Call Transfer, Hold/localReminder/ on page A-67

Administrator’s Guide SoundPoint IP / SoundStation IP

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

Central	Configuration file:	Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
(boot server)	sip.cfg	a=inactive) outgoing hold signaling is used.
		• For more information, refer to SIP <SIP/> on page A-10.
		Specify local hold reminder options.
		• For more information, refer to Hold, Local Reminder
		<hold/><localReminder/> on page A-67.
		Specify the Music on Hold URI.
		• For more information, refer to Music on Hold <musicOnHold/> on
		page A-17.

	Configuration file:	Specify the Music on Hold URI.
	phone1.cfg	• For more information, refer to Music on Hold <musicOnHold/> on
		page A-17.

Local	Web Server	Specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold
	(if enabled)	signaling. The alternative is RFC 3264 (a=sendonly or a=inactive).
	(if enabled)	Navigate to: http://<phoneIPAddress>/appConf.htm#ls
		Navigate to: http://<phoneIPAddress>/appConf.htm#ls
		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	Use the SIP Configuration menu to specify whether or not to use RFC
	Interface	2543 (c=0.0.0.0) outgoing hold signaling. The alternative is RFC 3264
		(a=sendonly or a=inactive).

Call Transfer

Call transfer enables the user (party A) to move an existing call (party B) into a new call between party B and another user (party C) selected by party A. The phone offers three types of transfers:

• Blind transfers—The call is transferred immediately to party C after party A has finished dialing party C’s number. Party A does not hear ring-back.

• Attended transfers—Party A dials party C’s number and hears ring-back and decides to complete the transfer before party C answers. This option can be disabled.

• Consultative transfers—Party A dials party C’s number and talks privately with party C after the call is answered, and then completes the transfer or hangs up.

4 - 18

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 Call Transfer, Hold/localReminder/ on page A-67

Models: SIP 3.1

Administrator’s Guide SoundPoint IP / SoundStation IP

<hold/><localReminder/> on page A-67.

Call Transfer

SIP 3.1 specifications