WEBPHONE • ARCHITECTURE
How WebPhone reaches a SIP PBX
The browser WebPhone does not connect directly to the SIP server or IP-PBX. It communicates with VaxServerWebRTC, which makes the call connection possible with the existing SIP platform.
This separation lets the website use browser-native WebRTC while the call center keeps its established SIP extensions, queues, routing rules and agent endpoints.
Browser WebPhone
Website visitor and WebRTC media
VaxServerWebRTC
WebRTC-to-SIP connection layer
SIP Server or IP-PBX
Call routing and agent destinations
REQUIRED • COMPONENTS
The three required components
HTML and JavaScript WebPhone
Provides the website call interface, browser controls and real-time media session.
VaxServerWebRTC
Receives the WebPhone connection and communicates with the SIP server or IP-PBX.
SIP server or IP-PBX
Applies dial plans and routes the call to a queue, extension, group or agent.
REAL-TIME • CALL • FLOW
Call flow from browser to agent
- 1
The visitor starts a call
The visitor selects the website call control and grants microphone permission when requested by the browser.
- 2
WebPhone connects to VaxServerWebRTC
The browser establishes its WebRTC session with the configured VaxServerWebRTC application.
- 3
VaxServerWebRTC connects with SIP
VaxServerWebRTC uses the configured SIP account or routing information to place the call through the SIP server or IP-PBX.
- 4
The PBX selects the destination
The dial plan routes the call to the required call center queue, extension, hunt group or SIP endpoint.
- 5
The visitor and agent communicate
VaxServerWebRTC maintains the media connection between the browser session and the SIP call until either side disconnects.
PBX • ROUTING
PBX routing and call destinations
Website calls enter the normal SIP routing environment. The PBX can apply business hours, queue priorities, overflow rules, recorded announcements and unavailable destinations before selecting an endpoint.
Support queue
Route visitors to available service agents.
Sales extension
Connect high-intent visitors with a sales team.
Customer portal
Use account context to select the appropriate destination.
Automated service
Route selected calls to a configured SIP-based voice service.
DEPLOYMENT • CHECKLIST
Deployment requirements
A production installation should verify the complete path instead of testing the browser and PBX separately.
- Secure website: Serve the WebPhone from HTTPS and request microphone access only when needed.
- VaxServerWebRTC: Configure the server component, public reachability and the intended SIP account.
- SIP routing: Define the extension, queue or dial-plan destination for website calls.
- Network: Verify firewall, NAT and media behavior in the actual hosting environment.
- Audio: Test supported codecs, two-way media, silence, mute and disconnect behavior.
- Capacity: Measure simultaneous calls and define a response when no agent or route is available.
PRACTICAL • USE • CASES
Practical use cases
Website support
Give visitors a direct voice path to a technical or customer-service queue.
Sales click-to-call
Connect product and pricing pages with the appropriate sales extension.
Authenticated portals
Add calling to customer, partner or internal service portals.
Remote assistance
Combine browser voice with supported video, chat or screen-sharing workflows.
FREQUENTLY • ASKED • QUESTIONS
WebPhone, VaxServerWebRTC and SIP questions
Does the browser WebPhone connect directly to the SIP server or IP-PBX?
No. The WebPhone communicates with VaxServerWebRTC. VaxServerWebRTC provides the connection to the existing SIP server or IP-PBX.
What is VaxServerWebRTC?
VaxServerWebRTC is the server-side component between the browser WebPhone and SIP infrastructure. It handles the WebRTC side of the browser session and communicates with the SIP server or IP-PBX.
Does the existing SIP server need native WebRTC support?
No. VaxServerWebRTC handles browser WebRTC communication and connects it with the existing SIP platform, so WebRTC support is not required on the SIP server itself.
Do website visitors need to install a softphone?
No. Visitors use the WebPhone inside a supported browser and grant microphone permission when starting a voice call.
Where can website calls be routed?
The SIP server or IP-PBX can route calls to an extension, hunt group, support queue, sales team, call center agent or another configured SIP destination.
Can existing call center agents answer WebPhone calls?
Yes. Once the PBX routes the call, agents can answer through their existing supported SIP phones, softphones or call center endpoints.
What should be configured before deployment?
Configure a secure web origin, browser microphone access, VaxServerWebRTC, SIP credentials or routing, network access, supported codecs and the destination queue or extension.
What should be tested before publishing website calling?
Test supported browsers, microphone permission, call setup, two-way audio, NAT and firewall behavior, PBX routing, agent answer, call termination and concurrent calls.
WEBPHONE • SDK
Keep the browser layer separate from SIP routing
The browser WebPhone communicates with VaxServerWebRTC, and VaxServerWebRTC connects the call to the SIP server or IP-PBX. This clear separation lets the website add real-time calling without replacing the existing call center platform.
Explore WebPhone SDK