Voice Calls via Messengers and Video Platforms Rise 11% in 2025
Voice communication is decoupling from the Public Switched Telephone Network (PSTN). According to the 2025 annual report from Technika (TK), users conducted approximately 19.6 billion voice calls via messenger and video conferencing platforms in 2025, marking an 11% increase over the previous year. This shift signals a systemic migration from traditional telephony to Over-the-Top (OTT) voice services.
- Traffic Spike: 19.6 billion OTT voice calls in 2025, an 11% YoY increase.
- Infrastructure Shift: Rapid decline in traditional phone number reliance in favor of data-driven VoIP protocols.
- Enterprise Impact: Increased demand for SIP trunking and WebRTC optimization to handle asynchronous voice loads.
The data indicates a fundamental architectural pivot. We aren’t just seeing a change in user preference; we are witnessing the erosion of the legacy circuit-switched network. For CTOs, this isn’t about “apps”—it’s about the transition to a fully packet-switched environment where voice is simply another data stream. This migration introduces significant challenges in Quality of Service (QoS) and end-to-end encryption (E2EE) at scale.
The Transition from PSTN to WebRTC and SIP
The 11% growth reported by TK reflects the broader adoption of WebRTC (Web Real-Time Communication) and Session Initiation Protocol (SIP). Unlike traditional telephony, which relies on a dedicated circuit, these calls operate over UDP (User Datagram Protocol) to minimize latency. However, as volume hits the 19.6 billion mark, the “jitter” and packet loss associated with public internet routing become critical bottlenecks.
To maintain voice clarity, enterprises are moving away from basic VoIP and toward sophisticated software-defined networking (SDN). This requires strict SOC 2 compliance to ensure that voice data, which is often more sensitive than text, is not intercepted during the handshake process. Companies struggling with this transition are increasingly engaging [Managed Service Providers] to implement dedicated voice VLANs and prioritize voice traffic over standard data packets.
From a developer perspective, the shift toward messenger-based calling often involves integrating Third-Party APIs to handle signaling. A typical implementation for triggering a voice session via a REST API might look like this:
curl -X POST https://api.voice-provider.com/v1/calls
-H "Authorization: Bearer YOUR_ACCESS_TOKEN"
-H "Content-Type: application/json"
-d '{
"to": "+1234567890",
"from": "Company_SIP_Gateway",
"options": {
"codec": "opus",
"encryption": "sri-aes-256",
"latency_threshold": "150ms"
}
}'
Comparative Analysis: OTT Voice vs. Legacy Telephony
The divergence between traditional phone numbers and voice-over-IP is stark when analyzing the underlying tech stack. The following table breaks down the operational differences driving the 2025 surge.
| Feature | Legacy PSTN | OTT / Messenger Voice |
|---|---|---|
| Transport | Circuit-Switched | Packet-Switched (IP) |
| Addressing | E.164 Phone Numbers | Account IDs / User Handles |
| Protocol | SS7 / SIGTRAN | SIP / WebRTC / RTP |
| Scaling | Hardware Dependent | Containerized / Kubernetes |
| Security | Centralized Trust | End-to-End Encryption (E2EE) |
Cybersecurity Risks in the VoIP Explosion
The move to 19.6 billion calls creates a massive attack surface. Traditional telephony was closed; OTT is open. The primary risk is now “SIP vishing” and Man-in-the-Middle (MitM) attacks targeting the signaling layer. When voice calls move to the cloud, the risk of session hijacking increases if the implementation lacks robust TLS (Transport Layer Security) for the signaling plane.
Moreover, the reliance on third-party messenger platforms means that enterprise data is often subject to the security posture of the platform provider. This has led to a surge in demand for [Cybersecurity Auditors] who can verify that voice data is encrypted in transit and that the keys are not stored on the provider’s servers. Without rigorous penetration testing, a “voice-first” strategy can become a backdoor for corporate espionage.
According to documentation found on RFC Editor, the evolution of the Real-time Transport Protocol (RTP) is essential to mitigate these vulnerabilities. Developers must ensure that SRTP (Secure RTP) is mandated for all voice streams to prevent eavesdropping.
Infrastructure Bottlenecks and the Path Forward
The 11% year-over-year increase puts immense pressure on NPU (Neural Processing Unit) integration within mobile chipsets. Modern voice calls aren’t just audio; they include real-time noise cancellation and AI-driven transcription. This requires dedicated hardware acceleration to prevent thermal throttling during extended calls.

As organizations scale their communication stacks, the move toward containerization via Kubernetes allows for the dynamic scaling of voice gateways based on real-time demand. This prevents the “dropped call” phenomenon common in legacy PBX systems. For firms unable to manage this complexity in-house, [Software Development Agencies] specializing in cloud-native communications are becoming essential for migrating legacy voice assets to the cloud.
The trajectory is clear: the phone number is becoming a legacy identifier, a mere relic of the 20th century. Voice is now a feature of the identity layer, not the hardware layer. As we move toward 2027, the integration of voice into decentralized identity (DID) frameworks will likely be the next frontier, further decoupling communication from the telecom giants.
Disclaimer: The technical analyses and security protocols detailed in this article are for informational purposes only. Always consult with certified IT and cybersecurity professionals before altering enterprise networks or handling sensitive data.