Administrator’s Guide SoundPoint IP / SoundStation IP

Local

Local Phone User

Use the SIP Configuration menu to specify the local SIP signaling

(continued)

Interface

port, a default SIP server to register to and registration information for

 

 

up to twelve registrations (depending on the phone model). The SIP

 

 

Configuration menu contains a sub-set of all the parameters available

 

 

in the configuration files.

 

 

Either the Web Server or the boot server configuration files or the

 

 

local phone user interface should be used to configure registrations,

 

 

not a mixture of these options. When the SIP Configuration menu is

 

 

used, it is assumed that all registrations use the same server.

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfgon the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfgis

 

 

removed from the boot server.

 

 

For more information, refer to Local <local/> on page A-6, Server

 

 

<server/> on page A-7, and Registration <reg/> on page A-107.

 

 

 

Automatic Call Distribution

The phone allows automatic call distribution (ACD) login and logout. This feature depends on support from a SIP server.

Configuration changes can 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.

 

 

 

 

Configuration file:

Enable this feature per registration.

 

phone1.cfg

For more information, refer to Registration <reg/> on page A-107.

 

 

 

 

The phone also supports ACD agent available and unavailable. This feature depends on support from a SIP server.

Configuration changes can 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.

 

 

 

 

Configuration file:

Enable this feature per registration.

 

phone1.cfg

For more information, refer to Registration <reg/> on page A-107.

 

 

 

 

Server Redundancy

Server redundancy is often required in VoIP deployments to ensure continuity of phone service for events where the call server needs to be taken offline for maintenance, the server fails, or the connection from the phone to the server fails.

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Polycom SIP 3.1 Automatic Call Distribution, Server Redundancy, Server/ on page A-7, and Registration reg/ on page A-107

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.