Configuration Files

User Preferences <user_preferences/>

Registration <reg/>

SoundPoint IP 301, 320, 330, and 430 support a maximum of two unique registrations, SoundPoint IP 501 supports three, the SoundPoint IP 550 and 560 supports four, and SoundPoint IP 600, 601, 650, and 670 support six. Up to three SoundPoint IP Expansion Modules can be added to a single host SoundPoint IP 601 and 650 phone increasing the total number of buttons to 12 registrations on the IP 601 and 34 registrations on the IP 650. Each registration can optionally be associated with a private array of servers for completely segregated signaling. The SoundStation IP 4000, 6000, and 7000 supports a single registration.

In the following table, x is the registration number. IP 301, 320, 330, 430: x=1-2;

IP 501: x=1-3; IP 550, 560: x=1-4; IP 600: x=1-6; IP 601: x=1-12;

IP 650, 670: x=1-34; IP 4000: x=1; IP 6000: x=1; IP 7000: x=1.

Attribute

Permitted

Default

Interpretation

Values

 

 

 

 

reg.x.csta

0, 1

Null

If set to 1, uaCSTA is enabled.

 

 

 

If reg.x.csta is not Null, this attribute

 

 

 

overrides the global CSTA flag in the sip.cfg

 

 

 

configuration file.

 

 

 

 

reg.x.displayName

UTF-8 encoded

Null

Display name used for local user interface as

 

string

 

well as SIP signaling.

 

 

 

 

reg.x.address

string in the format

Null

The user part or the user and the host part of

 

userPart from

 

the phone’s SIP URI.

 

userPart@domain

 

The user part of the phone's SIP URI. For

 

 

 

example, reg.x.address=”1002” from

 

 

 

1002@polycom.com or

 

 

 

reg.x.address=”1002@polycom.com”.

 

 

 

 

reg.x.label

UTF-8 encoded

Null

Text label to appear on the display adjacent

 

string

 

to the associated line key. If omitted, the label

 

 

 

will be derived from the user part of

 

 

 

reg.x.address.

 

 

 

 

reg.x.lcs

0, 1

0

If set to 1, the Microsoft Live Communications

 

 

 

Server is supported for registration x.

 

 

 

 

reg.x.type

private OR shared

private

If set to private, use standard call signaling.

 

 

 

If set to shared, augment call signaling with

 

 

 

call state subscriptions and notifications and

 

 

 

use access control for outgoing calls.

 

 

 

 

reg.x.thirdPartyName

string in the same

Null

This field must match the reg.x.address

 

format as

 

value of the other registration which makes

 

reg.x.address

 

up the bridged line appearance (BLA). It must

 

 

 

be Null in all other cases.

 

 

 

 

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Polycom SIP 3.1 manual Registration reg, User Preferences userpreferences

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.