Administrator’s Guide SoundPoint IP / SoundStation IP

Automatic Gain Control—Designed for hands-free operation, boosts the transmit gain of the local user in certain circumstances.

Background Noise Suppression—Designed primarily for hands-free operation, reduces background noise to enhance communication in noisy environments.

Comfort Noise Fill—Designed to help provide a consistent noise level to the remote user of a hands-free call.

DTMF Event RTP Payload—Conforms to RFC 2833, which describes a standard RTP-compatible technique for conveying DTMF dialing and other telephony events over an RTP media stream.

DTMF Tone Generation—Generates dual tone multi-frequency (DTMF) tones in response to user dialing on the dial pad.

IEEE 802.1p/Q—The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN header.

IP Type-of-Service—Allows for the setting of TOS settings.

Jitter Buffer and Packet Error Concealment—Employs a high-performance jitter buffer and packet error concealment system designed to mitigate packet inter-arrival jitter and out-of-order or lost (lost or excessively delayed by the network) packets.

Low-Delay Audio Packet Transmission—Designed to minimize latency for audio packet transmission.

Voice Activity Detection—Conserves network bandwidth by detecting periods of relative “silence” in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring.

Voice Quality Monitoring—Generates various quality metrics including MOS and R-factor for listening and conversational quality. This feature is part of the Productivity Suite.

Security Features

Local User and Administrator Privilege Levels—Several local settings menus are protected with two privilege levels, user and administrator, each with its own password.

Configuration File Encryption—Confidential information stored in configuration files must be protected (encrypted). The phone can recognize encrypted files, which it downloads from the boot server and it can encrypt files before uploading them to the boot server.

Custom Certificates—When trying to establish a connection to a boot server for application provisioning, the phone trusts certificates issued by widely recognized certificate authorities (CAs).

Incoming Signaling Validation—Levels of security are provided for validating incoming network signaling.

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Polycom SIP 3.1 manual Administrator’s Guide SoundPoint IP / SoundStation IP

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.