Configuration Files

 

 

 

 

 

 

 

 

 

Permitted

 

 

Attribute

Values

Default

Interpretation

voIpProt.SIP.connectionReuse.

0, 1

0

If set to 0, this is the old behavior.

useAlias

 

 

If set to 1, phone uses the connection reuse

 

 

 

draft which introduces "alias".

 

 

 

 

voIpProt.SIP.sendCompactHdrs

0, 1

0

If set to 0, SIP header names generated by

 

 

 

the phone use the long form, for example

 

 

 

‘From’.

 

 

 

If set to 1, SIP header names generated by

 

 

 

the phone use the short form, for example ‘f’.

 

 

 

 

voIpProt.SIP.keepalive.

0, 1

0

If set to 1, the session timer will be enabled.

sessionTimers

 

 

If set to 0, the session timer will be disabled,

 

 

 

and the phone will not declare “timer” in

 

 

 

“Support” header in INVITE. The phone will

 

 

 

still respond to a re-INVITE or UPDATE. The

 

 

 

phone will not try to re-INVITE or do UPDATE

 

 

 

even if remote end point asks for it.

 

 

 

 

voIpProt.SIP.requestURI.E164.

0, 1

0

If set to 1, ‘+’ global prefix is added to E.164

addGlobalPrefix

 

 

user parts in sip: URIs:.

 

 

 

 

voIpProt.SIP.

0, 1

1

If set to 1, a transfer can be completed during

allowTransferOnProceeding

 

 

the proceeding state of a consultation call.

 

 

 

If set to 0, a transfer is not allowed during the

 

 

 

proceeding state of a consultation call.

 

 

 

If set to Null, the default value is used.

 

 

 

 

voIpProt.SIP.dialog.useSDP

0, 1

0

If set to 0, new dialog event package draft is

 

 

 

used (no SDP in dialog body).

 

 

 

If set to 1, for backwards compatibility, use

 

 

 

this setting to send SDP in dialog body.

 

 

 

 

voIpProt.SIP.pingInterval

0 to 3600

0

The number in seconds to send "PING"

 

 

 

message. This feature is disabled by default.

 

 

 

 

voIpProt.SIP.useContactInReferTo

0, 1

0

If set to 1, the Contact URI is used.

 

 

 

If set to 0, the TO URI is used (previous

 

 

 

behavior).

 

 

 

 

voIpProt.SIP.serverFeatureControl.cf

0, 1

0

If set to 1, server-based call forwarding is

 

 

 

enabled. The call server has control of call

 

 

 

forwarding.

 

 

 

If set to 0, server-based call forwarding is not

 

 

 

enabled. This is the old behavior.

 

 

 

 

voIpProt.SIP.serverFeatureControl.

0, 1

0

If set to 1, server-based DND is enabled. The

dnd

 

 

call server has control of DND.

 

 

 

If set to 0, server-based DND is not enabled.

 

 

 

This is the old behavior.

 

 

 

 

A - 11

Page 164 Page 166
Page 165
Image 165
Polycom SIP 3.1 manual Permitted Attribute Values Default Interpretation
Page 164 Page 166

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.
Top Page Image Contents