Grandstream Networks GXP-280 User Manual page 28

Gxp series sip enterprise phone
Hide thumbs Also See for GXP-280:
Table of Contents

Advertisement

This field contains the number of voice frames to be transmitted in a single Ethernet
Voice Frames
packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500
per TX
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 GXP 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
http://www.grandstream.com/pdf/FAQ-Codec.pdf
This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or
Layer 3 QoS
Diff-Serv or MPLS. Default value is 48.
This contains the value used for layer 2 VLAN tag. Default setting is blank.
Layer 2 QoS
Default is 4 seconds.
No Key Entry
Timeout
This parameter allows users to configure the "#" key as the "Send" (or "Dial") key. If set
Use # as
to "Yes", the "#" key will immediately send the call. In this case, this key is essentially
Dial Key
equivalent to the "(Re)Dial" key. If set to "No", the "#" key is included as part of the dial
string.
This parameter defines the local RTP-RTCP port pair used to listen and transmit. It is the
Local RTP port
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.
This parameter, when set to "Yes", will force random generation of both the local SIP
Use Random
and RTP ports. This is usually necessary when multiple GXPs are behind the same
Port
NAT. Default is No.
This parameter specifies how often the GXP sends a blank UDP packet to the SIP
Keep-alive
server in order to keep the "hole" on the NAT open. Default is 20 seconds.
interval
NAT IP address used in SIP/SDP message. Default is blank.
Use NAT IP
Grandstream Networks, Inc.
affect
the
voice
GXP User Manual
Firmware 1.1.6.46
quality.
Please
for more technical detail.
refer
to
the
Codec
Last Updated: 03/2008
FAQ
at
Page 28 of 39

Hide quick links:

Advertisement

Table of Contents
loading

Table of Contents