Guide
1725-86984-000_P.docx
September 2016
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Web Info: SRTP RFC Resources
For more information on SRTP, see
. For the procedure describing how
two handsets set up SRTP for a call, see
Authentication proves to the handset receiving the RTP/RTCP streams that the packets are
from the expected source and have not been tampered with. Encryption modifies the data in the
RTP/RTCP streams so that, if the data is captured or intercepted, it sounds like noise and
cannot be understood. Only the receiver knows the key to restore the data.
A number of session parameters have been added to enable you to turn off authentication and
encryption for RTP and RTCP streams. This is done mainly to reduce the handset
’s processor
usage.
Summary
Parameter
Used to:
Enable SRTP
Include secure media in SDP of SIP INVITE
Include crypto in offered SDP
Secure media stream required in all SIP INVITEs
Check tag in crypto parameter in SDP
Specify if the handset offers and/or requires: RTP encryption, RTP
authentication, and RTCP encryption
Table 10-18: Secure Real Time Transport Protocol
Parameter
Permitted Values
Default
sec.srtp.enable
1
0 or 1
1
If 0, the handset always declines SRTP offers. If 1, the handset accepts SRTP offers.
sec.srtp.offer
1
0 or 1
0
If 1, the handset includes a secure media stream description along with the usual non-secure media description in
the SDP of a SIP INVITE. This parameter applies to the handset initiating (offering) a phone call. If 0, no secure
media stream is included in SDP of a SIP invite.
sec.srtp.offer.HMAC_SHA1_32
1
0 or 1
0
If 1, a crypto line with the
AES_CM_128_HMAC_SHA1_32
crypto-suite will be included in offered SDP. If 0, the
crypto line is not included.
sec.srtp.offer.HMAC_SHA1_80
1
0 or 1
1
If 1, a crypto line with the
AES_CM_128_HMAC_SHA1_80
crypto-suite will be included in offered SDP. If 0, the
crypto line is not included.