Rtp Base Port; Codec Preferences - Siemens A31003-D3000-M100-01-76A9 Administrator User Manual

Administration
Speech

3.16.1 RTP Base Port

3.16.2 Codec Preferences

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The port used for RTP is negotiated during the establishment of a SIP connection.
The RTP base port number defines the starting point from which the phone will
count up when negotiating. The default value is 5010.
The number of the port used for RTCP will be the RTP port number increased by
1.
Administration via WBM
Network > Port Configuration
Administration via Local Phone
|
--- Admin
|--- Network
|--- Port Configuration
|--- RTP base
Related Topics
If Silence suppression is activated, the transmission of data packets is
suppressed on no conversation, that is, if the user doesn't speak.
The OpenScape Desk Phone provides the codecs G.711, G.722, and G.729.
When a SIP connection is established between two endpoints, the phones
negotiate the codec to be used. The result of the negotiation is based on the
general availability and ranking assigned to each codec. The administrator can
allow or disallow a codec as well as assign a ranking number to it.
The Packet size, i. e. length in milliseconds, of the RTP packets for speech data,
can be set to 10ms, 20ms, 30ms, 60ms or to automatic detection.
OpenScape Desk Phone IP, Phone Administration, Administrator Documentation
A31003-D3000-M100-01-76A9, 02-2013