Administrator’s Guide SoundPoint IP / SoundStation IP

A wide range of performance metrics are generated. Some are based on current values, such as jitter buffer nominal delay and round trip delay, while others cover the time period from the beginning of the call until the report is sent, such as network packet loss. Some metrics are computed using other metrics as input, such as listening Mean Opinion Score (MOS), conversational MOS, listening R-factor, and conversational R-factor.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Specify the location of the central report collector, how often the

(boot server)

sip.cfg

reports are generated, and the warning and critical threshold values

 

 

that will cause generation of alert reports.

 

 

For more information, refer to Quality Monitoring <quality

 

 

 

monitoring/> on page A-52.

 

 

 

 

Dynamic Noise Reduction

Dynamic noise reduction (DNR) provides maximum microphone sensitivity, while automatically reducing background noise— from fans, projectors, heating and air conditioning—for clearer sound and more efficient conferencing.

There are no related configuration changes.

Treble/Bass Controls

The treble and bass controls equalize the tone of the high and low frequency sound from the speakers.

The SoundStation IP 7000 phone’s treble and bass controls can be modified by the user (through Menu > Settings > Basic > Audio > Treble EQ or Bass EQ).

Configuration changes can performed centrally at the boot server:

Central (boot server)

Configuration file: sip.cfg

Specify the user’s preferences for treble and bass.

For more information, refer to User Preferences <up/> on page A-25.

Setting Up Security Features

This section provides information for making configuration changes for the following security-related features:

Local User and Administrator Privilege Levels

Custom Certificates

Incoming Signaling Validation

4 - 80

Page 134
Image 134
Polycom SIP 3.1 Setting Up Security Features, Dynamic Noise Reduction, Treble/Bass Controls, Monitoring/ on page A-52

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.