Voice Quality Monitoring, Threshold | Polycom SIP 3.1 specs

		Configuring Your System

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

Central	Configuration file:	Specify default and protocol-specific 802.1p/Q settings.
(boot server)	sip.cfg	• For more information, refer to Ethernet IEEE 802.1p/Q
		<ethernet/> on page A-55.

Local	Web Server	Specify 802.1p/Q settings.
	(if enabled)	Navigate to http://<phoneIPAddress>/netConf.htm#qo

	Local Phone User	Specify whether CDP is to be used or manually set the VLAN ID or
	Interface	configure DHCP VLAN Discovery.
		Phase 1: bootRom - Navigate to: SETUP menu during auto-boot
		countdown.
		Phase 2: Application - Navigate to:
		Menu>Settings>Advanced>Admin Settings>Network
		Configuration
		• For more information, refer to Setting Up the Network on page
		3-2.

Voice Quality Monitoring

Note	This feature requires a license key for activation. Using this feature may require
	purchase of a license key or activation by Polycom channels. For more information,
	contact your Certified Polycom Reseller.

	The SoundPoint IP phones can be configured to generate various quality
	metrics for listening and conversational quality. These metrics can be sent
	between the phones in RTCP XR packets. The metrics can also be downloaded
	in SIP messages to a central voice quality report collector. The collection of
	these metrics is supported on the SoundPoint IP 330/320, 430, 501, 550, 560,
	600, 601, 650, and 670 phones and the SoundStation IP 4000 phone.
Note
Note	Voice Quality Monitoring is not supported on the SoundStation IP 6000 and 7000
	conference phones at this time.

	The RTCP XR packets are complaint with RFC 3611 - RTP Control Extended
	Reports (RTCP XR). The packets are sent to a report collector as specified in
	draft RFC draft-ietf_sipping_rtcp-summary-02.
	Three types of quality reports can be enabled:
	• Alert—Generated when the call quality degrades below a configurable
	threshold.
	• Periodic—Generated during a call at a configurable period.
	• Session—Generated at the end of a call.

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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 Voice Quality Monitoring, Three types of quality reports can be enabled

Models: SIP 3.1

Central

Voice Quality Monitoring

draft RFC draft-ietf_sipping_rtcp-summary-02.

Three types of quality reports can be enabled:

threshold.

• Periodic—Generated during a call at a configurable period.

• Session—Generated at the end of a call.

SIP 3.1 specifications