Configuring Your System

Hosted VoIP Service

Provider

Call Server 1B

Call Server 1A

DNS Server

Internet

PSTN

Phone Configuration

 

VoIP SMB Customer

SIP Capable Router

Premise

 

Server2

 

 

`

 

`

PSTN Gateway

 

 

`

 

`

The phones at the customer site are configured as follows:

Server 1 (the primary server) will be configured with the address of the service provider call server. The IP address of the server(s) to be used will be provided by the DNS server. For example:

reg.1.server.1.address="voipserver.serviceprovider.com"

Server 2 (the fallback server) will be configured to the address of the router/gateway that provides the fallback telephony support and is on-site. For example:

reg.1.server.2.address=172.23.0.1

Note

It is possible to configure the phone for more than two servers per registration, but

 

you need to exercise caution when doing this to ensure that the phone and network

 

load generated by registration refresh of multiple registrations do not become

 

excessive. This would be of particularly concern if a phone had multiple

 

registrations with multiple servers per registration and it is expected that some of

 

these servers will be unavailable.

 

 

 

Phone Operation for Registration

 

After the phone has booted up, it will register to all the servers that are

 

configured.

 

Server 1 is the primary server and supports greater SIP functionality than any

 

of servers. For example, SUBSCRIBE/NOTIFY services (used for features such

 

as shared lines, presence, and BLF) will only be established with Server 1.

 

Upon registration timer expiry of each server registration, the phone will

 

attempt to re-register. If this is unsuccessful, normal SIP re-registration

 

behavior (typically at intervals of 30 to 60 seconds) will proceed and continue

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Polycom SIP 3.1 manual Phone Configuration, Phone Operation for Registration, Configured

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.

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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.

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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.