OptiMobile UniPhone is a dual mode 3G/GSM and Voice over IP (VoIP) software client for Nokia Series 60 3rd edition devices. The user experience is very close to the telephony application already present in your mobile phone. With UniPhone you can dial and receive phone calls, and make phone calls from the built-in phone book and call log.

UniPhone together with a VoIP subscription from an operator converts your mobile phone into a perfect replacement for DECT phones. As a bonus you can close down your fixed line subscription. UniPhone is also ideal for small to medium enterprises that have an IP-PBX. With UniPhone enhanced mobile phones there will be no need for fixed IP desk phones and the office will be completely mobile.

UniPhone is a full featured SIP client enabling Voice over IP telephony towards SIP infrastructure, either pure point to point VoIP or VoIP to PSTN calls. By default UniPhone uses VoIP line for calling within Wi-Fi coverage and circuit switched 3G/GSM for calling out of Wi-Fi coverage. It is also possible to use 3G as data bearer for VoIP.

OptiMobile UniPhone features

• SIP compatibility allows you to:
  • communicate with other SIP compatible VoIP applications and systems.
  • call a regular phone using a SIP-PSTN gateway typically through a VoIP operator or corporate IP-PBX.
  • call a computer or IP-phone in a private network through a SIP proxy.
  • call any host by its name or IP address.
  • handle multiple calls: put call on hold and switch between two active calls.
• Support for sending DTMF signals.
• Support for SIP URL syntax.
• Configurable number handling with support for a wide variation of systems, from generic operators to more advanced enterprise PBX systems. The number handling settings are:
  • Country/Region code
  • Convert plus to
  • National prefix
  • External line prefix
  • Max internal num len
• Selectable preference of early media to support custom streamed ring tones and other feedback tones with early media.
• Several VoIP profiles with per profile selectable settings.
• AP handling with dynamic VoIP profile attachments.
• DTMF modes: RTP out-of-band, RTP in-band and SIP INFO.
• Selectable autostart of UniPhone at phone start-up.
• Adjustable Noise reduction to be able to set how sensitive the phone microphone is to sound before interpreting it as speech.
• Integration of platform phone Contacts, Main screen dialler and Call Log makes the phone easy to use.
• 3G/GSM compatibility allows you to using 3G/GSM networks for dialling when 3G/GSM Line used. Within Wi-Fi coverage VoIP line is used for dialling, otherwise the circuit switched 3G/GSM line is automatically selected.
• Support for DTMF based and operator based routing of outgoing 3G/GSM calls, for call routing through e.g. a corporate PBX.
• Handling emergency calls using cellular 3G/GSM network.
• Supported languages: UK English and Swedish.

RFCs at least supported

• RFC 3261: SIP: Session Initiation Protocol
• RFC 2327/4566: SDP: Session Description Protocol
• RFC 3264: An Offer/Answer Model with Session Description Protocol (SDP)
• RFC 4317: Session Description Protocol (SDP) Offer/Answer Examples
• RFC 2617: HTTP Authentication: Basic and Digest Access Authentication
• RFC 3489: STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
• RFC 3581: An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing
• RFC 3608: Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration
• RFC 4320: Actions Addressing Identified Issues with the Session Initiation Protocol's (SIP) Non-INVITE Transaction
• RFC 4321: Problems Identified Associated with the Session Initiation Protocol's (SIP) Non-INVITE Transaction
• RFC 2976: The SIP INFO Method
• RFC 3550: RTP: A Transport Protocol for Real-Time Applications
• RFC 3951: Internet Low Bit Rate Codec (iLBC)
• RFC 3952: Real-time Transport Protocol (RTP) Payload Format for Internet Low Bit Rate Codec (iLBC) Speech
• RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals.

Supported media types

G.711 (u-law and a-law), GIPS iLBC (30 ms) and Enhanced G.711. Note that the SDP codec priority list of the SIP proxy/media gateway is used when no audio codec is locked in VoIP profile advanced settings.

3G/GSM call interworking

Interworking with native telephony application with missed incoming VoIP call notifications.