Grandstream Networks GXP1760 Administration Manual page 29

audio packets) will be sent during the period of no talking. If set to "No", this
feature is disabled. The default setting is "No".
Selects either Fixed or Adaptive for jitter buffer type, based on network
Jitter Buffer Type
conditions. The default setting is "Adaptive".
Selects jitter buffer length from 100ms to 800ms, based on network conditions.
Jitter Buffer Length
The default setting is "300ms".
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
Voice Frames Per
with different configurations here. This value is related to the codec used and
TX
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. The default setting is 2.
G723 Rate Configure encoding rate for G723.1 codec.
G.726-32 Packing
Selects "ITU" or "IETF" for G726-32 packing mode. The default setting is "ITU".
Mode
Configure iLBC packet frame size.
iLBC Frame Size
iLBC Payload Type Specify iLBC payload type. Valid type is 96-127.
Specifies OPUS payload type. Valid range is 96 to 127. Cannot be the same as
OPUS Payload Type
iLBC or DTMF Payload Type.
Configures the payload type for DTMF using RFC2833. Cannot be the same as
DTMF Payload Type
iLBC or OPUS payload type
This parameter 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.
• In audio, which means DTMF is combined in the audio signal (not very reliable
Send DTMF with low-bit-rate codecs).
• RFC2833, which means to specify DTMF with RTP packet. Users could know
the packet is DTMF in the RTP header as well as the type of DTMF.
• SIP INFO, which use SIP info to carry DTMF. The defect of this mode is that
it's easily to cause desynchronized of DTMF and media packet for the reason
the SIP and RTP are transmitted respectively. The default setting is "RFC2833".
Account x  Call Settings
Selects whether or not to enable early dial. If it's set to "Yes", the SIP proxy
must support 484 responses. Early Dial means that the phone sends for each
pressed digit a SIP INVITE message to SIP server. SIP server looks into its
Early Dial
extensions and, if no match happened yet, it sends back a "484 Address
Incomplete" message. Otherwise, it executes the action. The default setting is
"No".
GXP17xx Administration Guide
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