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GXP17xx Administration Guide
audio packets) will be sent during the period of no talking.
If set to “No”, this
feature is disabled.
The default setting is “No”.
Jitter Buffer Type
Selects either Fixed or Adaptive for jitter buffer type, based on network
conditions. The default setting is
“Adaptive”.
Jitter Buffer Length
Selects jitter buffer length from 100ms to 800ms, based on network conditions.
The default setting is “300ms”.
Voice Frames Per
TX
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. The default setting is 2.
G723 Rate
Configure encoding rate for G723.1 codec.
G.726-32 Packing
Mode
Selects "ITU" or "IETF" for G726-32 packing mode. The default setting is “ITU”.
iLBC Frame Size
Configure iLBC packet frame size.
iLBC Payload Type
Specify iLBC payload type. Valid type is 96-127.
OPUS Payload Type
Specifies OPUS payload type. Valid range is 96 to 127. Cannot be the same as
iLBC or DTMF Payload Type.
DTMF Payload Type
Configures the payload type for DTMF using RFC2833. Cannot be the same as
iLBC or OPUS payload type
Send DTMF
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
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
Early Dial
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
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".