Bedienungsanleitung Grandstream GXP1450 (Seite 35 von 39) (Englisch)

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GXP1450 User Manual Page 34 of 38

Firmware 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 initiator of the

conference hangs up.

Default setting is set to “No”.

Preferred Vocoder

GXP1450 supports up to 7 different Vocoder types including G.711(a/µ) (also

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 preference

order in SDP message. Enter the first Vocoder in this list by choosing the

appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this 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 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 IS limit is based on the maximum size of Ethernet packet 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 confi

gured 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 the

first codec is G.729 o

r 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) frames; for G.726, it is 20

(x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms)

and 64 (x2.5ms) frames respectively.

Please be careful when editing these parameters. Adjustin

g these parameters will

also change the dynamic jitter buffer. The GXP1450

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 dialed number.

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