Yamaha CS-700 Operation Manual page 69
Video sound collaboration system for huddle rooms
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Also See for CS-700:
- Integrators reference manual (166 pages) ,
- Operation manual (145 pages) ,
- Quick start manual (76 pages)
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Property
Type
voip.use-srtp
INDEXED_
OPTION
voip.rtp-port
NUMBER
voip.set-qos
BOOLEAN
voip.udp-tcp-
BOOLEAN
selection
voip.local-port
NUMBER
voip.bound-addr
TEXT
voip.no-refer-sub
BOOLEAN
Values
0 : Disabled -- Do not use SRTP;
always use RTP
1 : Optional -- Use the optional
disposition for SRTP in SDP; if the
remote end supports SRTP then
use SRTP; otherwise use RTP
2 : Mandatory -- Force use of
SRTP; if the remote end does not
support SRTP the call does not
connect
1024-65535 : Port number
0 : Disable QoS
1 : Enable QoS
0 : UDP
1 : TCP
1024-65535 : Port number
ip-address : IP address
0 : Disable suppressing
subscription
1 : Enable suppressing
subscription
Default Value
Description
0
Controls Secure Real-time Transport Protocol (SRTP) usage. These
are the available options: 0 - Disabled - Do not use SRTP; always
use RTP. 1 - Optional - Use the optional disposition for SRTP in SDP.
If the remote end supports SRTP, then use SRTP; otherwise, use
RTP. 2 - Mandatory - Force use of SRTP. If the remote end does not
support SRTP, the call does not connect.
4000
Base port number for VoIP 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.
0
Option to enable QoS (Quality of Service) tagging for SIP and
media. For layer 3, at the Internet layer, the DiffServ
(Differentiated Services) precedence level is Class 3. The
Differentiated Services Code Point (DSCP) in the IP header is set to
24 (0x18). For layer 2, IEEE 802.1p tagging is supported.
0
Transport that will be used for SIP messages, 0 - UDP, 1 - TCP.
5060
Specifies the local port for SIP transport. The range is 1024 to
65535.
Configure bound IP address we intend to use.
0
Enable or disable suppress subscription during transfer. 0 - disable,
1 - enable. When transferring a SIP call, the SIP REFER process
automatically establishes a temporary event subscription to notify
the party initiating the transfer about the receiver's status in
handling the REFER. In some cases these event subscriptions and
notifications are not needed, such as when forking is not used.
Enable this option to suppress the automatic event subscriptions
when transferring calls. The option is disabled by default. (See IETF
RFC 4488.)
69
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