Yealink SIP-T41P User Manual page 123

The default codec is G722.
3. Click Confirm to accept the change.
Note
If G722 codec is used for multicast paging, the LCD screen will display the
indicate that it is providing high definition voice.
Default codec for multicast paging is configurable via web user interface only.
To send RTP stream via a multicast paging key:
1. Press the multicast paging key when the phone is idle.
The phone sends RTP to a preconfigured multicast address (IP: Port). Any phone in
the local network then listens to the RTP on the preconfigured multicast address (IP:
Port). For both sending and receiving of the multicast RTP , there is no SIP signaling
involved.
The multicast paging key LED illuminates solid green.
The following figure shows a multicast RTP session on the phone:
2. Press the Hold soft key to place the current multicast RTP session on hold.
3. Press the End Call soft key to cancel the multicast RTP session.
Note
Multicast RTP is one way only from the sender to the multicast address(es) (receiver). For
outgoing RTP multicasts, all other existing calls on the phone will be placed on hold.
Advanced Phone Features
icon to
111