Jensen Tools 100 Sereis - Manual (Page 32)

BudgeTone-100 User Manual
Grandstream Networks, Inc.
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iLBC frame size iLBC packet frame size. Default is 20ms.
For Asterisk IP-PBX, 30ms might need to be configured for compatibility
iLBC payload
type Payload type for iLBC. Default value is 97.
The valid range is between 96 and 127.
Silence
Suppression This controls the silence suppression/VAD feature of G723 and G729. If set to
“Yes”, when a silence is detected, 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.
Voice Frames per
TX This field contains the number of voice frames to be transmitted in a single
Ethernet packet (be advised the max. size of Ethernet packet is 1500 byte (or
120k bit) so user should be aware that there IS a limit there). When setting this
value, the user should be aware of the requested packet time (ptime, used in SDP
message) as 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 G723 and the “Voice Frames per TX” is
set to be 2, then the “ptime” value in the SDP message of an INVITE request
will be 60ms because each G723 voice frame contains 30ms of audio. Similarly,
if this field is set to be 2 and if the first codec chosen is G729 or G711 or G726,
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) frames; for
G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for
G729/G728, 64 (x10ms) and 64 (x2.5ms) frames respectively.
Please be very careful when massage those parameters. By adjust this, user also
get jitter buffer changed accordingly. BT-100 phone has patent dynamic jitter
buffer handling algorithm. The jitter buffer range from 20 ~ 200 ms.
Incorrect setting will affect voice quality so do not touch the parameter if not
understand and most of the case the default value will work in GS products.
Please refer to the Codec FAQ in our website for more technical details:
http://www.grandstream.com/FAQ-Codec.pdf
Layer 3 QoS This field defines the layer 3 QoS parameter, which can be used for IP
Precedence or Diff-Serv or MPLS. Default value is 48.
Layer 2 QoS Layer 2 QoS settings. Default setting is blank or “0”
Other VLAN supported equipments like VLAN switch/router required if user
wants to configure these settings.
Allow incoming
SIP messages
from SIP proxy
only If set to “Yes”, the phone will ignore any SIP message that does not come from
the IP address (Source IP in the IP header, the SIP server) that it is registered to.
Default is No.