Configuring Your System

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Effective

 

 

 

 

Sample

 

 

audio

Algorithm

MIME Type

Ref.

Bit Rate

Rate

Frame Size

 

bandwidth

Siren14

SIREN14/

SIREN14

24 Kbps

32 Ksps

20ms - 80ms

 

14 KHz

 

16000

 

32 Kbps

 

 

 

 

 

 

 

48 Kbps

 

 

 

 

 

 

 

 

 

 

 

 

Siren22

SIREN22/

SIREN22

32 Kbps

32 Ksps

20ms - 80ms

 

14 KHz

 

48000

 

48 Kbps

 

 

 

 

 

 

 

64 Kbps

 

 

 

 

 

 

 

 

 

 

 

 

RFC 2833

phone-event

RFC 2833

N/A

N/A

N/A

 

N/A

 

 

 

 

 

 

 

 

 

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 consumes 5xkbps of network bandwidth (one-way audio).

 

 

 

Two-way audio would take over 100kbps.

 

 

 

 

 

 

 

 

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-38 and Codec Profiles <audioProfile/> on page A-41.

 

 

 

 

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.

 

 

 

 

 

Background Noise Suppression

Background noise suppression (BNS) is designed primarily for hands-free operation and reduces background noise to enhance communication in noisy environments.

There are no related configuration changes.

Comfort Noise Fill

Comfort noise fill is designed to help provide a consistent noise level to the remote user of a hands-free call. Fluctuations in perceived background noise levels are an undesirable side effect of the non-linear component of most AEC

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Polycom SIP 3.1 manual Background Noise Suppression, Comfort Noise Fill
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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.
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