Grandstream Networks, Inc. | ||
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Voice Frames per
TX
Layer 3 QoS
Layer 2 QoS
Allow incoming SIP messages from SIP proxy only
Use DNS SRV
Use ID is phone number
SIP registration
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.
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:
This field defines the layer 3 QoS parameter, which can be used for IP Precedence or
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.
Default is No.
If set to Yes, the phone will use DNS SRV configured to lookup for the server
If “Yes” is set, a “user=phone” parameter will be attached to the
“From” header in SIP request, which will be processed by supported SIP proxy. Default is No.
This parameter controls whether the BudgeTone phone needs to send REGISTER messages to the proxy server.
The default setting is “Yes”.
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