Polycom SIP 3.1 manual Attribute Value/s Description

Page 41

 

 

 

Application Development

 

 

 

 

 

 

 

 

 

Attribute

Value/s

Description

 

 

 

 

 

CallingPartyName

number

If the line is registered, the value is

 

 

 

the registered line display name.

 

 

 

If the line is not registered, the

 

 

 

value is the IP address of the

 

 

 

calling party.

 

 

 

 

 

CallingPartyDirNum

number

If the line is registered, the value is

 

 

 

the registered line number.

 

 

 

If the line is not registered, the

 

 

 

value is the IP address of the

 

 

 

calling party.

 

 

 

 

 

CalledPartyName

name

If the line is registered, the value is

 

 

 

the registered line display name.

 

 

 

If the line is not registered, the

 

 

 

value is the IP address of the

 

 

 

called party.

 

 

 

 

 

CalledPartyDirNum

number

If the line is registered, the value is

 

 

 

the registered line number.

 

 

 

If the line is not registered, the

 

 

 

value is the IP address of the

 

 

 

called party.

 

 

 

 

 

CallReference

number

An internal identifier for the call.

 

 

 

 

 

CallDuration

number in seconds

Duration of the call in seconds.

 

 

 

 

When the phone state polling URL is set and the phone receives a Call

Processing Request, the following example shows the transmitted data:

<PolycomIPPhone>

<CallLineInfo>

<LineKeyNum>1</LineKeyNum>

<LineDirNum>10</LineDirNum>

<LineState>Connected</LineState>

<CallInfo>

<CallState>Offering</CallState>

<CallType>Incoming</CallType>

<CalledPartyName>10</CalledPartyName>

<CalledPartyNumber>10</CalledPartyNumber>

<CallingPartyName>21</CallingPartyName>

<CallingPartyNumber>21@172.24.128.61</CallingPartyNumber>

<CallReference>0</CallReference>

<CallDuration>0</CallDuration>

</CallInfo>

</CallLineInfo>

<CallLineInfo>

<LineKeyNum>2</LineKeyNum>

<LineDirNum>35</LineDirNum>

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Contents SIP Trademark Information About This Guide Developer’s Guide SoundPoint IP / SoundStation IP Contents XML Errors TroubleshootingOverview What is the MicrobrowserWhat is Xhtml How to Create Applications Microbrowser Following new features were introduced in SIP New Features in SIPDeveloper’s Guide SoundPoint IP / SoundStation IP Supported Xhtml Elements Application DevelopmentBasic Tags Input Tags Link TagsFollowing link tag is supported Defines an anchorMethod=post This method Specified URL Following attributes are supportedMethod=get This method Input Img element defines an image Image TagsTables with no borders Table TagsAttribute Value/s Description MB Support Attribute Value/s Description Attribute Value/s Description Thead Tbody Meta Information Tags SoundPoint IP/SoundStation IP XML API Application InterfaceAction name is displayed Programmable Soft KeysAction Predefined Soft Key Action Default Action Name DescriptionSupported actions are described in the following table Html Hello World! /pbr Telephone Integration URIsAction Type Description Following format is supportedApps.push.serverRootURL Push Requests Href=KeyDoNotDisturbDNDSettings/aPhone State Priority Description Following table describes when to use a specific priorityWhere media.xhtml is defined as follows For example, to push the display of an important message Telephony Notification Events Attribute Value/s Description Outgoing Call Event Offhook Event Onhook Event URI Phone State PollingCall Line Information Attribute Value/s Description Device Information Network Configuration API Security Microbrowser User Interface Http SupportLaunching the Microbrowser Idle Display Microbrowser Navigation and Form EditingSet mb.main.home to the URL used for Microbrowser home Changing Configuration ParametersDeveloping an Xhtml Application Application Development Sample Applications Http//WEBSERVERADDRESSPORT/PLCM/Sample.xhtml Getting the Path Where BMP File has to be Saved Xhtml file in the sip.cfg configuration file Configure the Web server to deploy the above JSP fileReboot the phones Enter a stock symbol, then select the Get Quote soft keyTo develop an XML API application Body Html String result = Set apps.push.password to Developer’s Guide SoundPoint IP / SoundStation IP Symptom Problem Corrective Action XML ErrorsDeveloper’s Guide SoundPoint IP / SoundStation IP Unsupported elements and attributes are Unsupported Xhtml ElementsTag Type Tag Description Basic Tags Character Format TagsBlock Tags Tag Type Tag Description Output TagsLink Tags Frame TagsImage Tags Tag Type Tag Description Input TagsList Tags Style Tags Tag Type Tag Description Table TagsMeta Information Tags Programming TagsIndex Developer’s Guide SoundPoint IP / SoundStation IP POLYCOM, INC Application Programming Interface License API License Agreement for Development Purposes Support Services Export Controls
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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.