Supported Audio Codec Specifications

The following table summarizes the specifications for audio codecs supported on Polycom phones.

Table 1. Audio Codec Specifications

Algorithm

Reference

Raw Bit Rate

Maximum IP Bit Rate

Sample Rate

Default Payload Size

Effective Audio Bandwidth

G.711 ยต -law

RFC 1890

64 Kbps

80 Kbps

8 Ksps

20 ms

3.5 KHz

G.711 a-law

RFC 1890

64 Kbps

80 Kbps

8 Ksps

20 ms

3.5 KHz

G.719

RFC 5404

32 Kbps

48 Kbps

64 Kbps

48 Kbps

64 Kbps

80 Kbps

48 Ksps

20 ms

20 KHz

G.711

RFC 1890

64 Kbps

80 Kbps

16 Ksps

20 ms

7 KHz

G.7221

RFC 3551

64 Kbps

80 Kbps

16 Ksps

20 ms

7 KHz

G.722.1

RFC 3047

24 Kbps

32 Kbps

40 Kbps

48 Kbps

16 Ksps

20 ms

7 KHz

G.722.1C

G7221C

24 Kbps

32 Kbps

48 Kbps

40 Kbps

48 Kbps

64 Kbps

32 Ksps

20 ms

14 KHz

G.729AB

RFC 1890

8 Kbps

24 Kbps

8 Ksps

20 ms

3.5 KHz

Opus

RFC 6716

8 - 24 Kbps

24 - 40 Kbps

8 Ksps

16 Ksps

20 ms

3.5 KHz

7 KHz

Lin16

RFC 1890

128 Kbps

256 Kbps

512 Kbps

705.6 Kbps

768 Kbps

132 Kbps

260 Kbps

516 Kbps

709.6 Kbps

772 Kbps

8 Ksps

16 Ksps

32 Ksps

44.1 Ksps

48 Ksps

10 ms

3.5 KHz

7 KHz

14 KHz

20 KHz

22 KHz

Siren 7

SIREN7

16 Kbps

24 Kbps

32 Kbps

32 Kbps

40 Kbps

48 Kbps

16 Ksps

20 ms

7 KHz

Siren14

SIREN14

24 Kbps

32 Kbps

48 Kbps

40 Kbps

48 Kbps

64 Kbps

32 Ksps

20 ms

14 KHz

Siren22

SIREN22

32 Kbps

48 Kbps

64 Kbps

48 Kbps

64 Kbps

80 Kbps

48 Ksps

20 ms

22 KHz

iLBC

RFC 3951

13.33 Kbps

15.2 Kbps

31.2 Kbps

24 Kbps

8 Ksps

30 ms

20 ms

3.5 KHz

SILK

SILK

Skype SILK

6 - 20 Kbps

7 - 25 Kbps

8 - 30 Kbps

12 - 40 Kbps

36 Kbps

41 Kbps

46 Kbps

56 Kbps

8 Ksps

12 Ksps

16 Ksps

24 Ksps

3.5 KHz

5.2 KHz

7 KHz

11 KHz

1 Per RFC 3551. Even though the actual sampling rate for G.722 audio is 16,000 Hz (16ksps), the RTP clock rate advertised for the G.722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility.

Note: The network bandwidth necessary to send the encoded voice is typically 5-10% higher than the encoded bit rate due to packetization overhead. For example, a G.722.1C call at 48kbps for both the receive and transmit signals consumes about 100kbps of network bandwidth (two-way audio).