Polycom SIP 2.2.0 manual Audio Codecs, Following table summarizes the phone’s audio codec support

Configuring Your System

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.

Note AEC on the SoundPoint IP 301 handset is not supported.

Audio Codecs

The following table summarizes the phone’s audio codec support:

						Effective
				Sample		audio
Algorithm	MIME Type	Ref.	Bit Rate	Rate	Frame Size	bandwidth
G.711μ-law	PMCU	RFC 1890	64 Kbps	8 Ksps	10ms - 80ms	3.5KHz
G.711a-law	PCMA	RFC 1890	64 Kbps	8 Ksps	10ms - 80ms	3.5KHz
G.722	G722/8000	RFC 1890	64 Kbps	16 Ksps	10ms - 80ms	7 KHz
G.722.1	G722/16000	RFC 3047	16 Kbps,	16 Ksps	20ms - 80ms	7 KHz
			24 Kbps,
			32 Kbps
G.729AB	G729	RFC 1890	8 Kbps	8 Ksps	10ms - 80ms	3.5KHz
SID	CN	RFC 3389	N/A	N/A	N/A	N/A
RFC 2833	phone-event	RFC 2833	N/A	N/A	N/A	N/A

Configuration changes can performed centrally at the boot server or locally:

Central	Configuration file:	Specify codec priority, preferred payload sizes, and jitter buffer tuning
(boot server)	sip.cfg	parameters.
		• For more information, refer to Codec Preferences <codecPref/>
		on page A-35and Codec Profiles <audioProfile/> on page A-36.
Local	Web Server	Specify codec priority, preferred payload sizes, and jitter buffer tuning
	(if enabled)	parameters.
		Navigate to http://<phoneIPAddress>/coreConf.htm#au
		Changes are saved to local flash and backed up to <Ethernet
		address>-phone.cfgon the boot server. Changes will permanently
		override global settings unless deleted through the Reset Local
		Config menu selection and the <Ethernet address>-phone.cfgis
		removed from the boot server.

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