Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server:

Central (boot

Configuration File:

Specify fonts to overwrite existing ones or specify new fonts.

server)

sip.cfg

For more information, refer to Fonts <font/> on page A-72.

 

 

Specify which bitmaps to use.

 

 

For more information, refer to Bitmaps <bitmap/>on page A-80.

 

 

Specify how to create animations and LED indicator patterns.

 

 

For more information, refer to Indicators <ind/> on page A-80.

 

 

 

Instant Messaging

The phone supports sending and receiving instant text messages. The user is alerted to incoming messages visually and audibly. The user can view the messages immediately or when it is convenient. For sending messages, the user can either select a message from a preset list of short messages or an alphanumeric text entry mode allows the typing of custom messages using the dial pad. Message sending can be initiated by replying to an incoming message or by initiating a new dialog. The destination for new dialog messages can be entered manually or selected from the contact directory, the preferred method.

Configuration changes can be performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

Multilingual User Interface

Note

This feature is not available on SoundPoint IP 301 phones.

 

 

The system administrator or the user can select the language. Support for major western European languages is included and additional languages can be easily added. Support for Asian languages (Chinese, Japanese, and Korean) is also included, but will display only on the SoundPoint IP 550, 560, 650, and 670 and SoundStation IP 4000, 6000, and 7000’s higher resolution displays. A WGL4 character set is displayed the SoundStation IP 7000. For more information, refer to http://www.microsoft.com/OpenType/otspec/WGL4E.HTM.

For basic character support and extended character support (available on SoundPoint IP 550, 560, 600, 601, and 650 and 670 and SoundStation IP platforms), refer to Multilingual <ml/> on page A-22. (Note that within a Unicode range, some characters may not be supported due to their infrequent usage.)

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Polycom SIP 3.1 manual Instant Messaging, Multilingual User Interface, Central boot, Server Sip.cfg

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.