Last updated on February 24, 2016

Widget: SIP Call

The SIP Call Widget utilizes the SIP (Session Initiation Protocol) signalling protocol for Internet Telephony. This allows Appspace to establish sessions for audio/video conferencing and call forwarding over IP networks effectively creating a virtual concierge.

SIP Call Widget
SIP Call Actions
SIP Call Events
SIP Widget Front End
SIP Call Properties

Prerequisites

Software Requirements:

Appspace 5.1.x.
Appspace player version – 5.1.x.
Cisco Unified Call Manager (CUCM) endpoint account credentials.

Hardware Requirements:

Edge 340 Firmware 1.1 onwards.
Web Cam (applicable for video calling).
Speakers and microphone (applicable for voice and video calling).
Input hardware such as touch screen monitor or mouse.

SIP Call Widget

The SIP Widget can be implemented in 2 main ways:

To act as an invisible widget that handles only voice calls.
To act as a video call window where the SIP Widget itself will function as the video feed display window.

Alone by itself, the SIP Widget does not provide a visible interface for user operation (no keypad to dial a number). The widget requires and depends on interoperability with other widgets such as the text and media zone widgets. The SIP Widget itself mainly receives messages and then proceeds to generate events to execute and trigger corresponding actions. The widget can be leveraged to automatically call, accept, and end phone/video communications. The high-level workflow for this functionality is defined as the following:

Insert the SIP Widget as a layout object into an application.
Configure the widget to be registered with the Cisco Unified Call Manager.
Start, accept and end calls through actions.
Respond to a collection of SIP events through application scripting.

SIP Call Actions

The following table lists various possible actions that can be mapped as action scripts within the application:

Do	When	Action
Call	On Click/Touch	Initiate a call based on a preset number
Terminate	On Click/Touch	Terminate an existing call
Accept	On Click/Touch	Accept incoming call
Decline	On Click/Touch	Decline incoming call (it is also possible to set a reason value on this i.e Declined, Busy)
Set Speaker Volume	On Click/Touch	This is to either turn on or turn off volume. Possible values are from 0 or 100
Enable Microphone	On Click/Touch	Boolean value to either mute or unmute the microphone
Send DTMF	On Click/Touch	1: Sends DTMF tone for key #1 2 or A, B, C: Sends DTMF tone for key #2 3 or D, E, F: Sends DTMF tone for key #3 4 or G, H, I: Sends DTMF tone for key #4 5 or J, K, L: Sends DTMF tone for key #5 6 or M, N, O: Sends DTMF tone for key #6 7 or P, Q, R, S: Sends DTMF tone for key #7 8 or T, U, V: Sends DTMF tone for key #8 9 or W, X, Y, Z: Sends DTMF tone for key #9 0: Sends DTMF tone for key #0 “star” or +: Sends DTMF tone for key * #: Sends DTMF tone for #

Note

These are just examples of different actions that can be issued/assigned in order to map a keypad or call function buttons for the SIP Call Widget.

SIP Call Events

The SIP widget is able to generate events based on certain conditions. The following table lists various events that can be read/accepted by the Appspace framework and be used as part of the event list that is available for application scripting:

Event Details
On Idle	This event is raised when the SIP client is on idle
Incoming Received	This event is raised when an incoming call signal is detected at the SIP client
Outgoing (Progress)	This event is raised when a call from the SIP client to another endpoint is in progress
Outgoing (Ringing)	This event is raised when an outgoing call is ringing at the targeted SIP endpoint
Connected	This event is raised when a call is connected/answered between the SIP client and another endpoint
Connected Stream Running	This event is raised when the media streams are established and running
Call Transferred	This event is raised when a call gets transferred to another SIP client
Call Error	This event is raised when an error status occur
Call Ended	This event is raised when a call ends at the SIP client
Call Paused	This event is raised when a call is paused at the SIP client
Call Resumed	This event is raised when a call in Paused state gets resumed
Call Released	This event is raised when the SIP client no longer retains the call object

Note

These are just examples of different events that can be read/accepted in order to map actions based on said events.

SIP Call Properties

This article lists the available SIP widget player properties. By using these player properties, you are able to configure your Edge 340 with specific behavior for the SIP widget.

Requirements

Only supported on Edge 340 devices running firmware 1.2 and above
Only supported on Appspace Core 5.3.x and above

Set Output Video Resolution (to be viewed on the Edge 340 endpoint)

Property: custom.video.endpoint.size
Format: <width>x<height>
Unit: pixels
Sample: 800×600

Set Local Preview Resolution

Property: custom.video.local.size
Format: <width>x<height>
Unit: pixels
Sample: 800×600

Set Download Bandwidth

Property: custom.network.bandwidth.download
Format: integer
Unit: KBytes
Sample: 800×600

Set Upload Bandwidth

Property: custom.network.bandwidth.upload
Format: integer
Unit: KBytes
Sample: 800×600

Set Audio UDP/RTP Port Range

Property: custom.network.portrange.audio
Format: <minPort>,<maxPort>
Unit: N/A
Sample: 2000,2010

Set Video UDP/RTP Port Range

Property: custom.network.portrange.video
Unit: N/A
Format: <minPort>,<maxPort>
Sample: 2000,2010

Set SIP Call Signalling Transport Protocol

Property: custom.network.protocol
Format: <protocol>,<port>
Unit: N/A
Sample: udp,2010

Set DSCP value for SIP Signalling Packet

Property: custom.network.dscp
Format: <dscp>
Unit: N/A

Enable Software Echo Cancellation

Property: custom.echocancellation.enable
Format: Boolean
Sample: True or False