Packetization Time (Ptime); Supported Ptime Of Audio Codec - Yealink SIP-T2 Series Administrator's Manual

Administrator's Guide for SIP-T2 Series/T4 Series/T5 Series/CP920 IP Phones
When the audio codec is iLBC_13_33kpbs, the default value is 0;
Web UI Account->Codec->Audio Codec
Parameter account.X.codec.opus.para
It configures the sample rate of the Opus audio codec.
Description
Note: It is only applicable to T54S/T52S/T48S/T46S/T42S/T41S/T27G/CP920 IP phones.
opus-fb-Opus-FB (48KHz) (only for CP920)
opus-swb-Opus-SWB (24KHz) (only for CP920)
Permitted
opus-wb-Opus-WB (16KHz)
Values
opus-mb-Opus-MB (12KHz) (only for CP920)
opus-nb-Opus-NB (8KHz)
Default opus-wb (opus-fb for CP920 IP phones)
Web UI Account->Codec->Opus Sample Rate
Parameter voice.g726.aal2.enable
Description It enables or disables the IP phone to use the AAL2-G726-16, 24, 32 and 40 MIME type.
Permitted
0-Disabled
1-Enabled
Values
Default 0
[1]
X is the account ID. For T54S/T48S/T48G/T46S/T46G/T29G, X=1-16; for T52S/T42G/T42S, X=1-12; for T41P/T41S/T27G,
X=1-6; for T40P/T40G/ T23P/T23G, X=1-3; for T21(P) E2, X=1-2; for T19(P) E2/CP920, X=1.

Packetization Time (PTime)

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.
Topics

Supported PTime of Audio Codec

PTime Configuration
Supported PTime of Audio Codec
The following table summarizes the valid values of ptime for each audio codec:
Codec Packetization Time (Minimum)
G722
PCMA
PCMU
354
[1]
10ms
10ms
10ms
<MAC>.cfg
<y0000000000xx>.cfg
Packetization Time (Maximum)
40ms
40ms
40ms