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WebRTC enables peer-to-peer audio, video, and data streaming directly in browsers without plugins. The core protocol is open and free, but production-ready implementations require significant infrastructure: STUN servers for NAT traversal, TURN servers for relay fallback, signaling servers for session coordination, and media servers for recording or mixing.

Building this yourself means maintaining globally distributed infrastructure, handling edge cases across browser versions, and scaling capacity on demand. For teams focused on customer-facing applications: contact centers, telehealth platforms, or Voice AI agents. That overhead delays time to market.

Managed WebRTC providers bundle these components into APIs and SDKs, letting you integrate real-time communication in days rather than months. The tradeoff is cost versus control, which makes provider selection critical.

Core evaluation criteria

When comparing WebRTC providers, focus on these six areas:

APIs and SDKs. Look for mature client libraries across web, iOS, and Android. The SDK should abstract connection management, obeying retry logic and handling browser quirks so your team writes application code instead of plumbing. Telnyx Voice SDKs support JavaScript, React Native, iOS, and Android with consistent interfaces across platforms.

TURN and STUN infrastructure. TURN servers relay media when direct peer connections fail—which happens frequently behind corporate firewalls. Providers should operate geographically distributed TURN nodes to minimize latency. Ask where their points of presence are located and whether media stays on a private backbone or traverses the public internet.

Signaling and call control. Signaling establishes and tears down sessions. Some providers offer only raw WebSocket signaling; others include higher-level call control APIs for features like hold, transfer, and conferencing. If you're bridging WebRTC to SIP or the PSTN, verify the provider supports that natively.

Recording and storage. Recording real-time sessions requires media server infrastructure. Evaluate whether recordings are stored regionally for compliance, how long they're retained, and what the per-GB storage costs look like at scale.

Security and compliance. WebRTC encrypts media with SRTP by default, but enterprise deployments need more: TLS for signaling, SOC 2 certification, GDPR compliance, and potentially HIPAA BAAs for healthcare. Confirm the provider's certifications match your requirements.

Global reach. Latency directly impacts call quality. Providers with points of presence in your users' regions deliver better experiences. North America accounts for 36% of WebRTC market activity, but if you serve APAC or LATAM, verify the provider has local infrastructure.

Pricing comparison

WebRTC pricing models vary significantly. Some charge per participant minute, others per room, and many add fees for recording, storage, or PSTN connectivity. Here's how leading providers structure costs:

Watch for hidden costs: egress fees, support tier pricing, and overage charges can inflate bills quickly. Agora's support plans, for example, range from free to $4,900/month depending on response time SLAs.

Security that scales with your requirements

Real-time communication carries sensitive data. Beyond WebRTC's built-in SRTP encryption, verify your provider offers:

• TLS 1.2+ for all signaling traffic
• SOC 2 Type II certification
• GDPR-compliant data handling and regional storage
• HIPAA BAAs for healthcare applications
• IP allowlisting and authentication controls

Telnyx maintains SOC 2 compliance, GDPR readiness, and HIPAA eligibility across its carrier-grade network, with media anchored on a private IP backbone rather than the public internet.

Latency and global infrastructure

Physics constrains real-time communication. The farther data travels, the higher the latency, and latency above 300ms degrades conversational flow. Around 87% of remote and hybrid workers prefer video conferencing for engagement, which means poor quality directly impacts user satisfaction.

Providers solve this by distributing infrastructure globally. Telnyx colocates TURN and media servers with telephony points of presence and GPUs, enabling sub-200ms round trips and HD audio quality. This architecture also supports Voice AI applications where low latency is critical for natural conversation.

When evaluating providers, request their PoP locations and test latency from your users' primary regions. A provider strong in North America may underperform in Southeast Asia if they lack local infrastructure.

Connecting WebRTC to the PSTN

Many business communication use cases require bridging browser-based calls to traditional phone networks. Contact centers need agents using web interfaces to reach customers on mobile phones. Sales teams need click-to-call from CRM dashboards.

Not all WebRTC providers include PSTN connectivity. Some require you to integrate a separate telephony provider, adding complexity and potential points of failure. Telnyx operates as a licensed telecom provider in 30+ markets with PSTN calling in 100+ countries, so you can provision numbers, configure SIP trunks, and build Voice AI agents on the same platform.

Build real-time communication faster with Telnyx

Choosing a WebRTC provider comes down to matching capabilities to your requirements. If you need browser-based video conferencing with minimal PSTN interaction, a pure-play video SDK might suffice. But if you're building contact center applications, Voice AI experiences, or anything that bridges web and telephony, you need a platform that unifies both.

Telnyx WebRTC delivers carrier-grade voice and video with native PSTN connectivity, global number provisioning, call recording, and compliance support, all backed by 24/7 engineering support at no extra cost. Compare Telnyx to Twilio or contact our team to discuss your architecture.

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