Session Initiation Protocol (SIP)

Hold Implementation

 

The phone supports both currently accepted means of signaling hold.

 

The first method, no longer recommended due in part to the RTCP problems

 

associated with it, is to set the “c” destination addresses for the media streams

 

in the SDP to zero, for example, c=0.0.0.0.

 

The second, and preferred, method is to signal the media directions with the

 

“a” SDP media attributes sendonly, recvonly, inactive, or sendrecv. The hold

 

signaling method used by the phone is configurable (refer to SIP <SIP/>on

 

page A-10), but both methods are supported when signaled by the remote end

 

point.

Note

 

Even if the phone is set to use c=0.0.0.0, it will not do so if it gets any sendrecv,

 

sendonly, or inactive from the server. These flags will cause it to revert to the other

 

hold method.

 

 

Reliability of Provisional Responses

The phone fully supports RFC 3262 - Reliability of Provisional Responses.

Transfer

The phone supports transfer using the REFER method specified in draft-ietf-sip-cc-transfer-05 and RFC 3515.

Third Party Call Control

The phone supports the delayed media negotiations (INVITE without SDP) associated with third party call control applications.

When used with an appropriate server, the User Agent Computer Supported Telecommunications Applications (uaCSTA) feature on the phone may be utilized for remote control of the phone from computer applications such as Microsoft Office Communicator.

The phone is compliant with “Using CSTA for SIP Phone User Agents (uaCSTA), ECMA TR/087” for the Answer Call, Hold Call, and Retrieve Call functions and “Services for Computer Supported Telecommunications Applications Phase III”, ECMA – 269 for the Conference Call function.

This feature is enabled by configuration parameters described in SIP <SIP/> on page A-10and Registration <reg/> on page A-107and needs to be activated by a feature application key.

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Polycom SIP 3.1 manual Hold Implementation, Reliability of Provisional Responses, Transfer, Third Party Call Control

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.