WorldCast Equinox - Release 3.1 - user manual
– 07/2011
Page 112
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6.3.15.5 Making the SIP call
When all of the above steps are completed (only required once) then the actual SIP call can be made.
This can also be achieved by right clicking on the phone icon on the SIP Connect tab.
The Listen button is required to be pressed first, this will send a command to the SIP server indicating
that it is now ready to receive calls. When other devices contact the server, it will be able to forward SIP
calls on.
Dial will then connect the 2 devices.
The order of events is as below:
INVITE -> (contains codec and connection information supported)
<- TRYING(comes from peer/server)
<- RINGING (requesting connect)
<- OK (filters codec and connection information supported)
-> ACK (acknowledgement)
RTP packets start.
6.3.15.6 Changing the algorithm once a SIP call is up
It is currently possible to change the algorithm when a SIP call is in progress.
If the algorithm is changed from the AUDIO SETTINGS tab in the NMS, then SIP will TRY to negotiate
the new algorithm selected “on-the-fly”.
Algorithm will be offered whether it is on the preferred list of the local unit or not.
Algorithm will only be accepted if it is in the preferred list of the receiving unit.
6.3.15.7 Dropping the SIP call
To drop the SIP call press the „Hang Up‟ button on the SIP dialler as shown below. This leaves the unit in
a state where the receive route is still active so the unit will be able to receive other incoming SIP calls.
