SaudiVoip APK
APK Version History
- Version
- 3.2 (3003)
- Architecture
- arm64-v8a,armeabi-v7a,x86,x86_64
- Release Date
- September 30, 2021
- Requirement
- Android 4.1+
Download [ 4.4 MB ]
Safe
- Version
- 3.0 (3001)
- Architecture
- armeabi
- Release Date
- June 11, 2021
- Requirement
- Android 4.1+
Download [ 3.5 MB ]
Safe
- Version
- 2.0 (2001)
- Architecture
- armeabi
- Release Date
- June 11, 2021
- Requirement
- Android 4.0+
Download [ 3.0 MB ]
Safe
- Version
- 1.3 (1005)
- Architecture
- armeabi
- Release Date
- June 11, 2021
- Requirement
- Android 4.0+
Download [ 2.5 MB ]
Safe
- Version
- 1.2 (1004)
- Architecture
- armeabi
- Release Date
- June 11, 2021
- Requirement
- Android 4.0+
Download [ 2.5 MB ]
Safe
About Radio FM 90s
Saudi Voip dialer is a mobile app compatible with all Android OS smartphones, which enables the users for making VoIP calls from their mobile to anywhere around the world. Calls can be made from data-enabled mobile devices with internet connectivity via 3G/Edge/Wifi.
After installing the users need to provide the SIP Username and Password for registering to the app.
Features
★Elegant UI
★Cheap rate
★crystal clear sound quality
★It uses SIP-based protocol for Signaling
★It supports all SIP standard switches
★It RUNS behind NAT or private IP
★Unremitting call connection even in low bandwidth areas
★The jitter buffer technology ensures smooth voice during the call
★Integration of native contact book
★The silent suppression and comfort noise generation feature reduces the bandwidth usage
★Auto balance display
★Actual SIP response message display
★User-friendly interface
★Recent call history display
★Balance display
★Call duration display
★Features to make the call mute, enable the speaker mode, view the keypad, increase and decrease the media volume, enable the Bluetooth during a call/
After installing the users need to provide the SIP Username and Password for registering to the app.
Features
★Elegant UI
★Cheap rate
★crystal clear sound quality
★It uses SIP-based protocol for Signaling
★It supports all SIP standard switches
★It RUNS behind NAT or private IP
★Unremitting call connection even in low bandwidth areas
★The jitter buffer technology ensures smooth voice during the call
★Integration of native contact book
★The silent suppression and comfort noise generation feature reduces the bandwidth usage
★Auto balance display
★Actual SIP response message display
★User-friendly interface
★Recent call history display
★Balance display
★Call duration display
★Features to make the call mute, enable the speaker mode, view the keypad, increase and decrease the media volume, enable the Bluetooth during a call/