Polycom SIP 3.1 manual Codec Profiles audioProfile, Permitted Attribute Values Interpretation

Configuration Files

Codec Profiles <audioProfile/>

The following profile attributes can be adjusted for each of the five supported codecs. In the table, x=G711Mu, G711A, G722, G7221, G7221C, and G729AB, Lin16, Siren14, and Siren22.

	Permitted
Attribute	Values	Interpretation
voice.audioProfile.x.payloadSize	10, 20, 30, ...80	Preferred Tx payload size in milliseconds to be
		provided in SDP offers and used in the
		absence of ptime negotiations. This is also the
		range of supported Rx payload sizes.
		The payload size for G7221, G7221C, Siren14,
		and Siren22 are further subdivided.
voice.audioProfile.x.jitterBufferMin	20, 40, 50, 60,	The smallest jitter buffer depth (in milliseconds)
	... (multiple of	that must be achieved before play out begins
	10)	for the first time. Once this depth has been
		achieved initially, the depth may fall below this
		point and play out will still continue. This
		parameter should be set to the smallest
		possible value which is at least two packet
		payloads, and larger than the expected short
		term average jitter. The IP4000 values are the
		same as the IP30x values.
voice.audioProfile.x.jitterBufferShrink	10, 20, 30, ...	The absolute minimum duration time (in
	(multiple of 10)	milliseconds) of RTP packet Rx with no packet
		loss between jitter buffer size shrinks. Use
		smaller values (1000 ms) to minimize the delay
		on known good networks. Use larger values to
		minimize packet loss on networks with large
		jitter (3000 ms).
voice.audioProfile.x.jitterBufferMax	>	The largest jitter buffer depth to be supported
	jitterBufferMin,	(in milliseconds). Jitter above this size will
	multiple of 10,	always cause lost packets. This parameter
	<=300 for IP	should be set to the smallest possible value
	320, 330, 430,	that will support the expected network jitter.
	501,550, 600,
	601, and 650
	<= 200 for IP
	301

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