Grandstream Networks, Inc. GXP21xx User Manual Page 37 of 41
Firmware version: 1.0.1.66 Last Updated: 05/2011
Refer-To Use Target
Contact
Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header
uses the transferred target’s Contact header information.
Transfer on Conference
Hangup
Defines whether or not the call is transferred to the other party if the init iator of the
conference hangs up.
Default setting is set to “No”.
Preferred Vocoder
GXP21xx supports up to 7 different Vocoder types including G.711(a/µ) (a lso known
as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).
Configure Vocoders in a preference list that is included with the same pr eference
order in SDP message. Enter the first Vocoder in this list by choosing th e
appropriate option in “Choice 1”. Similarly, enter the last Vocoder in th is list by
choosing the appropriate option in “Choice 8”.
SRTP Mode
Enable SRTP mode based on selection. Default is “No”.
Symmetric RTP
Selects whether or not symmetric RTP is supported.
Silence Suppression
This controls the silence suppression/VAD feature of the audio codec G.723 and
G.729. If set to “Yes”, when silence is detected, a small quantity of VAD pac kets
(instead of audio packets) will be sent during the period of no talking. If s et to “No”,
this feature is disabled.
Voice Frames per TX
This field contains the number of voice frames to be transmitted i n a s in gl e Ether net
packet (be advised the IS limit is based on the maximum size of Ethernet pac
ket is
1500 byte (or 120kbps)).
When setting this value, be aware of the requested packet time (ptime, used in SDP
message) is a result of configuring this parameter. This parameter is associated
with the first codec in the above codec Preference List or
the actual used payload
type negotiated between the 2 conversation parties at run time. E.g.
, if the first
codec is configured as G.723 and the “Voice Frames per TX” is set to 2, then the
“ptime” value in the SDP message of an INVITE request will be 60ms
because each
G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and t he
first codec is G.729 or G.711 or G.726, then the “ptime” value in the SDP message
of an INVITE request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the IP
phone will use and save the maximum allowed value for the corresponding first
codec choice. The maximum value for PCM is 10 (x10ms) fram es; for G.726, it is 20
(x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10 ms)
and 64 (x2.5ms) frames respectively.
Please be careful when editing these parameters. Adjusting these p arameters will
also change the dynamic jitter buffer. The GXP21xx has a patent dynamic
jitter
buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.
We recommend using the default settings provided. We do not recommend
adjusting these parameters if you are an average user. Incorrect settings will affect
the voice quality.
No Key Entry Timeout
Default is 4 seconds. After the timeout, the phone will send out the diale d number.