Administrator’s Guide SoundPoint IP / SoundStation IP

The syslog protocol is a very simplistic protocol: the syslog sender sends a small textual message (less than 1024 bytes) to the syslog receiver. The receiver is commonly called “syslogd”, “syslog daemon” or “syslog server”. Syslog messages can be sent through UDP, TCP, or TLS. The data is sent in cleartext.

Syslog is supported by a wide variety of devices and receivers. Because of this, syslog can be used to integrate log data from many different types of systems into a central repository.

The syslog protocol is defined in RFC 3164. For more information on syslog, go to http://www.ietf.org/rfc/rfc3164.txt?number=3164 .

The following syslog configuration parameters can be modified on the Syslog menu:

Name

Possible Values

Description

 

 

 

Server Address

dotted-decimal IP address

The syslog server IP address or host name.

 

OR

The default value is NULL.

 

domain name string

 

 

 

 

 

Server Type

None=0,

The protocol that the phone will use to write to the syslog

 

UDP=1,

server.

 

TCP=2,

If set to “None”, transmission is turned off, but the server

 

TLS=3

 

address is preserved.

 

 

 

 

 

Facility

0 to 23

A description of what generated the log message. For

 

 

more information, refer to section 4.1.1 of RFC 3164.

 

 

The default value is 16, which maps to “local 0”.

 

 

 

Render Level

0 to 6

Specifies the lowest class of event that will be rendered to

 

 

syslog. It is based on log.render.level and can be a

 

 

lower value.

 

 

Refer to Basic Logging <level/><change/> and <render/>

 

 

on page A-86.

 

 

Note: Use left and right arrow keys to change values.

 

 

 

Prepend MAC

Enabled, Disabled

If enabled, the phone’s MAC address is prepended to the

Address

 

log message sent to the syslog server.

 

 

 

Setting Up the Boot Server

The boot server can be on the local LAN or anywhere on the Internet.

Multiple boot servers can be configured by having the boot server DNS name map to multiple IP addresses. The default number of boot servers is one and the maximum number is eight. The following protocols are supported for redundant boot servers: HTTPS, HTTP, and FTP. For more information on the protocol used on each platform, refer to Supported Provisioning Protocols on page 3-4.

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Polycom SIP 3.1 manual Setting Up the Boot Server, Refer to Basic Logging level/change/ and render

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.