WebRTC/Test Plan
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Purpose
This document intends to provide a high level overview of the testing that's needed for testing the entire stack of the WebRTC feature set.
Feature Ownership
| Feature | QA Lead |
| getUserMedia | Jason Smith |
| Peer Connection | Jason Smith |
| Data Channel | Jason Smith |
Other Resources
| Resource | Lead |
| Security | Christoph Diehl |
| Automation Development | Henrik Skupin |
Testing Scope
In Scope
Functionality
- getUserMedia and Local Media Streams DOM functionality
- End-to-end integration with real cameras/mics across operating systems with getUserMedia
- Peer Connection, Session Description, and Ice Candidate DOM API functionality
- Data Channel DOM API and it's integration with global Peer Connection DOM API
- End-to-end local and remote Peer Connection handshaking, ice candidates, data connections, etc.
- End-to-end local and remote Data Channel integration with Peer Connection and sending of data
Software Qualities
- Security - Fuzzing SDP, the DOM APIs, ordering of handshake, etc
- Performance - Time for round-trip calls, video/audio response time upon input, etc
- Scalability - Scaling to large network topologies - do we hold together without functionality/performance issues?
- Availability & Resilience - Downtime during & after handshake, lame network handling, etc
- Location - location of peers - close? Different country?