User's Manual

GXP1760W Administration Guide
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SRTP Mode
Enable SRTP mode based on your selection from the drop-down menu.
The default setting is “Disabled”.
Crypto Life Time
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”.
Symmetric RTP
Defines whether symmetric RTP is supported or not. The default setting is No.
Silence
Suppression
Controls the silence suppression/VAD feature of the audio codec 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”.
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 in payload during the 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: The default setting is “RFC2833”.
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 uses 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.