BudgeTone-100 User Manual

Grandstream Networks, Inc.

 

 

 

iLBC frame size

iLBC payload type

Silence

Suppression

Voice Frames per

TX

Layer 3 QoS

Layer 2 QoS

Allow incoming SIP messages from SIP proxy only

iLBC packet frame size. Default is 20ms.

For Asterisk IP-PBX, 30ms might need to be configured for compatibility

Payload type for iLBC. Default value is 97.

The valid range is between 96 and 127.

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.

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

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 settings. Default setting is blank or “0”

Other VLAN supported equipments like VLAN switch/router required if user wants to configure these settings.

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.

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Jensen Tools 100 Sereis user manual Grandstream Networks, Inc