Account/Audio Settings - Grandstream Networks WP810 Administration Manual

Validate Certificate
Chain
Validate Incoming
Messages
Check SIP User ID for
Incoming INVITE
Accept Incoming SIP
from Proxy Only
Authenticate Incoming
INVITE

Account/Audio Settings

Preferred Vocoder
Use First Matching
Vocoder in 200OK SDP
Codec Negotiation
Priority
Disable Multiple m line
in SDP
SRTP Mode
SRTP Key Length
Crypto Lifetime
Symmetric RTP
Silence Suppression
Validate certification chain when TCP/TLS is configured. Default setting is
"No".
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 does not 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 does not 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 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 lifetime when using SRTP. If users set to disable
this option, phone does not add the crypto lifetime 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"
WP810 Administration Guide
Version 1.0.1.1
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