Advanced Phone Features
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4.
Press the
OK
soft key to accept the change or the
Cancel
soft key to cancel.
If you want to delete all paging groups, you can press the
Del All
soft key.
You can also configure the phone to use a default codec for sending multicast RTP stream via
web user interface.
To configure a default codec for multicast paging via web user interface:
1.
Click on
Features
->
General Information
.
2.
Select the desired codec from the pull-down list of
Multicast Codec
.
The default codec is G722.
3.
Click
Confirm
to accept the change.
Note
Receiving RTP Stream
You can configure the phone to receive a Real Time Transport Protocol (RTP) stream from the
pre-configured multicast address(es) and channel(s) without involving SIP signaling. You can
specify up to 31 multicast addresses and channels that the phone listens to on the network.
Note
If G722 codec is used for multicast paging,
the LCD screen will display the icon to indicate
that high definition voice is provided.
Default codec for multicast paging is configurable via web user interface only.
RTP stream is listened in the hands-free (speakerphone) mode by default. If you want to listen the
RTP stream using the engaged audio device (speakerphone, handset or headset), contact your
system administrator for more information.
Fixed volume to play RTP stream for specified paging group is configurable by your system
administrator.
