Audio Codecs - Polycom SIP 2.2.2 Manual

Sip 2.2.2
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Note

Audio Codecs

Algorithm
MIME Type
G.711μ-law
PMCU
G.711a-law
PCMA
G.722
G722/8000
G.722.1
G722/16000
G.729AB
G729
SID
CN
RFC 2833
phone-event
Central
Configuration file:
sip.cfg
(boot server)
Local
Web Server
(if enabled)
For the SoundPoint IP 501 and 601, utilizing acoustic echo cancellation will
introduce a small delay increase into the audio path which might cause a lower
voice quality.
AEC on the SoundPoint IP 301 handset is not supported.
The following table summarizes the phone's audio codec support:
Ref.
Bit Rate
RFC 1890
64 Kbps
RFC 1890
64 Kbps
RFC 1890
64 Kbps
RFC 3047
16 Kbps,
24 Kbps,
32 Kbps
RFC 1890
8 Kbps
RFC 3389
N/A
RFC 2833
N/A
Configuration changes can performed centrally at the boot server or locally:
Specify codec priority, preferred payload sizes, and jitter buffer tuning
parameters.
For more information, refer to
on page
Specify codec priority, preferred payload sizes, and jitter buffer tuning
parameters.
Navigate to http://<phoneIPAddress>/coreConf.htm#au
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Sample
Rate
Frame Size
8 Ksps
10ms - 80ms
8 Ksps
10ms - 80ms
16 Ksps
10ms - 80ms
16 Ksps
20ms - 80ms
8 Ksps
10ms - 80ms
N/A
N/A
N/A
N/A
Codec Preferences <codecPref/>
A-35
and
Codec Profiles <audioProfile/>
Configuring Your System
Effective
audio
bandwidth
3.5KHz
3.5KHz
7 KHz
7 KHz
3.5KHz
N/A
N/A
on page A-36.
4 - 45

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