Bridged Line Appearance

		Configuring Your System

	Configuration changes can performed centrally at the boot server or locally:

Central	Configuration file:	Specify whether diversion should be disabled on shared lines.
(boot server)	sip.cfg	• For more information, refer to Shared Calls <shared/> on page
		A-67.
		Specify line-seize subscription period.
		• For more information, refer to Server <server/> on page A-7.
		Specify standard or non-standard behavior for processing line-seize
		subscription for mutual exclusion feature.
		• For more information, refer to Special Events <specialEvent/> on
		page A-16.

	Configuration file:	Specify per-registration line type (private or shared), barge-in
	phone1.cfg	capabilities, and line-seize subscription period if using per-registration
		servers. A shared line will subscribe to a server providing call state
		information.
		• For more information, refer to Registration <reg/> on page A-107.
		Specify per-registration whether diversion should be disabled on
		shared lines.
		• For more information, refer to Diversion <divert/> on page A-114.

Local	Web Server	Specify line-seize subscription period.
	(if enabled)	Navigate to http://<phoneIPAddress>/appConf.htm#se
		Specify standard or non-standard behavior for processing line-seize
		subscription for mutual exclusion feature.
		Navigate to http://<phoneIPAddress>/appConf.htm#ls
		Specify per-registration line type (private or shared) and line-seize
		subscription period if using per-registration servers, and whether
		diversion should be disabled on shared lines.
		Navigate to http://<phoneIPAddress>/reg.htm
		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.

	Local Phone User	Specify per-registration line type (private or shared) using the SIP
	Interface	Configuration menu. 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.

Calls and lines on multiple phones can be logically related to each other. A call that is active on one phone will be presented visually to phones that share that line. Mutual exclusion features emulate traditional PBX or key system privacy for shared calls. Incoming calls can be presented to multiple phones

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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.

Polycom SIP 3.1 manual Bridged Line Appearance

Models: SIP 3.1

phone1.cfg

Bridged Line Appearance

SIP 3.1 specifications