Administrator’s Guide for SIP-T2 Series/T19(P) E2/T4 Series IP Phones
818
Note: It is only applicable to the IP phones running firmware version 81 or prior.
Web User Interface:
None
Phone User Interface:
None
The configuration of audio codecs via web user interface for old configuration behavior is
the same as the newer one. For more information, refer to the introduction in the section
New Configuration Behavior.
Ptime is a measurement of the duration (in milliseconds) of the audio data in each RTP packet
sent to the destination, and defines how much network bandwidth is used for the RTP stream
transfer. Before establishing a conversation, codec and ptime are negotiated through SIP
signaling. The valid values of ptime range from 10 to 60, in increments of 10 milliseconds. The
default ptime is 20ms. You can also disable the ptime negotiation.
The following table summarizes the valid values of ptime for each audio codec:
Packetization Time
(Minimun)
Packetization Time (Maximun)
30ms
(40ms for
SIP-T40P/T40G/T23P/T23G/T21(P)
E2/T19(P) E2)