WebRTC vs SIP for Voice AI: Lessons from Production Deployments
The article discusses the pros and cons of using WebRTC vs SIP protocols for enterprise Voice AI deployments, based on the author's experience managing such projects.
Why it matters
Understanding the trade-offs between WebRTC and SIP is crucial for enterprises deploying Voice AI solutions, as the choice of protocol can significantly impact the complexity and cost of the deployment.
Key Points
- 1SIP is required for calls from real phone numbers (mobile/landline) as the PSTN speaks SIP
- 2WebRTC is better for browser-based voice interfaces (click-to-call, web widgets) due to Opus codec, no carrier cost, and automatic NAT traversal
- 3Most enterprise deployments need both WebRTC and SIP, bridged via a Session Border Controller (SBC), which adds complexity
Details
The author shares their experience in choosing the right protocol for Voice AI deployments. They highlight that SIP is necessary when users call from real phone numbers, as the PSTN (Public Switched Telephone Network) primarily uses SIP. On the other hand, WebRTC is a better fit for browser-based voice interfaces due to its Opus codec, lack of carrier costs, and automatic NAT traversal. However, most enterprise deployments end up requiring both WebRTC and SIP, which need to be bridged using a Session Border Controller (SBC) - a complexity that is often underestimated. The author also shares a cautionary tale of choosing WebRTC initially, only to discover the client's contact center platform only accepted SIP, leading to additional development work to build a gateway.
No comments yet
Be the first to comment