Receiving Rtp Stream - Yealink T41P Skype User Manual

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.
Paging list is configurable via web user interface at the path Directory->Multicast IP.
You can also configure the phone to use a codec for sending multicast RTP stream via web user
interface.
To configure a default codec for multicast paging via web user interface:
Click on Features->General Information.
1.
Select the desired codec from the pull-down list of Multicast Codec.
2.
The default codec is G722.
Click Confirm to accept the change.
3.
If G722 codec is used for multicast paging, the LCD screen will display the
Note
indicate 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) without involving SIP signaling. You can specify up to 10
multicast addresses that the phone listens to on the network.
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
Basic Call Feature
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