Polycom IP 320 Administrator's Manual page 290

Administrator's Guide for the SoundPoint IP / SoundStation IP / VVX Family
Note
Attribute
sec.srtp.enable
sec.srtp.offer
sec.srtp.require
sec.srtp.offer.
HMAC_SHA1_80
sec.srtp.offer.
HMAC_SHA1_32
A - 110
SRTP <srtp/>
As per RFC 3711, you cannot turn off authentication of RTCP.
This configuration attribute is defined as follows:
Permitted
Values Default
0 or 1 Null
0 or 1 Null
0 or 1 Null
0 or 1 Null
0 or 1 Null
.
Interpretation
If set to 1 or Null, the phone accepts SRTP offers.
If set to 0, the phone always declines SRTP offers.
Note: The default behavior changed in SIP 3.2.0 .
In previous SIP releases, the default value was 0
when null or not defined.
If set to 1 or Null, the phone includes a secure
media stream description along with the usual
non-secure media description in the SDP of a SIP
INVITE. This is for the phone initiating (offering) a
phone call.
If set to 0, no secure media stream is included in
SDP of a SIP invite.
If set to 1, the phone is only allowed to use secure
media streams. Any offered SIP INVITEs must
include a secure media description in the SDP or
the call will be rejected. For outgoing calls, only a
secure media stream description is include in the
SDP of the SIP INVITE, meaning that the
non-secure media description is not included. If
sec.srtp.require is set to 1, sec.srtp.offer is
logically set to 1 no matter what the value in the
configuration file.
If set to 0 or Null, secure media streams are not
required.
If set to 1 or Null, a crypto line with the
AES_CM_128_HMAC_SHA1_80 crypto-suite will
be included in offered SDP.
If set to 0, the crypto line is not included.
Note: This parameter was added in SIP 2.2.1 .
If set to 1, a crypto line with the
AES_CM_128_HMAC_SHA1_32 crypto-suite will
be included in offered SDP.
If set to 0 or Null, the crypto line is not included.
Note: This parameter was added in SIP 2.2.1 .