Grandstream Networks BudgeTone-200 Bedienungsanleitung PDF herunterladen (Seite 24)

Grandstream Networks, Inc. BT200 User Manual Page 24 of 33

Firmware 1.1.5.15 Last Updated: 01/2008

iLBC frame size

iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required.

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 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

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 the 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) 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. Adjusting these parameters will also

change the dynamic jitter buffer. The BT200 has a patent dynamic jitter buffer handling

algorithm. The jitter buffer range is 20 ~ 200 ms.

Grandstream recommends using the default settings provided. Grandstream does not

recommend adjusting these parameters if you are an average user. Incorrect settings

will affect the voice quality. Please refer to the Codec FAQ at

http://www.grandstream.com/faqscodec.html for more technical detail.

Layer 3 QoS

This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or

Diff-Serv or MPLS. Default value is 48.

Layer 2 QoS

This contains the value used for layer 2 VLAN tag. Default setting is blank.

No Key Entry

Timeout

Default is 4 seconds.

Use # as

Dial Key

This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If set

to “Yes”, the “#” key will immediately send the call. In this case, this key is essentially

equivalent to the “Dial” key. If set to “No”, the “#” key is included as part of the dial string.

Local RTP port

This parameter defines the local RTP-RTCP port pair used to listen and transmit. It is the

base RTP port for channel 0. When configured, channel 0 will use this port _value for

RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and

port_value+3 for its RTCP. The default value is 5004.

1 2 ... 19 20 21 22 23 24 25 26 27 28 29 ... 32 33

Kommentare zu diesen Handbüchern

Keine Kommentare

Grandstream Networks BudgeTone-200 Bedienungsanleitung Seite 24

Kommentare zu diesen Handbüchern

Verwandte Produkte und Handbücher für IP-Telefone Grandstream Networks BudgeTone-200