Polycom SIP 3.1 manual Application Development

Page 49

Application Development

aSet apps.push.messageType to the appropriate display priority. For example, apps.push.messageType=3

bSet apps.push.serverRootURL to the application server root URL.

For example,

apps.push.serverRootURL=http://172.24.128.85:8080/sampleapps

cSet apps.push.username to the appropriate username. For example, apps.push.username=bob

The username and password are required to authenticate incoming push requests to the phone.

dSet apps.push.password to the appropriate password. For example, apps.push.password=1234

10.(Optional.) If you are including telephone event notifications in your application, do the following:

aSet apps.telNotification.URL to the location where notifications should be sent.

For example,

apps.telNotification.URL=http://172.24.128.85:8080

If this URL is set to Null, the notifications events will not be sent.

bSet apps.telNotification.incomingEvent to 1 or 0 (for Enable or Disable respectively).

For example, apps.telNotification.incomingEvent=1

cSet apps.telNotification.outgoingEvent to 1 or 0 (for Enable or Disable respectively).

For example, apps.telNotification.outgoingEvent=1

dSet apps.telNotification.offhookEvent to 1 or 0 (for Enable or Disable respectively).

For example, apps.telNotification.offhookEvent=1

eSet apps.telNotification.onhookEvent to 1 or 0 (for Enable or Disable respectively).

For example, apps.telNotification.onhookEvent=1

11.(Optional.) If you are including phone state polling requests in your application, do the following:

aSet apps.statePolling.URL to the location where requested information should be sent.

For example, apps.statePolling.URL=http://172.24.128.85:8080 If this URL is set to Null, the requested information will not be sent.

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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=get This method Specified URL Following attributes are supportedMethod=post 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 KeysSupported actions are described in the following table Action Default Action Name DescriptionAction Predefined Soft Key Html Hello World! /pbr Telephone Integration URIsAction Type Description Following format is supportedApps.push.serverRootURL Push Requests Href=KeyDoNotDisturbDNDSettings/aWhere media.xhtml is defined as follows Following table describes when to use a specific priorityPhone State Priority Description 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 EditingDeveloping an Xhtml Application Changing Configuration ParametersSet mb.main.home to the URL used for Microbrowser home 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 TagsList Tags Tag Type Tag Description Input TagsImage 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.
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