Administrator’s Guide SoundPoint IP / SoundStation IP

Application Configuration Files

Typically, the files are arranged in the following manner although parameters may be moved around within the files and the filenames themselves can be changed as needed. These files dictate the behavior of the phone once it is running the executable specified in the master configuration file.

The application files are:

Application—It contains parameters that affect the basic operation of the phone such as voice codecs, gains, and tones and the IP address of an application server. All phones in an installation usually share this category of files. Polycom recommends that you create another file with your organization’s modifications. If you must change any Polycom templates, back them up first. By default, sip.cfg is included.

Per-phone—It contains parameters unique to a particular phone user. Typical parameters include:

display name

unique addresses

Each phone in an installation usually has its own customized version of user files derived from Polycom templates. By default, phone1.cfg is included.

Central Provisioning

The phones can be centrally provisioned from a boot server through a system of global and per-phone configuration files. The boot server also facilitates automated application upgrades, logging, and a measure of fault tolerance. Multiple redundant boot servers can be configured to improve reliability.

In the central provisioning method, there are two major classifications of configuration files:

System configuration files

Per-phone configuration files

Parameters can be stored in the files in any order and can be placed in any number of files. The default is to have 2 files, one for per-phone setting and one for system settings. The per-phone file is typically loaded first, and could contain system level parameters, letting you override that parameter for a given user. For example, it might be desirable to set the default CODEC for a remote user differently than for all the users who reside in the head office. By adding the CODEC settings to a particular user’s per-phone file, the values in the system file are ignored.

Note

Verify the order of the configuration files. Parameters in the configuration file loaded

 

first will overwrite those in later configuration files.

 

 

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Polycom SIP 3.1 manual Application Configuration Files

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.