Browser phone setup for CallOrbit webphone

Who this guide is for

Audience: Admins and agents setting up browser-based calling.

Launch a new CallOrbit workspace with the right customer, users, and voice basics.

Use this guide when you want the setup to be correct the first time and easy for another admin, manager, or supervisor to verify later.

What this workflow helps you accomplish

Configure the CallOrbit browser webphone so every team member can make and receive calls directly from their browser without installing any desktop software or hardware phone.

This workflow matters because numbers, routing, access, and reporting in CallOrbit are connected. Skipping one setup detail usually creates avoidable support work later.

• Step 1: Open the Voice Setup page from the portal sidebar and navigate to the Browser Phone section to check that your workspace has browser calling enabled and available SIP seats assigned to users.
• Step 2: For each team member who needs browser calling, confirm their user profile has a browser seat allocated and that their extension number is set. The extension acts as their internal reachable line.
• Step 3: Have each user open the Webphone tab in the portal. The browser will prompt for microphone permission — grant access so the webphone can transmit and receive audio during calls.
• Step 4: Test the webphone by placing a call to a test number. If audio does not work, check that no other application is using the microphone, that the correct audio device is selected, and that the browser has not blocked the microphone permission.
• Step 5: Configure webphone preferences such as default caller ID, call recording consent behaviour, and notification sounds so each agent has a personalised calling experience that matches their workflow.

Setup checklist

• Open the Voice Setup page from the portal sidebar and navigate to the Browser Phone section to check that your workspace has browser calling enabled and available SIP seats assigned to users.
• For each team member who needs browser calling, confirm their user profile has a browser seat allocated and that their extension number is set. The extension acts as their internal reachable line.
• Have each user open the Webphone tab in the portal. The browser will prompt for microphone permission — grant access so the webphone can transmit and receive audio during calls.
• Test the webphone by placing a call to a test number. If audio does not work, check that no other application is using the microphone, that the correct audio device is selected, and that the browser has not blocked the microphone permission.
• Configure webphone preferences such as default caller ID, call recording consent behaviour, and notification sounds so each agent has a personalised calling experience that matches their workflow.

Operational follow-up

After you complete this flow, confirm the live experience from both the agent and customer side so ownership, routing, permissions, and reporting all match what the workspace expects.

If your team is rolling this out across multiple users, queues, or phone numbers, pair this article with the broader knowledge base and the relevant routing or numbers guides to keep deployment consistent.

• What is the CallOrbit Knowledge Base for? — It is the public help hub for how CallOrbit works, covering numbers, webphone setup, SIP, extensions, routing, users, roles, and billing basics.
• Can customers read this without signing in? — Yes. The Knowledge Base now lives on a public route so customers can read setup guidance before or after they enter the portal.
• Does the portal still have its own Knowledge Base page? — No. The signed-in portal navigation no longer carries a separate Knowledge Base page, and the old portal path now redirects to this public version.
• What is VoIP and how does it work? — VoIP (Voice over Internet Protocol) converts analogue voice signals into digital packets and transmits them over IP networks. Unlike traditional PSTN phone lines that require dedicated copper wiring per line, VoIP calls use your existing internet connection, which makes them cheaper, more flexible, and easier to scale.
• What is SIP trunking? — SIP trunking is a virtual connection that replaces traditional analogue phone lines or PRI circuits. A SIP trunk carries multiple concurrent voice channels over a single IP connection to your PBX or phone system, eliminating per-line hardware costs and monthly line rental fees.
• What is the difference between hosted PBX and cloud PBX? — Hosted PBX runs on dedicated virtual infrastructure managed by a provider, while cloud PBX uses shared multi-tenant cloud infrastructure. Hosted PBX suits organisations needing custom configuration and predictable pricing. Cloud PBX is better for instant scalability and per-user monthly billing.
• What is a DID number? — A DID (Direct Inward Dialling) number is a virtual phone number that routes directly to a specific extension, IVR menu, queue, or user within a phone system without an operator. DIDs decouple the phone number from the physical phone line, so you can have hundreds of numbers routed through a single SIP trunk.
• What are G.711, Opus, and G.729 codecs used for? — These are VoIP codecs that convert voice into digital data. G.711 uses 64 Kbps for toll-grade quality and is the PSTN standard. Opus uses 6-510 Kbps and adjusts to network conditions. G.729 uses 8 Kbps for bandwidth-constrained links. The right codec depends on your available bandwidth and call quality requirements.

Related CallOrbit guides