Grandstream Networks GXV3175 User Manual page 53

Ip multimedia phone
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H.263 En
ncoder Resol
lution
SRTP M
ode
Silence S
Suppression
Voice Fr
ame per TX
G723 Ra
ate
Jitter Buf
ffer Type
Jitter Buf
ffer Length
Account/
/Call Settings
s
Dial Plan
n Prefix
Grandstr
ream Networ
ks, Inc.
Selec
cts the resol
ution (CIF/Q
resolu
ution is CIF.
This i
s option to se
ecure the RTP
This c
controls the G
G723/G729 si
no au
udio is detect
ted, the phon
(inste
ead of audio p
packets). If Se
settin
g is NO.
This c
configures th
he number of
recom
mmended to s
set this based
Byte o
or 120kb/s).
When
n configuring
this, it should
will ch
hange with d
different confi
codec
c used and th
e actual fram
For ex
xample: if the
e codec is con
TX" is
s set to 2. The
en in the INVI
60 ms
s. This is beca
ause the G723
is 2 a
nd the codec
is G729 or G
INVIT
TE request wi
ll be 20 ms.
If the
e voice frame
es per TX c
allowe
ed, the phone
e will choose
value
allowed.
The m
maximum valu
ue is 64 frame
Adjus
sting these va
alues will also
algori
thm for setti
ing the jitter
20-20
00ms.
For e
end-users, it i
is recommen
settin
gs may influe
ence the quali
This d
defines the e
encoding rate
5.3Kb
b/s. Please co
onfirm with yo
Users
s can select e
either Fixed or
defau
ult is Adaptive
e.
User
can select L
ow, Medium,
defau
ult is Medium.
Sets t
the prefix add
ded to each d
GXV3175
USER MANU
FIRM
MWARE VER
RSION 1.0.3.4
QCIF) used f
for H.263 co
P audio
ilence suppre
ession/VAD. If
ne will send a
a small numb
et to NO, this
feature is dis
f voice frame
es transmitted
d on the Ethe
ernet IS packe
d be noted tha
at the "ptime"
gurations he
re. This valu
mes transmitte
ed during the
nfigured to be
e G.723 and
TE request, t
the "ptime" va
3 voice frame
e is 30 ms. Sim
711 or G726,
the "ptime" v
configuration
exceeds the
e the preferre
d codec and
es.
o change the
jitter buffer.
buffer and t
the jitter buff
nded to use t
the default se
ity of audio.
e for G723 vo
ocoder, and
our service pro
ovider.
r Adaptive ba
ased on netwo
, or High bas
sed on netwo
ialed number
r.
UAL
40
odec. The d
default
f set to YES,
when
er of VAD pa
ackets
sabled. The d
default
d per packet.
. (It is
et limitation –
– 1500
" value for the
e SDP
e is related t
to the
in payload ca
all.
"Voice Frame
es per
alue for SDP w
will be
milarly, if the s
setting
value for SDP
in the
e maximum
value
use the max
ximum
The phone h
as an
ffer rate is a
round
etting, as inc
correct
can be 6.3K
b/s or
ork conditions
s. The
ork conditions
s. The
P
Page 53 of 13
30
Updated : 11
/2011

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