Grandstream Networks GXP1610 Administration Manual page 28

Validate Incoming
Messages
Check SIP User ID for
incoming INVITE
Accept Incoming SIP
from Proxy Only
Authenticate Incoming
INVITE
Account x ->Audio Settings
Send DTMF
DTMF Payload Type
Preferred Vocoder
Use First Matching
Vocoder in 200OK
SDP
Disable Multiple m line
in SDP
SRTP Mode
Symmetric RTP
Silence Suppression
Voice Frames Per TX
G.726-32 Packing
Mode
Jitter Buffer Type
Jitter Buffer Length
Account x -> Call Settings
Early Dial
Dial Plan Prefix
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. The default setting is
"No".
Specifies the mechanism to transmit DTMF digits. There are 3 supported
modes: in audio which means DTMF is combined in the audio signal (not
very reliable with low-bit-rate codecs), via RTP (RFC2833), or via SIP INFO.
Configures the payload type for DTMF using RFC2833. The default value is
101.
6 different vocoder types are supported on the phone, including G.711 U-law
(PCMU), G.711 A-law (PCMA), G.723.1(pending), G.729A/B, G.722 (wide
band), and G726-32. Users can configure vocoders in a preference list that
is included with the same preference order in SDP message.
When set to "Yes", the device will use the first matching vocoder in the
received 200OK SDP as the codec. The default setting is "No".
If enabled, the phone always responses 1 m line in SDP regardless multiple
m lines are offered.
Enables the SRTP mode based on your selection. The default setting is
"Disabled".
Defines whether symmetric RTP is supported or not. The default setting is
"No".
Controls the silence suppression/VAD feature of the audio codec except for
G.723 (pending) and G.729. If set to "Yes", when silence is detected, a small
quantity of VAD packets (instead of audio packets) will be sent during the
period of no talking. If set to "No", this feature is disabled. The default setting
is "No".
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the in payload call. For end
users, it is recommended to use the default setting, as incorrect settings may
influence the audio quality.
Selects "ITU" or "IETF" for G.726-32 packing mode. The default setting is
"IETF".
Selects either Fixed or Adaptive based on network conditions. The default
setting is "Adaptive".
Defines jitter buffer length based on network conditions.
Selects whether or not to enable early dial. If it's set to "Yes", the SIP proxy
must support 484 response. The default setting is "No".
Sets the prefix added to each dialed number.
GXP1610/GXP1620/GXP1625/GXP1628/
GXP1630 Administration Guide
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