Administrator’s Guide SoundPoint IP / SoundStation IP

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

voIpProt.server.x.transport

DNSnaptr or

DNSnapt

If set to Null or DNSnaptr:

 

TCPpreferre

r

If voIpProt.server.x.address is a

 

d or

 

hostname and voIpProt.server.x.port is 0 or

 

UDPOnly or

 

Null, do NAPTR then SRV look-ups to try to

 

TLS or

 

discover the transport, ports and servers, as

 

TCPOnly

 

per RFC 3263. If

 

 

 

voIpProt.server.x.address is an IP

 

 

 

address, or a port is given, then UDP is used.

 

 

 

If set to TCPpreferred:

 

 

 

TCP is the preferred transport, UDP is used if

 

 

 

TCP fails.

 

 

 

If set to UDPOnly:

 

 

 

Only UDP will be used.

 

 

 

If set to TLS:

 

 

 

If TLS fails, transport fails. Leave port field

 

 

 

empty (will default to 5061) or set to 5061.

 

 

 

If set to TCPOnly:

 

 

 

Only TCP will be used.

 

 

 

NOTE: TLS is not supported on SoundPoint

 

 

 

IP 300 and 500 phones.

 

 

 

 

voIpProt.server.x.expires

positive

3600

The phone’s requested registration period in

 

integer,

 

seconds.

 

minimum

 

Note: The period negotiated with the server

 

300

 

 

 

may be different. The phone will attempt to

 

 

 

re-register at the beginning of the overlap

 

 

 

period. For example, if “expires”=3600 and

 

 

 

“overlap”=60, the phone will re-register after

 

 

 

3540 seconds (3600 – 60).

 

 

 

 

voIpProt.server.x.expires.overlap

positive

60

The number of seconds before the expiration

 

integer,

 

time returned by server x at which the phone

 

minimum 5,

 

should try to re-register. The phone will try to

 

maximum

 

re-register at half the expiration time returned

 

65535

 

by the server if that value is less than the

 

 

 

configured overlap value.

 

 

 

 

voIpProt.server.x.register

0, 1

1

If set to 0, calls can be routed to an outbound

 

 

 

proxy without registration.

 

 

 

 

voIpProt.server.x.retryTimeOut

Null or

0

If set to 0 or Null, use standard RFC 3261

 

non-negativ

 

signaling retry behavior. Otherwise

 

e integer

 

retryTimeOut determines how often retries

 

 

 

will be sent.

 

 

 

Units = milliSeconds. (Finest resolution =

 

 

 

100ms).

 

 

 

 

A - 8

Page 161 Page 163
Page 162
Image 162
Polycom SIP 3.1 manual VoIpProt.server.x.address is an IP
Page 161 Page 163

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