Configuring Your System

For example, if a user parks a call, the following message appears on their phone:

Configuration changes can be performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off and specify which warnings are

(boot server)

sip.cfg

displayable.

 

 

• For more information, refer to SIP <SIP/> on page A-10.

 

 

 

Setting Up Audio Features

Proprietary state-of-the-art digital signal processing (DSP) technology is used to provide an excellent audio experience.

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

•Low-Delay Audio Packet Transmission

•Jitter Buffer and Packet Error Concealment

•Voice Activity Detection

•DTMF Tone Generation

•DTMF Event RTP Payload

•Acoustic Echo Cancellation

•Audio Codecs

•Background Noise Suppression

•Comfort Noise Fill

•Automatic Gain Control

•IP Type-of-Service

•IEEE 802.1p/Q

•Voice Quality Monitoring

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Polycom SIP 3.1 manual Setting Up Audio Features, Central
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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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