Overview

The WebRTC conformance tester is designed to support the testing of the signaling component of WebRTC solutions. WebRTC does not define call control signaling leading to a variety of vendor specific solutions.

The flexibility of the Protocol Engine allows rapid prototyping of these protocols whether they are based on SIP, JSON, XML, REST/HTTP or other methods.

Multi-protocol support allows testing of complex scenarios including authorization APIs, interaction with standard based network elements based on SIP or other VoIP based products, AAA and EPC infrastructure.

The Protocol Engine ICE stack is available for testing media features including codec negotiation, RCTP Mux, DTLS, SDES

Features

  • Enabled rapid development of test cases for vendor specific call control protocols

  • Supports HTTP/REST, JSON, XML based protocols

  • Pass/fail including plain English diagnostic reason

  • Valid/Invalid testing

  • Built-in reporting and test suite documentation generation

  • Message templates reduce code-size, duplication and errors

  • Input constraints for simplified testing

  • Easy to configure

  • Test Planner to define test sequences

  • Supports pre-amble/post-amble common setup/teardown sequences

  • UDP, TCP, WebSocket transport

  • socket.io emulation

  • Compatible with mature SIP Conformance Test Suite

  • Suitable for Development and QA test lab environments, verifying protocol compliance, negative and robustness testing, Regression testing and Reproducing customer issues in the field

WebRTC Conformance

What can it do?

The WebRTC Conformance solution is capable of simulating and testing several devices individually or in parallel:

It can simulate

UEs

Browsers

Gateways

SBC

ESBC

It can test

Gateways

SBC

ESBC

Specifications

Available technology stacks

IETF RFC6455 - The WebSocket Protocol

IETF RFC3261 - SIP

IETF RFC4347 - DTLS

IETF RFC2327 - SDP

IETF RFC2617 - HTTP Authentication

IETF RFC2976 - SIP INFO

IETF RFC3264 - SDP Offer Answer

IETF RFC3266 - SDP IPv6 support

IETF RFC3311 - SIP UPDATE

IETF RFC3428 - SIP MESSAGE

IETF RFC3516 - IPv6

IETF RFC3550 - RTP / RTCP

IETF RFC7159 - The JavaScript Object Notation (JSON) Data Interchange Format

IETF RFC2616 - Hypertext Transfer Protocol -- HTTP/1.1

IETF draft-ietf-sipcore-sip-websocket

draft-perkins-avt-rtp-and-rtcp-mux

socket.io emulation

Audio

All major audio codecs including OPUS

Video

All major video codecs including VP8, H.263, H.264

Network Layer Capabilities and Security

IPv4, IPv6

UDP, TCP

WS, WSS

DTLS

SRTP

STUN, TURN, ICE

Additional Details

Options

WebRTC Conformance Tester

Related Solutions

WebRTC Load Tester

Contact

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Specifications subject to change at any time