Grandstream Networks GRP2612 Administration Manual page 39

Validate Incoming
Messages
Check SIP User ID for
Incoming INVITE
Accept Incoming SIP
from Proxy Only
Authenticate Incoming
INVITE
Account x → Audio Settings
Preferred Vocoder
Use First Matching
Vocoder in 200OK SDP
Codec Negotiation
Priority
Hide Vocoder
Disable Multiple m line
in SDP
SRTP Mode
SRTP Key Length
Crypto Life Time
Symmetric RTP
Silence Suppression
Jitter Buffer Type
Choose whether the incoming messages will be validated or not. The default
setting is "No".
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
If set to "Yes", the phone will challenge the incoming INVITE for
authentication with SIP 401 Unauthorized response. Default setting is "No".
Multiple vocoder types are supported on the phone, the vocoders in the list
is a higher preference. Users can configure vocoders in a preference list that
is included with the same preference order in SDP message.
When it is set to "Yes", the device will use the first matching vocoder in the
received 200OK SDP as the codec. The default setting is "No".
Configures the phone to use which codec sequence to negotiate as the
callee. When set to "Caller", the phone negotiates by SDP codec sequence
from received SIP Invite. When set to "Callee", the phone negotiates by audio
codec sequence on the phone. The default setting is "Callee".
When option Hide Vocoder is set as Yes, the coded will be hidden from call
screen as bellow. The default setting is "No".
When it is set to "No", the device will reply with multiple m lines; Otherwise,
it will reply 1 m line. The default setting is "No".
Enable SRTP mode based on your selection from the drop-down menu. The
default setting is "Disabled".
Allows users to specify the length of the SRTP calls. The available options
are: AES 128&256 bit, AES 128 bit and AES 256 bit.
Default setting is: AES 128&256 bit
Enable or disable the crypto life time when using SRTP. If users set to disable
this option, phone does not add the crypto life time to SRTP header. The
default setting is "Yes".
Defines whether symmetric RTP is supported or not. Default setting is "No".
Controls the silence suppression/VAD feature of the audio codecs except
forG.723 (pending) and G.729. If set to "Yes", a small quantity of RTP
packets containing comfort noise will be sent during the periods of silence. If
set to "No", this feature is disabled. Default setting is "No"
Selects either Fixed or Adaptive for jitter buffer type, based on network
conditions. The default setting is "Adaptive".
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