Configuration Files

Keep-Alive <keepalive/>

Allowing for the configuration of TCP keep-alive on SIP TLS connections, the phone can detect a failures quickly (in minutes) and attempt to re-register with the SIP call server (or its redundant pair).

This configuration attribute is defined as follows:

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

tcpIpApp.keepalive.tcp.idleTransmitInterval

10 to 7200

Null

After idle x seconds, the

 

 

 

keep-alive message is sent to

 

 

 

the call server.

 

 

 

If set to Null, the default value is

 

 

 

30 seconds.

 

 

 

Note: If this parameter is set to a

 

 

 

value that is out of range, the

 

 

 

default value is used.

 

 

 

 

tcpIpApp.keepalive.tcp.

5 to 120

Null

If no response is received to

noResponseTrasmitInterval

 

 

keep-alive message, another

 

 

 

keep-alive message is sent to

 

 

 

the call server after x seconds.

 

 

 

If set to Null, the default value to

 

 

 

20 seconds.

 

 

 

Note: If this parameter is set to a

 

 

 

value that is out of range, the

 

 

 

default value is used.

 

 

 

 

tcpIpApp.keepalive.tcp.sip.tls.enable

0, 1

0

If set to 1, enable TCP keep-alive

 

 

 

for SIP signalling connections

 

 

 

that use TLS transport.

 

 

 

If set to 0, disable TCP

 

 

 

keep-alive for SIP signalling

 

 

 

connections that use TLS

 

 

 

transport.

 

 

 

 

Web Server <httpd/>

The phone contains a local web server for user and administrator features. This can be disabled for applications where it is not needed or where it poses a security threat. The web server supports both basic and digest authentication. The authentication user name and password are not configurable for this release.

This configuration attribute is defined as follows:

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

httpd.enabled

0, 1

1

If set to 1, the HTTP server will be enabled.

 

 

 

 

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Polycom SIP 3.1 manual Web Server httpd, Value that is out of range, Default value is used

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.