GDS3710 User Manual
Version 1.0.7.8
This option allows 3rd party Service Provider or Cloud Solution to monitor
the operation status of the GDS3710 by using related SIP Calls.
By default, it’s disabled. Users can choose either RTCP or RTCP-XR.
The H.264 payload type can now be configured to be compatible with 3rd
party video phones, as well as other advanced SIP settings, to easy system
integration process. Default is 99.
Accept Incoming SIP from
Proxy Only
When set to “Yes”, the SIP address of the Request URL in the incoming
SIP message will be checked. If it doesn’t match the SIP server address of
the account, the call will be rejected. The default setting is disabled.
Select multiple audio codecs by priority order (lowest is the highest priority).
Supported codecs are: PCMU, PCMA, G.722 and G.729A/B.
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the “ptime” value for the SDP will
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the in-payload call. For end
users, it is recommended to use the default setting, as incorrect settings
may influence the audio quality.
The default setting is 2.
Range is from 1-64.
Phone Settings
The phone settings allow users to configure the GDS3710 phone settings and the White list for all the SIP
accounts.
Phone Settings
This page allows users to configure the GDS3710 phone settings.
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