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内容概要:本文介绍了一种用于WebRTC应用和服务自动化测试的新方法及其相关工具。该方法基于黑盒端到端测试,捕获QoS(服务质量)和QoE(用户体验质量)指标。通过自动化的容器化云环境对浏览器客户端进行仪器化,从而模拟系统的真实负载情况。该工具能够生成统计上可靠的服务质量和媒体体验指标,为开发人员提供简单易用的测试断言接口。此外,还支持在不同网络场景下测试系统的性能和兼容性。 适合人群:从事Web开发、实时通信、多媒体技术和WebRTC应用的软件开发者、测试工程师和研究人员。 使用场景及目标:①评估WebRTC应用的功能正确性和非功能性属性;②模拟真实世界的网络条件,确保应用程序在网络拓扑复杂的情况下仍能正常运行;③优化应用的性能和用户满意度;④在开发阶段早期发现并解决问题。 其他说明:作者团队已将该方法应用于开源项目Kurento,验证了其在大规模和复杂WebRTC系统测试中的有效性。未来的工作计划包括增强网络流量控制组件,支持网络切换等功能。
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36 IEEE Communications Standards Magazine • June 2017
2471-2825/17/$25.00 © 2017 IEEE
Abstract
WebRTC comprises a set of novel technolo-
gies and standards that provide Real-Time Com-
munication on Web browsers. WebRTC makes
simple the embedding of voice and video com-
munications in all types of applications. However,
releasing those applications to production is still
very challenging due to the complexity of their
testing. Validating a WebRTC service requires
assessing many functional (e.g. signaling logic,
media connectivity, etc.) and non-functional (e.g.
quality of experience, interoperability, scalability,
etc.) properties on large, complex, distributed and
heterogeneous systems that spawn across client
devices, networks and cloud infrastructures. In
this article, we present a novel methodology and
an associated tool for doing it at scale and in an
automated way. Our strategy is based on a black-
box end-to-end approach through which we use
an automated containerized cloud environment
for instrumenting Web browser clients, which
benchmark the SUT (system under test), and fake
clients, that load it. Through these benchmarks,
we obtain, in a reliable and statistically signifi-
cant way, both network-dependent QoS (Quality
of Service) metrics and media-dependent QoE
(Quality of Experience) indicators. These are fed,
at a second stage, to a number of testing asser-
tions that validate the appropriateness of the func-
tional and non-functional properties of the SUT
under controlled and configurable load and fail
conditions. To finish, we illustrate our experiences
using such tool and methodology in the context
of the Kurento open source software project and
conclude that they are suitable for validating large
and complex WebRTC systems at scale.
Introduction
Web applications and services are among the
most popular ones in today’s Internet. Due to
this, Web developers are subject to an increas-
ing demand for delivering more complex, scal-
able and reliable applications in less and less time.
The most limiting factor for satisfying this demand
is software verification and validation that may
account for more than 70 percent of the total
development effort [1]. Because of this, during
the last few years, we have witnessed an explo-
sion of DevOps methodologies and tools that aim
to simplify the validation process. Thanks to these,
nowadays Web developers are able to automate
unit and system tests. These tests typically com-
prise one or several probes that query the system
under test (SUT), or any of its components, and
gather simple outcomes as a result. After that,
a set of testing assertions evaluate the system
correctness by comparing such outcomes with
pre-set or model-inferred values. An assertion (or
predicate) is a mechanism for determining wheth-
er a test has passed or failed. It involves compar-
ing an outcome of the SUT with the expected
value.
The popularization of multimedia technologies
among Web developers and, very particularly, the
emergence of WebRTC as a built-in feature of
Web browsers, is significantly increasing the com-
plexity of the testing. WebRTC comprises a set
of standards designed for embedding RTC (Real-
Time Communication) capabilities as part of Web
browsers’ APIs [2]. A recent report predicts that
with Apple and Microsoft supporting WebRTC in
their browsers, there might be 7 billion devices
compliant with WebRTC by 2020 [3].
WebRTC applications cannot be tested using
the usual simple comparison-based assertion. For
example, validating the functional correctness
of a WebRTC application requires the ability to
evaluate aspects such as media connectivity (e.g.
whether the media bits are being sent end-to-end)
or media continuity (e.g. whether the media is
decodable). Assessing non-functional properties
is even more complex. The real-time nature of
WebRTC traffic causes QoS (Quality of Service)
parameters such as network latency, network
jitter or packet loss to affect significantly, and in
non-trivial ways, the end-user’s QoE (Quality of
Experience). Measuring accurately such QoE and
enabling assertions to automate their validation is
a huge challenge.
As an additional complication, testing WebRTC
applications is challenging due to the heteroge-
neous nature of the networks to be traversed in
order to enable P2P communication. Some prob-
lems include clients consuming media behind a
NAT or a firewall. As a result, WebRTC services
involve complex, distributed and heterogeneous
network topologies where failures or inefficiencies
may prevent the service from providing a success-
ful user experience.
All in all, creating reliable and automated tests
for WebRTC applications, services and infrastruc-
tures is a task that cannot be performed in a
simple way based on commonly available state-
of-the-art DevOps tools and methodologies. In
this article, we contribute a solution to this prob-
lem, proposing both a strategy and technologi-
cal toolbox that are suitable for the automated
Boni García, Francisco Gortázar, Luis López-Fernández, Micael Gallego, and Miguel París
REALTIME COMMUNICATIONS IN THE WEB
The authors are with the Universidad Rey Juan Carlos.
WebRTC Testing:
Challenges and Practical Solutions
Digital Object Identifier:
10.1109/MCOMSTD.2017.1700005
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