Sip Settings/Tos; Table 75: Sip Settings/Tos - Grandstream Networks UCM6510 IP PBX User Manual

Local Network Address

SIP SETTINGS/TOS

ToS For SIP
ToS For RTP Audio
ToS For RTP Video
Default Incoming/Outgoing
Registration Time
Max Registration/Subscription
Time
Min Registration/Subscription
Time
Music On Hold Interpret
Music On Hold Suggest
Enable Relaxed DTMF
DTMF Mode
RTP Timeout
Firmware Version 1.0.1.12
static NAT or PAT.
Specify a list of network addresses that are considered inside of the NAT
network. Multiple entries are allowed. If not configured, the external IP
address will not be set correctly.
A sample configuration could be as follows:
192.168.0.0/16

Table 75: SIP Settings/ToS

Configure the Type of Service for SIP packets. The default setting is None.
Configure the Type of Service for RTP audio packets. The default setting is
None.
Configure the Type of Service for RTP video packets. The default setting is
None.
Configure the default duration (in seconds) of incoming/outgoing
registration. The default setting is 120.
Configure the maximum duration (in seconds) of incoming registration and
subscription allowed by the UCM6510. The default setting is 3600.
Configure the minimum duration (in seconds) of incoming registration and
subscription allowed by the UCM6510. The default setting is 60.
Configure the Music On Hold class for the channel when being put on hold.
This is used when the Music On Hold class is not set on the channel and
the peer channel placing the call on hold doesn't have "Music On Hold
Suggest".
Configure the Music On Hold class to suggest to the peer channel when
placing the peer on hold.
Select to enable relaxed DTMF handling. The default setting is "No".
Select DTMF mode to send DTMF. The default setting is RFC2833. If "Info"
is selected, SIP INFO message will be used. If "Inband" is selected, 64-kbit
codec PCMU and PCMA are required. When "Auto" is selected,
"RFC2833" will be used if offered, otherwise "Inband" will be used. The
default setting is "RFC2833".
During an active call, if there is no RTP activity within the timeout (in
seconds), the call will be terminated. The default setting is no timeout.
Note:
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