Administrator’s Guide SoundPoint IP / SoundStation IP

Voice Coding Algorithms <codecs/>

The following voice codecs are supported:

 

 

 

 

Sample

 

Effective Audio

Algorithm

MIME Type

Label

Bit Rate

Rate

Frame Size

Bandwidth

 

 

 

 

 

 

 

G.711μ-law

PMCU

G711mu

64 Kbps

8 Ksps

10ms - 80ms

3.5 KHz

 

 

 

 

 

 

 

G.711a-law

PCMA

G711A

64 Kbps

8 Ksps

10ms - 80ms

3.5 KHz

 

 

 

 

 

 

 

G.722

G722/8000

G722

64 Kbps

16 Ksps

10ms - 80ms

7 KHz

 

 

 

 

 

 

 

G.722.1

G7221/16000

G7221

16 Kbps,

16 Ksps

20ms - 80ms

7 KHz

 

 

 

24 Kbps,

 

 

 

 

 

 

32 Kbps

 

 

 

 

 

 

 

 

 

 

G.722.1C

G7221/

G7221C

24 Kbps

32 Ksps

20ms - 80ms

14 KHz

 

32000

 

32 Kbps

 

 

 

 

 

 

48 Kbps

 

 

 

 

 

 

 

 

 

 

G.729AB

G729

G729AB

8 Kbps

8 Ksps

10ms - 80ms

3.5 KHz

 

 

 

 

 

 

 

Lin16

L16/16000

L16

25.6 Kbps

16 Ksps

10ms

7 KHz

 

L16/32000

 

51.2 Kbps

32 Ksps

 

14 KHz

 

L16/48000

 

76.8 Kbps

48 Ksps

 

22 KHz

 

 

 

 

 

 

 

Siren14

SIREN14/

SIREN14

24 Kbps

32 Ksps

20ms - 80ms

14 KHz

 

16000

 

32 Kbps

 

 

 

 

 

 

48 Kbps

 

 

 

 

 

 

 

 

 

 

Siren22

SIREN22/

SIREN22

32 Kbps

48 Ksps

20ms - 80ms

22 KHz

 

48000

 

48 Kbps

 

 

 

 

 

 

64 Kbps

 

 

 

 

 

 

 

 

 

 

These codecs include:

Codec Preferences <codecPref/>

Codec Profiles <audioProfile/>

Codec Preferences <codecPref/>

Attribute

Permitted

Default

Interpretation

Values

 

 

 

 

voice.codecPref.G711Mu

Null, 1-3

1

Specifies the codec preferences for

 

 

 

SoundPoint IP 320, 330, 430, 500, 501,

voice.codecPref.G711A

 

2

 

600 and 601 platforms.

 

 

 

voice.codecPref.G729AB

 

3

1 = highest

 

 

 

3 = lowest

 

 

 

Null = do not use

 

 

 

Give each codec a unique priority, this will

 

 

 

dictate the order used in SDP negotiations.

 

 

 

 

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Polycom SIP 3.1 manual Following voice codecs are supported, These codecs include, Codec Preferences codecPref

SIP 3.1 specifications

Polycom SIP 3.1 is an advanced session initiation protocol designed to enhance voice and video communication in various business environments. As a pivotal component of Polycom’s telecommunication solutions, SIP 3.1 offers several features and characteristics that cater to the evolving needs of modern enterprises, particularly those that rely on seamless and efficient communication.

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Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.