Administrator’s Guide SoundPoint IP / SoundStation IP

When using the handset on any SoundPoint IP phones, AEC is not normally required. In certain situations, where echo is experienced by the far-end party, when the user is on the handset, AEC may be enabled to reduce/avoid this echo. To achieve this, make the following changes in the sip.cfg configuration file (default settings for these parameters are disabled):

voice.aec.hs.enable = 1 voice.aes.hs.enable = 1 voice.ns.hs.enable = 1 voice.ns.hs.signalAttn = -6 voice.ns.hs.silenceAttn = -9

For more information, refer to Acoustic Echo Cancellation <aec/> on page A-37, Acoustic Echo Suppression <aes/> on page A-46, and Background Noise Suppression <ns/> on page A-47.

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

G7221/16000

RFC 3047

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

RFC 1890

8 Kbps

8 Ksps

10ms - 80ms

3.5KHz

 

 

 

 

 

 

 

SID

CN

RFC 3389

N/A

N/A

N/A

N/A

 

 

 

 

 

 

 

Lin16

L16/16000

RFC 1890

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

 

 

 

 

 

 

 

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Polycom SIP 3.1 manual Audio Codecs, Following table summarizes the phone’s audio codec support, Effective

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.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

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.