Yealink SIP-T41P - Manual (Page 116)

User Guide for the SIP-T41P IP Phone
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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
To send RTP stream:
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.
If G722 codec is used for multicast paging, the LCD screen displays the icon to
indicate that is providing high definition voice.
Default codec for multicast paging is configurable via web user interface only.