Configuration Files

 

 

 

 

 

 

 

 

Server <server/>

 

 

 

 

 

This configuration attribute is defined as follows:

 

 

 

 

 

 

 

Attribute

 

Permitted Values

Default

Interpretation

 

 

 

 

 

 

dialplan.routing.server.x.

 

dotted-decimal IP address

Null

IP address or host name and port of

address

 

or host name

 

a SIP server that will be used for

 

 

 

 

routing calls. Multiple servers can

dialplan.routing.server.x.port

 

1 to 65535

5060

 

be listed starting with x=1, 2, ... for

 

 

 

 

fault tolerance.

 

 

 

 

 

 

 

 

Emergency <emergency/>

 

 

 

 

 

In the following attributes, x is the index of the emergency entry description

 

and y is the index of the server associated with emergency entry x. For each

 

emergency entry (index x), one or more server entries (indexes (x,y)) can be

 

configured. x and y must both use sequential numbering starting at 1.

 

 

 

 

 

 

 

Attribute

 

Permitted Values

Default

 

 

Interpretation

 

 

 

 

 

 

 

dialplan.routing.emergency.x.

 

Single entry representing

for x =1,

 

 

This determines the URLs

value

 

a SIP URL

value = “911”, Null

 

that should be watched for.

 

 

 

for all others

 

 

When one of these defined

 

 

 

 

 

 

URLs is detected as having

 

 

 

 

 

 

been dialed by the user, the

 

 

 

 

 

 

call will automatically be

 

 

 

 

 

 

directed to the defined

 

 

 

 

 

 

emergency server.

 

 

 

 

 

 

dialplan.routing.emergency.x.

 

positive integer

for x=1, y =1, Null

 

Index representing the

server.y

 

 

for all others

 

 

server defined in Server

 

 

 

 

 

 

<server/> on page A-21 that

 

 

 

 

 

 

will be used for emergency

 

 

 

 

 

 

routing.

 

 

 

 

 

 

 

Localization <lcl/>

The phone has a multilingual user interface. It supports both North American and international time and date formats. The call progress tones can also be customized. For more information, refer to Chord-Sets <chord/> on page A-29, and Call Progress Patterns on page A-33.

This attribute includes:

•Multilingual <ml/>

•Date and Time <datetime/>

A - 21

Page 174 Page 176
Page 175
Image 175
Polycom SIP 3.1 manual Localization lcl, Server server, Emergency emergency, Multilingual ml Date and Time datetime
Page 174 Page 176

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