Contents

Instant Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30 Multilingual User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30 Downloadable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–31 Synthesized Call Progress Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32 Microbrowser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32 Real-Time Transport Protocol Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 4–33 Network Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34 Corporate Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34 Recording and Playback of Audio Calls . . . . . . . . . . . . . . . . . . . . . . . 4–37 Daisy-Chaining Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–38 Provisioning Phones Over CLink . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–39 Enhanced Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–40 Configurable Soft Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–50 Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–54 Multiple Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–55 Automatic Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–56 Server Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–56 Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–60 Microsoft Live Communications Server 2005 Integration . . . . . . . . 4–61 Access URL in SIP Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–65 Static DNS Cache . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–68 Display of Warnings from SIP Headers . . . . . . . . . . . . . . . . . . . . . . . 4–72

Setting Up Audio Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–73 Low-Delay Audio Packet Transmission . . . . . . . . . . . . . . . . . . . . . . . 4–74 Jitter Buffer and Packet Error Concealment . . . . . . . . . . . . . . . . . . . . 4–74 Voice Activity Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–74 DTMF Tone Generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75 DTMF Event RTP Payload . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75 Acoustic Echo Cancellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75 Audio Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–76 Background Noise Suppression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77 Comfort Noise Fill . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77 Automatic Gain Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78 IP Type-of-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78 IEEE 802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78 Voice Quality Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–79 Dynamic Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80 Treble/Bass Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80

Setting Up Security Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80 Local User and Administrator Privilege Levels . . . . . . . . . . . . . . . . . 4–81

vii

Page 6 Page 8
Page 7
Image 7
Polycom SIP 3.1 manual Contents
Page 6 Page 8

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.
Top Page Image Contents