StartACall /Enterprise
DTLS-SRTP ENCRYPTED

Enterprise-Grade
WebRTC Infrastructure

Secure, scalable, and integrated voice communications for the modern enterprise. Replace legacy SIP trunks with browser-native technology.

View Architecture
constconnection=newRTCPeerConnection(config);
// Mandatory Encryption
connection.addTransceiver('audio',{ dtls: true });
// Global Low Latency
await connection.setLocalDescription(offer);
> Connection established: 24ms latency

Uncompromising Security

WebRTC mandates encryption. We utilize DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-time Transport Protocol) to ensure no voice data can be intercepted or tampered with in transit.

  • End-to-End Encryption
  • SOC2 Compliant Infrastructure
  • No PII Storage

Infinite Scalability

Forget hardware limitations. Our cloud-native architecture scales elastically. Whether you have 10 concurrent calls or 10,000, our global mesh of media servers handles the load automatically.

  • 99.99% Uptime SLA
  • Global Edge Network
  • Auto-Scaling Clusters

Seamless Integration

StartACall isn't just a tool; it's a platform. Integrate voice capabilities directly into your CRM, Helpdesk, or custom internal tools using our comprehensive APIs and Webhooks.

  • REST API Access
  • Real-time Webhooks
  • Custom CRM Widgets

Global Low-Latency Network

We don't just route calls; we optimize the path. Our intelligent routing engine selects the nearest media server to the user, minimizing latency and jitter.

12 Regions
Data Centers
< 50ms
Average Latency
System Status: Operational

Modernize Your Voice Infrastructure

Move away from hardware PBX and legacy SIP. Embrace the future of browser-based communication.

Deploying WebRTC Calling Across an Enterprise Network

In short

Adopting browser calling at enterprise scale is less about the protocol and more about your environment: managed browsers, firewall policy, audio hardware, and rollout sequencing. This section walks through the network prerequisites, browser policy settings, and evaluation criteria that determine whether a WebRTC deployment lands smoothly or generates helpdesk tickets.

Network prerequisites and firewall policy

WebRTC media prefers UDP, so the first check is whether your egress firewall permits outbound UDP to the provider's media ranges. Where security policy forbids that, calls should still connect by relaying over TCP port 443 through TURN, at a small latency cost. Confirm the provider supports this fallback before piloting on a locked-down network.

Quality of service matters more than raw bandwidth. A single voice stream needs well under 100 kbps, but it needs consistent delivery. Tagging voice traffic with DSCP EF on managed networks, or simply keeping large backups off the same egress during business hours, prevents most jitter complaints.

Proxy configuration deserves its own line item. Deep packet inspection appliances that intercept TLS can interfere with media negotiation, so most deployments exempt the calling service's domains from inspection, the same exception commonly granted to video conferencing platforms.

Managing browsers and microphone permissions at scale

In a managed fleet, browser policy decides the user experience. Chrome, Edge, and Firefox all accept enterprise policies that pre-approve microphone access for specific origins, so staff never see a permission prompt they might mistakenly deny. Setting this once in group policy or your MDM removes the single most common onboarding failure.

Also standardize on current browser versions. WebRTC implementations improve continuously, and echo cancellation or device switching bugs that plague an outdated build are often already fixed upstream. Chrome, Safari, Firefox, and Edge all ship mature WebRTC stacks today, so version currency matters more than browser choice.

Audio hardware is the quality variable you control

Once the network is sound, perceived call quality tracks the headset more than anything else. Laptop microphones pick up keyboard noise and room echo that no codec can fully remove. Issuing wired USB headsets to calling-heavy roles is the cheapest quality upgrade available, and it sidesteps Bluetooth pairing issues on shared desks.

Run a short pilot with the actual hardware mix your staff use before the full rollout. Ten pilot users across your real office networks will surface device and VPN interactions that a lab test never will.

Sequencing the rollout and measuring success

Roll out by team rather than by site, starting with a group that makes frequent outbound calls and can give structured feedback. Because a browser-based service like StartACall needs no installation and bills per minute with no subscription, a pilot carries little sunk cost: the trial team simply signs in from their existing browsers, and outbound calling needs no phone number provisioning at all.

Define success metrics before you start. Call setup success rate, user-reported audio quality, and helpdesk ticket volume in the first month tell you more than any vendor benchmark.

Document the escalation path early too. When a call fails, the user should know whether to retry, switch networks, or file a ticket, and the helpdesk should know which diagnostics to collect, browser version, network location, and time of failure, before contacting the vendor.

Frequently asked questions

What ports does WebRTC need open on a corporate firewall?+

Ideally outbound UDP for direct media, but when policy forbids it, WebRTC relays media through TURN over TCP port 443, which firewalls already allow for HTTPS. Confirm your provider supports the 443 fallback for strict networks.

Can IT pre-approve microphone access so users are not prompted?+

Yes. Chrome, Edge, and Firefox enterprise policies can whitelist specific origins for microphone access through group policy or MDM. This removes accidental permission denials, which are the most common cause of failed first calls in managed environments.

How much bandwidth does browser calling use per user?+

A single voice call typically consumes well under 100 kbps in each direction. Consistency matters more than volume, so concurrent heavy transfers on the same link are a bigger risk to call quality than total connection size.

Does WebRTC calling work over a corporate VPN?+

Usually, but full-tunnel VPNs add latency and can degrade audio. Where policy allows, split tunneling for the calling service's traffic gives better quality. Test the VPN path during the pilot rather than discovering issues at rollout.

How should we pilot a browser calling service before committing?+

Pick one outbound-heavy team, use their real hardware and networks, and measure call setup success and reported quality for a few weeks. Pay-as-you-go services like StartACall make this cheap since there is no contract, installation, or per-seat license to unwind.

Last reviewed June 2026Reviewed by the StartACall calling teamDialing rules cross checked against ITU international dialing procedures
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