Configuring Your System

When a phone is configured with a DNS server, it will behave as follows by default:

•An initial attempt to resolve a hostname that is within the static DNS cache, for example to register with its SIP registrar, results in a query to the DNS.

•If the initial DNS query returns no results for the hostname or cannot be contacted, then the values in the static cache are used for their configured time interval.

•After the configured time interval has elapsed, a resolution attempt of the hostname will again result in a query to the DNS.

•If a DNS query for a hostname that is in the static cache returns a result, the values from the DNS are used and the statically cached values are ignored.

When a phone is not configured with a DNS server, it will behave as follows

•An attempt to resolve a hostname that is within the static DNS cache will always return the results from the static cache.

Support for negative DNS caching as described in RFC 2308 is also provided to allow faster failover when prior DNS queries have returned no results from the DNS server. For more information, go to http://tools.ietf.org/html/rfc2308 .

Configuration changes can be performed centrally at the boot server:

Central

Configuration file:

Specify DNS NAPTR, SRV, and A records for use when the phone is

(boot server)

sip.cfg

not configured to use a DNS server.

 

 

• For more information, refer to DNS Cache <dns/> on page A-100.

 

 

 

Configuration File Examples

Polycom recommends that you create another file with your organization’s modifications. If you must change any Polycom templates, back them up first.

For more information, refer to the “Configuration File Management on SoundPoint IP Phones” whitepaper at www.polycom.com/support/voice/.

Example 1

This example shows how to configure static DNS cache using A records IP addresses in SIP server address fields.

When the static DNS cache is not used, the sip.cfg configuration would look as follows:

reg.1.address="1001"

reg.1.server.1.address="172.23.0.140"

reg.1.server.1.port="5075"

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Polycom SIP 3.1 manual Example, Central
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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.
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