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 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 affect the voice quality. Please refer to the Codec FAQ

 

 

 

 

at http://www.grandstream.com/pdf/FAQ-Codec.pdf

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.

 

 

 

 

 

 

 

 

 

 

 

 

Data VLAN Tag

 

Default is 0. Enabling the Data VLAN filtering will help reduce the load on the

 

 

 

 

phone, but it isn’t necessary in most environments. This is primarily for VLAN

 

 

 

 

filtering where tagged traffic will be forwarded to the DSP.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

No Key Entry Timeout

 

Default is 4 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

Use # as

 

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

 

 

Dial Key

 

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

 

 

 

 

key is essentially equivalent to the “(Re)Dial” key. If set to “No”, the “#” key is

 

 

 

 

included as part of the dial string.

 

 

 

 

 

 

 

 

 

 

 

 

Local RTP port

 

This parameter defines the starting 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.

 

 

 

 

 

 

 

 

 

 

 

 

Use Random Port

 

This parameter, when set to “Yes”, will force random generation of both the local

 

 

 

 

SIP and RTP ports. This is usually necessary when multiple GXPs are behind

 

 

 

 

the same NAT. Default is No.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

GXP User Manual

Page 32 of 44

 

 

 

Firmware 1.2.5.3

Last Updated: 03/2011

Page 31 Page 33
Page 32
Image 32
Grandstream Networks GXP2010 Voice Frames per TX, Layer 3 QoS, Layer 2 QoS, Data Vlan Tag, No Key Entry Timeout, Use # as
Page 31 Page 33

GXP1200, GXP2000, GXP280/GXP285, GXP2020, GXP2010 specifications

The Grandstream Networks GXP2010 is a versatile and reliable SIP telephone designed for small to medium-sized businesses. With its robust feature set and user-friendly interface, the GXP2010 stands out in the world of IP telephony. This model accommodates the growing demand for efficient communication tools while blending seamlessly into various business environments.

One of the defining characteristics of the GXP2010 is its ability to support multiple SIP accounts. It can manage up to four SIP accounts, allowing users to handle multiple lines of communication efficiently. This feature is especially beneficial for organizations that often deal with a high volume of calls or need to maintain different lines for various departments.

The GXP2010 boasts high-definition audio quality, utilizing Wideband CODEC technology to ensure crystal clear voice communication. This technology helps eliminate background noise, producing a more pleasant and professional experience during calls. The device also includes various audio codecs, providing flexibility to suit different network environments.

In terms of design, the GXP2010 features a user-friendly interface with a backlit LCD display that offers intuitive navigation. The phone is equipped with programmable keys, enabling users to customize their experience and access frequently used functions quickly. The inclusion of HD handset and speakerphone capabilities ensures high audio quality regardless of the speaking method preferred by the user.

Furthermore, the GXP2010 supports Power over Ethernet (PoE), simplifying the installation process by reducing the number of cables needed. This feature allows the phone to receive power and network connectivity through a single Ethernet cable, resulting in a more organized workspace.

Security is another priority for Grandstream, and the GXP2010 reflects this focus with advanced encryption protocols for secure communications. The device supports TLS and SRTP for secure signaling and media encryption, safeguarding sensitive business conversations.

The GXP2010 is not only compatible with a variety of SIP-based platforms, but it also includes support for advanced telephony features such as call forwarding, call transfer, and call waiting. This versatility makes it an excellent choice for businesses seeking to streamline their communication processes.

In conclusion, the Grandstream Networks GXP2010 is an ideal solution for businesses looking for a reliable, high-quality SIP phone that combines versatility and advanced technology. Its multiple SIP account support, high-definition audio capabilities, user-friendly design, and strong security measures make it a valuable addition to any modern office environment.
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