IPitomy Cloud

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Template:Short description

IPitomy Acrobits Desktop Phone is a SIP-based softphone client designed for the IPitomy Cloud PBX. It provides VoIP calling, presence monitoring, call transfer, call parking, diagnostic logging, and configurable audio routing. Built on the Acrobits softphone engine, the application supports dynamic codec negotiation, NAT traversal, SRTP encryption (when enabled), and multi-account SIP operation depending on provisioning.

The application is used in call centers, offices, distributed workforces, and IPitomy reseller environments.

Overview

The IPitomy Acrobits Desktop Phone connects to the IPitomy Cloud PBX via SIP registration using the user's extension credentials. Once registered, the software synchronizes presence data, directory listings, call history, and provisioning settings.

The interface consists of:

  • A navigation panel (Keypad, Contacts, History, Settings)
  • A main workspace (active module display)
  • Presence indicators (BLF subscription results)
  • Call control overlays (during active calls)

The application supports both IPv4 and IPv6 where configured by the PBX.

SIP Architecture

The softphone registers to the PBX using standard SIP REGISTER messaging. It maintains the connection using periodic re-registration and SIP OPTIONS probing depending on PBX configuration.

Features include:

  • SIP over UDP or TCP (provider dependent)
  • SRTP capable (if PBX enforces or allows it)
  • NAT traversal using STUN and rport parameters
  • Adaptive jitter buffering
  • Codec negotiation with priority list provisioning

Supported codecs typically include:

  • G.711u (PCMU)
  • G.711a (PCMA)
  • G.722 HD Voice
  • Opus (if enabled)
  • GSM (fallback)

Interface Modules

Keypad

The keypad provides:

  • Manual number entry
  • DTMF for in-call menu navigation
  • On-screen call controls
  • Quick Dial presence list

During calls, keypad usage sends DTMF tones using RFC 2833 (out-of-band) signaling. In-band tones may be used depending on codec and PBX preference.

Quick Dial and BLF Monitoring

Quick Dial entries can be configured with Busy Lamp Field (BLF) monitoring using SIP SUBSCRIBE/NOTIFY mechanisms.

Presence states:

  • Available (SIP NOTIFY with "idle")
  • Busy (call active or ringing)
  • Unavailable (extension unregistered)
  • Unknown (no subscription response)

BLF allows:

  • Quick calling
  • Attended or blind transfer to monitored extensions
  • Monitoring parked call slots
  • Supervisory observation in call centers

BLF presence is polled periodically or pushed through NOTIFY packets depending on PBX behavior.

Contacts

The contacts module includes:

  • System directory (retrieved from PBX provisioning)
  • User-created Quick Dial entries
  • Extensions, departments, and general contacts

Search behavior supports substring matching on:

  • Name
  • Extension
  • URI phone number

Call History

The application stores:

  • Missed calls
  • Answered calls
  • Outgoing calls
  • Calls answered on another device registered to the same extension

Each call record includes:

  • Caller ID and name
  • Timestamp
  • Call duration (when applicable)
  • Completed or canceled call state
  • Ability to redial immediately

Call history is stored locally and may sync from server depending on PBX configuration.

Call Handling

The application supports:

  • Outbound calling
  • Voicemail server access
  • Multi-stage dialing (international)
  • In-call switching of audio devices
  • Call merging (if PBX conferencing enabled)

In-Call Controls

Mute

Disables sending audio via the microphone while maintaining RTP reception.

Hold

Places the remote party on server-based hold using SIP RE-INVITE. The PBX plays Music On Hold (MOH), not the softphone.

Audio Switching

Users may switch:

  • Microphone device
  • Playback device
  • Ringtone device

Switching creates a new local audio device binding without interrupting the SIP session.

Call Transfers

Blind Transfer

Blind transfers use SIP REFER immediately without first calling the target.

Procedure:

  1. In an active call, select Transfer.
  2. Enter extension or number.
  3. Select Blind Transfer.
  4. Softphone sends REFER with target URI.
  5. Remote party connects directly to recipient.

Blind transfers fail if:

  • The destination does not exist
  • PBX blocks the transfer
  • Target extension is forbidden by Class of Service

If a blind transfer fails, the original call may return depending on PBX settings.

Attended Transfer

Attended transfers involve placing the caller on hold and calling the intended recipient first.

Procedure:

  1. During call, select Transfer.
  2. Enter destination.
  3. Select Attended Transfer.
  4. Caller placed on PBX hold.
  5. Softphone initiates second SIP INVITE to contact.
  6. If recipient answers:
    1. User announces caller.
    2. User selects Complete Transfer (REFER or REFER via Re-INVITE).
  7. If recipient declines:
    1. User selects Return to Call.

If network or signaling interrupts the second call, the user may still retrieve the original caller.

Semi-Attended Transfer

If a user initiates an attended transfer but completes it before speaking to the recipient, it becomes a semi-attended (ringing) transfer.

Call Parking

Parking allows callers to be placed into shared PBX holding slots.

Typical park slots:

  • 701
  • 702
  • 703

The softphone monitors these slots via BLF.

Parking a Call

  1. Place the call on hold.
  2. Dial park slot (e.g., 701).
  3. Hang up or complete transfer.
  4. BLF indicator for park slot becomes active.

Retrieving a Parked Call

Users can retrieve by:

  • Clicking the BLF Park slot in Quick Dial
  • Manually dialing the slot number

If a slot times out, the PBX typically rings the original extension back.

Settings

About

Displays:

  • Version, build number
  • Platform info
  • Provisioning metadata
  • Licensing information

Account Setup

Fields include:

  • SIP username (extension ID)
  • SIP password
  • Registrar server
  • Outbound proxy (if provisioned)

Notifications

Allows selection of:

  • Ringtone tone
  • Text alert sound
  • Quiet modes (OS-dependent)

Sound

Sound configuration supports:

  • Device enumeration for microphones and speakers
  • Gain control (software amplification)
  • Ringtone device independent of call audio
  • Keypad tone output
  • Outgoing noise suppression using DSP
  • "Mute other applications" uses OS-level audio session attenuation

RTP audio stream analysis adjusts jitter buffer sizes dynamically.

Call Recording

If enabled, softphone recordings:

  • Are stored locally
  • Use raw PCM or compressed format depending on engine
  • May synchronize with PBX policies
  • Must follow local compliance laws (user must verify)

Controls

Options include:

  • Launch at login
  • Incoming call alerts (Full, Minimal, Notification-only)
  • Default calling app registration
  • Always-on-top mode
  • Setup wizard for initial permissions

Troubleshooting

This tab includes advanced diagnostic functions.

SIP Log

Shows SIP registration and messaging:

  • REGISTER
  • INVITE/BYE
  • REFER
  • SUBSCRIBE/NOTIFY
  • OPTIONS keep-alives

Developers and admins use logs to analyze:

  • Registration errors
  • NAT traversal behavior
  • Codec negotiation
  • Transfer issues

Diagnostic Data

Collects:

  • Pre-DSP mic audio
  • Post-DSP mic audio
  • Playback processing stages

Problem Reports

Exports:

  • SIP logs
  • Device info
  • OS audio routing data
  • Crash logs (if present)

Logout and Reset

Logout unregisters SIP credentials.

Reset Application returns all settings to defaults by wiping:

  • Cached SIP credentials
  • Audio preferences
  • Contact lists
  • Provisioning files

Troubleshooting

Common issues and behaviors:

One-Way Audio

Typically caused by:

  • NAT firewall blocking RTP
  • Wrong audio device bound
  • VPN routing SIP incorrectly

Registration Timeouts

Caused by:

  • Incorrect credentials
  • ISP SIP ALG interference
  • PBX unreachable

BLF Not Updating

Caused by:

  • PBX disabling presence
  • Network packet loss blocking NOTIFY

Network Requirements

Recommended:

  • 100 kbps per call (G.711)
  • QoS prioritization (DSCP EF)
  • Disable SIP ALG on routers
  • Stable latency under 150ms

Security

Compatible with:

  • SRTP (AES-128)
  • TLS SIP signaling
  • PBX-side authentication policies

Users may enforce:

  • Strong passwords
  • Encrypted transport
  • Limited IP ACLs on the PBX

Support

Support is provided by IPitomy Communications.

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