Configuring Your System

appropriate URL upon arrival of the appropriate SIP signaling, subject to some conditions described below.

Since new web content URLs can be received at any time—as the first URL for a call or a replacement URL—rules are needed to match displayed web content with automatic phone behaviour, which are valid actions from within the Microbrowser context.

Spontaneous web content will only be retrieved and displayed for a call if that call occupies, or will occupy, the UI focus at the time of the event.

Passive Mode. Web content can also be retrieved when the user chooses to do so. The fact that web content is available for viewing is shown through the call appearance-based web content icon descibed in Web Content Status Indication on page 4-66. The Select key can be used to fetch the associated web content for the call that is in focus. If the web content has expired, the icon will be removed and the Select key will perform no function.

Passive mode is recommended for applications where the Microbrowser is used for other applications. In the SIP 2.2 feature, interactive microbrowser sessions will be interrupted by the arrival of active-mode web content URLs, which may cause annoyance, although the Back navigation function will work in this context.

Settings Menu

If enabled, a new SIP web content entry is added to the Setting > Basic > Preferences menu to allow the user to change the current content retrieval mode. Two options are provided: passive mode and active mode.

Signaling Changes

A new SIP header must be used to report web content associated with SIP phone calls (the SSAWC header follow the BNF for the standard SIP header Alert-Info):

Alert-Info = "Alert-Info" HCOLON alert-param *(COMMA alert-param) alert-param = LAQUOT absoluteURI RAQUOT *( SEMI generic-param )

The web content must be located with an absolute URI, which begins with the scheme identifier. Currently only the HTTP scheme is supported.

So an example header might look like:

Access-URL: <http://server.polycom.com/content23456.xhtml>

This header may be placed in SIP requests and responses, as appropriate so long as the messages are part of an INVITE-initiated dialog and the phone can associate them with an existing phone call.

This feature also requires the definition of two optional parameters:

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Polycom SIP 3.1 manual Settings Menu

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.