Siperb is a Softphone for Asterisk, FreeSWITCH or any SIP/VOIP PBX. We provide both the Softphone and WebRTC-to-SIP Proxy to enable WebRTC calling on your existing PBX. Your PBX may not be WebRTC ready, or it may be, but you lack the Browser Phone required to make use of WebRTC, or you may just want more of the features that Siperb offers. Either way, Siperb can provide this for you.

Use your existing PBX to seamlessly integrate with the advanced WebRTC capabilities we provide. At Siperb, we act as a proxy, bridging your current systems to our robust WebRTC client. This allows you the flexibility to connect with us directly or continue independently if your PBX is fully WebRTC capable. Choose the path that best suits your infrastructure needs.

WebRTC-to-SIP Proxy Features:

• Supports SIP Over WebRTC: Our WebRTC client supports both Asterisk PBX and FreeSWITCH PBX.
• Web and Mobile WebRTC Client: Our WebRTC based Softphone client runs as a PWA and as a native mobile application.
• End-to-end Encrypted: We can act as a proxy, ensuring WebRTC calls are end-to-end encrypted.
• Three Ways to Connect with Us: We provide inbound and outbound registrations, and even offer register-less connections.
• Auto-Provisioned WebRTC: Forget about passwords - WebRTC details are automatically provisioned.
• Media Transcoding: If selected, call media can be transcoded to suit your PBX configuration.
• Mobile & Web Push Notifications: Our SIP Proxy supports mobile and web push notifications, ensuring users are able to receive calls at any time, on any device.

Softphone Features:

• Call Recording (Audio & Video): Our Softphone supports call recording for both audio and video calls. Recordings can be stored locally or in the cloud, and are easily accessible for playback and download.
• Call Transfer (Blind and Attended): Our Softphone supports both blind and attended call transfer. Users can easily transfer calls to other extensions or external numbers with just a few clicks.
• 3 Way Call Conference: Our Softphone supports client side 3-way call conferencing. Users can easily add additional participants to an existing call without any server configuration.

Admin Control Panel:

• WebRTC Device Provisioning: All the important information like SIP details, settings, and even authentication details are automatically provisioned.
• Device Management: Configure device settings, and view device history including SIP trace and Registration Log Data.
• Connection Management: Connections are the SIP endpoints that connect us to you. Like normal SIP registrations, we provide various ways to configure connections so you can always make and receive your calls.