Related Features
System Networking
H.323 v4 Service
Hardware
SIP Trunk
4.12.3 SIP Virtual CO Line (VCOL)
Description
Normally, SIP Trunks employ a VoIP channel for connection between the SIP Trunk and the
system device placing or receiving the call. The VoIP channel acts as a relay point for port
forwarding, conversion of signaling protocols, and a DSP is used to convert between Codecs
and to deliver certain system tones. When the network allows for a direct connection between
the device and the SIP trunk, and the same codec is supported by both ends a virtual SIP
Line can be used. In this case, the system does not assigned a VoIP channel for RTP relay,
a DSP for voice processing, and DTMF signaling must be set to out-of band.
Operation
Conditions
Operation of Virtual SIP CO Line Service is automatic.
1. DTMF tone detection is not supported, only out-band DTMF type (SIP Info
Message) is supported.
2. SIP phones require an available VoIP channel, built-in the UCP or from a VOIM, to
relay Page announcements, BGM and MOH packets.
3. A system VoIP channel is required to support local features. A VoIP channel is
required to serve the below features.
a. DSP for generation of Busy/Error/Confirm/Ring-Back/Hold/ Page/Warning/
OHVA/Intrusion/Dial tones from system to SIP Trunk
b. Relay of Music On Hold from system to SIP Trunk
c. Relay of Paging from/to SIP Trunk
d. Voice RTP Packet Relay between private LAN and public WAN, local and
remote, NAT resolution
A physical VoIP channel will be provided dynamically by the system to support
these functions. It is recommended that the system be equipped with a number of
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