WebRTC Global Summit, Joachim Stegmann Deutsche Telekom, Opus versus VoLTE
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Transcript of WebRTC Global Summit, Joachim Stegmann Deutsche Telekom, Opus versus VoLTE
WebRTC in a real Mobile Scenario
Dr. Joachim Stegmann, Telekom Innovation Laboratories
WebRTC in a real Mobile Scenario
2
01 Introduction
02 Unlocking new services with WebRTC
03 WebRTC based services: What about quality?
04 Audio quality tests
05 Conclusion
WebRTC in a real Mobile Scenario
INTRODUCTION
WebRTC provides a lot of benefits to simplify communication outstanding HD user experience in desktop use @home, @office on any device with a browser that is supported
Trend is more traffic on mobile than desktop.
Need to use WebRTC in mobile environments is growing significantly In private and corporate communication: everywhere, easier and cheaper e.g. in Customer Service: Unlocking new service ideas (video support, video chat substitutes in-person interaction)
In OTT to support the change from circuit-switched call to IP/WebRTC
3
Thanks to stable steady
conditions!
What about outside the
ideal world?
Challenges emerge from the mobile environment’s instability: Changes in connectivity and network load, interferences etc.
UNLOCKING NEW SERVICES WITH WebRTC
4
Connecting TELCOs AND WEB
Managed OTTCore network + cloud servicesOptimized cost structure
Disruptive services
ENTERPRISE COMMUNICATION
Customer service +++ Sales & support +++ Pricing & license models +++ M2M +++ Industry 4.0 +++ Content delivery networks
Efficiency and cost savings
Makes collaboration easier, cheaper and more flexible!
Extend Telekom core offerings e.g. VoLTE
WebRTC Conferencing
http://www.immmr.com
UNLOCKING NEW SERVICES WITH WebRTCTypical Use case IN Customer Service
REMOTE VIDEO HELP
Initiates video stream from customer’s smartphone (back cam) to agent’s desktop
Live-video or single shot alternatively
Allows drawing markers by mouse pointer
Requires no installation on the agent or customer side
– pure browser-based solution (WebRTC)
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UNLOCKING NEW SERVICES WITH WebRTCENTERPRISE COMMUNICATION
WEBRTC CONFERENCING
Browser-based video call conferencing
Access from PSTN to allow participation in web-conference
Text chat
Desktop share
Whiteboard functionality with import / export
File exchange, sending files via drag & drop
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…immmr…
Immmr – A Managed ott operator
Virtualized primary number
Make / receive calls on any device
WebRTC, Wi-Fi Calling
/ GSM fallback
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WebRTC provides excellent audio quality under ideal, steady state conditions.
What is the situation under mobile conditions?
How strong is the impact of fluctuating field strength, network transitions etc.?
WebRTC based Services – What about Quality?
Audio Quality TestsTest Location And Routes
MOBILITY TESTCASES (2 hours per test case)
performed on a pre-defined route in Duesseldorf that contains many handoversand RAT changes
STATIONARY TESTING (20 min. per test case)
performed in P3 Communications Aachen office (Telekom indoor coverageinfrastructure) in static coverage condition depending on the required technology
10
Evaluating voice quality of NG networks
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Perceptual Objective Listening Quality Analysis
Global standard for benchmarking voice quality of fixed, mobile and IP-based networks
Applied for classical band-limited telephony VoIP, HD Voice, 3G and 4G/VoLTE networks
© OPTICOM GmbH 2010
POLQA is suited to evaluate and monitor voice quality of NG networks.
Su
per
-wid
e-b
and
(SW
B)
14kH
z
The different bandwidths are mapped on the MOS scale 1 - 5.
WebRTC under stationary conditionS
On-net Off-net (Landline) Off-net (Mobile)
WebRTC (Wi-Fi, Opus, 40kbps) MOS 4,1 MOS 2,9 MOS 2,9
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REFERENCE
WebRTC on-net calls show significantly better audio quality than circuit-switched (CS) HD audio calls.
WebRTC off-net calls to PSTN (landline and mobile) show audio quality as good as narrow-band CS calls.
Influence of transcoding seems to be negligible.
AMR wide-band (HD voice) AMR narrow-band
Circuit-switched MOS 3,8 MOS 2,8
COMPARISON of WebRTC VS. CS Calls on Mobility
WebRTC on-net calls with
better audio quality than CS calls.
HD audio, once switched off for transition from 4G/3G to 2G, will never be
switched on again during a CS call.
WebRTC calls benefit from higher data rates in 3G/4G
after coming back from 2G.
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4
4
4
WebRTC1
WebRTC2
CS
MOS
„No audio transmitting“ cases are often caused by bearer changes between 4G/3G
COMPARISON of WebRTC VS. CS Calls on Mobility
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4
4
4
Significant problems of
WebRTC with bearer changes between 2G/3G/4G
WebRTC1
WebRTC2
CS
MOS
ConSideration of used Bitrates for WebRTC
Higher bitrates for WebRTC
result in better audio quality by decreasing stability.
Bitrate of 12 kbps shows already sufficient audio quality.
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4
4
4
MOS
WebRTC1
12 kbps
WebRTC1
20 kbps
WebRTC1
40 kbps
WebRTC in a real Mobile ScenarioConclusions
PREDICTING NETWORK CONDITONS
To avoid instabilities by high bitrates and bearer changes try to predict the network conditions to be expected
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ADAPTING BITRATES
Under mobile condition already 12 kbps with good audio quality and better stability
4G and 40 kbps in case of stationary conditions
HANDLING BEARER CHANGES
Issues observed for bearer changes between 2G, 3G and 4G under mobile condition.
Optimization of bearer changes handling on WebRTC platform side is elementary!
WebRTC on-net calls offer outstanding quality under stationary conditions. In mobile use possible instabilities caused by changing conditions can be minimized considering the influence quantities.
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WE SHAPE THE FUTURE
Telekom Innovation Laboratories