Administrator’s Guide SoundPoint IP / SoundStation IP

SIP <SIP/>

This configuration attribute is defined as follows:

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

voIpProt.SIP.useContactInReferTo

0, 1

0

If set to 0, the “To URI” is used in the REFER.

 

 

 

If set to 1, the “Contact URI” is used in the

 

 

 

REFER.

 

 

 

 

voIpProt.SIP.useRFC2543hold

0, 1

0

If set to 1, use the obsolete c=0.0.0.0

 

 

 

RFC2543 technique, otherwise, use SDP

 

 

 

media direction attributes (such as

 

 

 

a=sendonly) per RFC 3264 when initiating

 

 

 

hold. In either case, the phone processes

 

 

 

incoming hold signaling in either format.

 

 

 

 

voIpProt.SIP.useSendonlyHold

0, 1

1

If set to 1, the phone will send a reinvite with

 

 

 

a stream mode attribute of “sendonly” when a

 

 

 

call is put on hold. This is the same as the

 

 

 

previous behavior.

 

 

 

If set to 0, the phone will send a reinvite with

 

 

 

a stream mode attribute of “inactive” when a

 

 

 

call is put on hold.

 

 

 

NOTE: The phone will ignore the value of this

 

 

 

parameter if set to 1 when the parameter

 

 

 

voIpProt.SIP.useRFC2543hold is also set

 

 

 

to 1 (default is 0).

 

 

 

 

voIpProt.SIP.lcs

0, 1

0

If set to 1, the proprietary “epid” parameter is

 

 

 

added to the From field of all requests to

 

 

 

support Microsoft Live Communications

 

 

 

Server.

 

 

 

 

voIpProt.SIP.ms-forking

0, 1

0

If set to 0, support for MS-forking is disabled.

 

 

 

If set to 1, support for MS-forking is enabled

 

 

 

and the phone will reject all Instant Message

 

 

 

INVITEs. This parameter is relevant for

 

 

 

Microsoft Live Communications Server

 

 

 

server installations.

 

 

 

Note that if any end point registered to the

 

 

 

same account has MS-forking disabled, all

 

 

 

other end points default back to non-forking

 

 

 

mode. Windows Messenger does not use

 

 

 

MS-forking so be aware of this behavior if

 

 

 

one of the end points is Windows Messenger.

 

 

 

 

voIpProt.SIP.dialog.usePvalue

0, 1

0

If set to 0, phone uses "pval" field name in

 

 

 

Dialog. This obeys the

 

 

 

draft-ietf-sipping-dialog-package-06.txt draft.

 

 

 

If set to 1, phone uses a field name of

 

 

 

"pvalue".

 

 

 

 

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Polycom SIP 3.1 manual Parameter if set to 1 when the parameter, To 1 default is

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.