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User Guide for the SIP-T48S IP Phone
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If you want to delete all paging groups, you can tap the Delete 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 touch screen will display the icon to indicate
that it is providing high definition voice.
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.
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