Receiving Rtp Stream - Yealink SIP-T42G User Manual

User Guide for the SIP-T42G IP Phone
The default codec is G722.
Click Confirm to accept the change.
3.
Note If G722 codec is used for multicast paging, the LCD screen will display the icon
that high definition voice is provided.
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 31
multicast addresses that the phone listens to on the network.
RTP stream is listened in the
Note
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.
136
speakerphone mode by default. If you want to listen the RTP
to indicate