Receiving Rtp Stream - Yealink SIP-T56A User Manual

The default codec is G722.
Click Confirm to accept the change.
3.
If G722 codec is used for multicast paging, the touch screen will display the icon
Note
that it is providing high definition voice.
Default codec for multicast paging is configurable via web user interface only.

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 channel(s) that the phone listens to on the network.
Note 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.
How the phone handles incoming multicast paging calls depends on Paging Barge and Paging
Priority Active parameters configured via web user interface.
Paging Barge
The paging barge parameter defines the priority of the voice call in progress. If the priority of an
incoming multicast paging call is lower than that of the active call, it will be ignored
automatically. If Disabled is selected from the pull-down list of Paging Barge, the voice call in
progress will take precedence over all incoming multicast paging calls. Valid values in the
Paging Barge field:
1 to 31: Define the priority of the active call, 1 with the highest priority, 31 with the lowest.

Disabled: The voice call in progress will take precedence over all incoming paging calls.

Advanced Phone Features
to indicate
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