Overview

Multiple Call Appearances—The phone supports multiple concurrent calls. The hold feature can be used to pause activity on one call and switch to another call.

Multiple Line Keys per Registration—More than one line key can be allocated to a single.

Multiple Registrations—SoundPoint IP desktop phones support multiple registrations per phone. However, SoundStation IP conference phones support a single registration.

Network Address Translation—The phones can work with certain types of network address translation (NAT).

Presence—Allows the phone to monitor the status of other users/devices and allows other users to monitor it. Requires call server support.

Real-Time Transport Protocol Ports—The phone treats all real- time transport protocol (RTP) streams as bi-directional from a control perspective and expects that both RTP end points will negotiate the respective destination IP addresses and ports.

Recording and Playback of Audio Calls — Recording and playback allows the user to record any active conversation using the phone on a USB device. The files are date and time stamped for easy archiving and can be played back on the phone or on any computer with a media playback program what supports the .wav format. This feature is part of the Productivity Suite.

Server Redundancy—Server redundancy is often required in VoIP deployments to ensure continuity of phone service for events where the call server needs to be taken offline for maintenance, the server fails, or the connection from the phone to the server fails.

Shared Call Appearances—Calls and lines on multiple phones can be logically related to each other. Requires call server support.

Static DNS Cache—Set up a static DNS cache and provide for negative caching.

Synthesized Call Progress Tones—In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment, call progress tones are synthesized during the life cycle of a call. Customizable for certain regions, for example, Europe has different tones from North America.

Voice Mail Integration—Compatible with voice mail servers.

Audio Features

Acoustic Echo Cancellation—Employs advanced acoustic echo cancellation for hands-free operation.

Audio Codecs—Supports the standard audio codecs.

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Polycom SIP 3.1 manual Overview

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.