Skylar

BroTecs Technologies, Inc.

Achats via l'appli

À propos de l'application

Skylar is a highly secured, standard-based mobile VoIP softphone that works over 3G, 4G and Wi-Fi networks. Skylar incorporates advanced security settings, which allow for secure calls. Skylar provides superb sound quality and smooth VoIP experience using it’s intelligent network quality sensing capability including Adaptive Jitter Buffer, Packet Lost Concealment, Noise Suppression, Automatic Resampling and Gain Control. Satellite network friendly special audio codec algorithm ensures smooth call experience through low bandwidth channels. Graphical live call statistics including network details helps to provide the best and quick customer support.


Skylar is very flexible for branding, customization and licensing for service providers and enterprises. Please contact to [e-mail].


IMPORTANT: Skylar softphone is a standalone SIP based VoIP client application and is not bundled with VoIP service. A SIP server or subscription with a SIP-based VoIP provider is required to make and receive VoIP calls.


Key Features:
- Using the device’s existing contact list, Skylar facilitates easy and effective communication management with an intuitive interface.
- Multi-tasking support for background operation, such as receive incoming calls while using other applications.
- SIP account specific DTMF configurable option: ability to enter numbers and use an auto attendant via RFC 2833 and Inband
- Auto NAT type detection and NAT Traversal using STUN, TURN and ICE.
- Automatically discover the server by using DNS SRV.
- VPN Support
- TLS and SRTP Support for securing SIP signaling and voice.
- Advanced network quality detection for VoIP calls and live graphical statistics for individual call line.
- Multiple account support for up to 50 accounts on any SIP-compliant server
- Call history with a list of received, missed and dialed calls.
- Speakerphone, mute and hold functions.
- Dial Plan support for individual SIP account.
- Multiple call support – swap between two active calls, merge and split calls, transfer calls (attended and unattended)
- HD Wideband audio through G.722 codec.
- Enhanced/Optimized OPUS codec support for low bandwidth network.
- Audio codecs include G.722 (HD), G.729, G.711A/U (PCMA/PCMU), GSM and OPUS
- Automatic codec selection to ensure optimal call quality

N.B: If existing customers face any issues after update, please contact [e-mail] for further queries.

For Support: [e-mail]

Nouveautés

- Android 11 support
- Attended call transfer for SIP extension
- Bug fixes and performance improvement