Network Address Translation, Corporate Directory

Administrator’s Guide SoundPoint IP / SoundStation IP

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

Central	Configuration file:	Specify whether to filter incoming RTP packets by IP address,
(boot server)	sip.cfg	whether to require symmetric port usage, whether to jam the
		destination port and specify the local RTP port range start.
		• For more information, refer to RTP <rtp/> on page A-57.

Local	Web Server	Specify whether to filter incoming RTP packets by IP address,
	(if enabled)	whether to require symmetric port usage, whether to jam the
	(if enabled)	destination port and specify the local RTP port range start.

		Navigate to: http://<phoneIPAddress>/netConf.htm#rt
		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.

Network Address Translation

The phone can work with certain types of network address translation (NAT). The phone’s signaling and RTP traffic use symmetric ports (the source port in transmitted packets is the same as the associated listening port used to receive packets) and the external IP address and ports used by the NAT on the phone’s behalf can be configured on a per-phone basis.

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

Central	Configuration file:	Specify the external NAT IP address and the ports to be used for
(boot server)	sip.cfg	signaling and RTP traffic.
		• For more information, refer to Network Address Translation
		<nat/> on page A-120.

Local	Web Server	Specify the external NAT IP address and the ports to be used for
	(if enabled)	signaling and the RTP traffic.
	(if enabled)	Navigate to: http://<phoneIPAddress>/netConf.htm#na
		Navigate to: http://<phoneIPAddress>/netConf.htm#na
		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.

Corporate Directory

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.

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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 Network Address Translation, Corporate Directory, Nat/ on page A-120

Models: SIP 3.1

Administrator’s Guide SoundPoint IP / SoundStation IP

Network Address Translation

<nat/> on page A-120.

Corporate Directory

SIP 3.1 specifications