Config attributes
Default
Value
Type
Values
Description
added.
voip.vm-number
TEXT
""
Provides the number that is dialed when voicemail is called from the main
menu.
voip.use-srtp
0
INDEXED_
OPTION
0="Diabled"
1="Optional"
2="Mandatory"
Controls Secure Real-time Transport Protocol (SRTP) usage. These are the
available options: Disabled - Do not use SRTP; always use RTP. Optional -
Use the optional disposition for SRTP in SDP. If the remote end supports
SRTP, then use SRTP; otherwise, use RTP. Mandatory - Force use of SRTP. If
the remote end does not support SRTP, the call does not connect.
voip.rtp-port
4000
NUMBER
1024 - 65535
This is the base port number for RTP. RTP is originated and received on
even port numbers, and the associated RTCP uses the next higher odd port
number. The range is 1024 to 65535.
voip.udp-tcp-
selection
0
INDEXED_
OPTION
0="UDP"
1="TCP"
Selects the transport that will be enabled for SIP messages.
voip.local-port
5060
NUMBER
1024 - 65535
Specifies the local port for SIP transport. The range is 1024 to 65535.
voip.stun-srv
TEXT
""
Optional. Specifies the STUN (Session Traversal Utilities for NAT) server to
use to determine if the phone is behind a NAT, the type of NAT, and the
public address of the phone. The field can contain a comma separated list
of servers. Each server can be a domain name, host name, or IP address,
and it may contain an optional port number. (For STUN see IETF RFC 5389.)
voip.use-ice
0
BOOLEAN
0="False"
1="True"
This option enables the use of the ICE (Interactive Connectivity
Establishment) protocol for NAT traversal. ICE takes advantage of STUN and
TURN to identify candidates (IP addresses and ports) for communication,
evaluating and prioritizing the candidate pairs to select the best route.
Expensive candidates, such as using a media relay, are selected only as a
last resort. (For ICE see IETF RFC 5245.)
