WebRTC/Test Plan

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?

Out of Scope

• Actual third-party WebRTC apps using the API (outside of Mozilla-specific needs)
• Android and B2G for now, as we're targeting desktop primarily first
• Other unsupported getUserMedia, Local Media Streams, Peer Connections, and Data Channels DOM API functions we are not targeting for v1 release