Siperb (Session Initiation Protocol Endpoint Relay Bridge)

Siperb is a bridge between your traditional VoIP SIP PBX (like Asterisk) and WebRTC. It provides all the tools you need to enable WebRTC calling on your traditional PBX. Your Asterisk 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 a more of the features that Siperb offers. Either way, Siperb can provide this for you. Learn more about Siperb.

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About Siperb

Siperb sits in the cloud between your existing PBX and your users, to provide them with a state-of-the-art WebRTC enabled client endpoint (UAC).

There are various ways to connect with us if you want, or you don't even need to connect with us at all. It's up to you. You are welcome to use our system without connections for free, but then you must make sure that your PBX is fully WebRTC capable.

We can connect to your PBX (UAS) via our outbound registrations, or your PBX (UAS) can connect to us via our inbound registrations. This links us with you. You are not limited by these connections. You can make as many as you want. You will probably want to make one per extension on your PBX that you want to use with WebRTC. One of these connections can also be a TISP, like Twillio Elastic SIP Trunking. You are also not limited to the number of devices you can register your client endpoint (UAC) with.

We offer 3 client formats; Web, Tablet, and Mobile. While the Web format also works well on most tablets, it's a better experience to use the mobile app on the tablets and mobile devices so that you can make use of push notifications. They are fully compatible and respond easily to the screen size changes. The client is available on the Play and App store, and for the Web application you can install the PWA or simply access the web page with the help of a simple Browser Extension. The extension is available for Safari (Mac), Firefox (Mac/Windows/Linux) and Chrome (Mac/Windows/Linux).

You then need to select how the signaling and media will flow over these connections. For example, if you have an older PBX that doesn't support WebRTC, you should choose full media relay. This will take the WebRTC media (Opus DTLS) and transcode it into something suitable for your PBX like G711. This comes at the most cost, as media traverses our network. It also will have the highest latency because of the extra hop. We have many points of presence around the world, so you (or the system) can choose the lowest latency. In this instance, signaling is still relayed to your server, so you can perform transfers, holds, park, and all the features you are used to within your PBX.

If your systems are more up-to-date, and can create a DTLS stream, but you don't have WebSocket's active or configured, we can relay the offer using UDP (via your connection). As with full media relay, signaling is relayed with your server, but the media is end-to-end encrypted, meaning that we are not even in the media path. This means the cost is less and the latency should be lower. Media will flow directly from the client endpoint (UAC) to your server (UAS), so there may be some firewall changes to make. We try to send and receive over the same port, but this may not always be possible - it will depend on your network. If this doesn't work, you may have to use the full media option.

Lastly, if you are fully ready with WebRTC, and just want to make use of some of the many features of Siperb, you are welcome to use the direct client to server method. With this option, your own server details are entered into the system and provisioned to the client endpoint, this makes the WebRTC connection directly between the client endpoint (UAC) and your server (UAS). Media is also end-to-end encrypted and does not pass through us. In this configuration, we are not even in the signaling path, so we are not part of any calls. This means that features like Push Notifications, Video Conferencing and Voicemail will not work. It does however mean the Web Portal features, like provisioning, call details records, call recording storage, transcribing and analysis will still work.

Features you'll love

Siperb Features

Features you will find in all Account Types. (* Fees Apply)

  • Provisioning
  • SIPS Proxy
  • DTLS to RTP transcoding *
  • Push Notifications
  • Audio & Video Calling
  • Voicemail (to email)
  • Instant Messaging
  • Video Conferencing *

Client Endpoint Features

Features found on the Web and Mobile Application. (* Fees Apply)

  • WebRTC Phone
  • Call Recording (Audio & Video)
  • Call Transfer (Blind and Attended)
  • Call Hold
  • Call Mute
  • 3 Way Call Conference
  • Buddy Management & Presence
  • Chat (with file transfer)

Web Portal Features

Features found in the Admin Web Portal. (* Fees Apply)

  • Call Recording Storage *
  • Call Recoding Transcribing & Analysis *
  • Call CDR Storage *
  • Call QOS Storage and Analysis *

Screenshot Gallery

Samples of how the UI looks in the Web and Mobile Application

Frequently Asked Questions

We often get the following questions.


Do I need my own Asterisk PBX?

This application works best when connected to your own on-site or hosted PBX. It doesn't have to be Asterisk.


Can I connect to another service provider like Twillio or my own TISP?

Yes, you can. You will need to be using the full media connection.


Do you have a free option?

Yes, there are various free options.


Does my own PBX have to be WebRTC enabled?

If you are using a Direct Connection, then yes, but if you are using Full Media Connection, no.