PBX Networking
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transfers the voice data directly from terminal to terminal via the RTP protocol. In
certain cases, for example, when an incoming external call is placed via multiple TK
systems, one or more RTP proxies may be used to forward the connection.
Currently, there are no standards for the necessary extensions to the Q.SIG protocol.
This means that you can only use Q.SIG-IP between Aastra 800 and OpenCom
systems.
Networking two OpenCom 100 systems using Q.SIG-IP requires 2 licences – one
licence per system. The number of possible voice connections is not restricted by the
licence.
Go to the Telephony: Trunks: Trunk group page in the Configurator to set up a
Q.SIG-IP connection. Create a new bundle and select the Access type “System
Access”. Select “Q.SIG-IP” under Protocol. Configure the IP address of the other
system, the port numbers to be used, the number of possible voice connections.
Select a VoIP profile for the codec selection. Please refer to the relevant help topics in
the Online Help for the OpenCom 100 as well.
Note
Q.SIG-IP cannot be operated using a connection with NAT. For a Q.SIG-IP connection,
a branch connection or another VPN connection is required.
Connection via SIP tie line
The OpenCom 100 communications system supports connections using a SIP tie line
for TC system networking. A SIP tie line is a SIP line which requires no login which can
establish multiple call connections simultaneously. No SIP provider is required for
establishing the connection via SIP tie line.
The number of simultaneous calls possible depends on the network or internet con-
nection capacity and the compression procedure being used. The channels of a media
gateway card are used for a SIP tie line (see MGW Interface Card starting on page 119).
The data of a SIP tie line connection is subjected to codec compression (see under
Fundamentals starting on page 110 in the chapter Voice over IP (VoIP)). Call data is
transmitted directly from terminal to terminal via the RTP protocol with SIP tie line as
well. In certain cases, for example when an incoming external call is switched via mul-
tiple TC systems, there may be one or multiple RTP proxies involved.
One of the special features of a SIP tie line connection is using transparent codec
interconnecting, for example to make use of HQ audio or video telephony with appro-
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