11/25/2013

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WebRTC Impact on Your Network

11/25/2013

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Tsahi Levent-Levi tsahil@bloggeek.me @tsahil

Bandwidth…

A picture may be worth a thousand words

BUT it takes a lot more bandwidth to store and send around

in Real time… with low latency

11/25/2013

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HikingArtist

Speakers

• Goran Stromberg – Senior Solution Manager

– Ericsson

• Frederic Huve Industry Principal and Chief Solution Architect Network Applications Communications & Media Solutions HP Enterprise Services

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WEBRTC IN THE WIRELESS CARRIER SPACE

Goran Stromberg

Senior Solution Manager

Ericsson

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Instant Messaging

Group Message

File Transfer

Video Share

Capabilities Discovery

CS Voice

HD Voice Call

Video Call

Voice/Video Conf.

Address and

Contact Bar

Augmented Web

Communication

MeTab

Communication

Dashboard

Extended Web communication Consumer, Enterprise, Wholesale

Building Front End

Web server

Web Application

Browser

Web App

Web API’s

WebRTC

Find & Connect

IMS

H2S

GW

Wireless

Device

PSTN

Device

Telecom Network

Helper

(ICE)

Server

HTTPS/REST

SRTP/SRTCP

EXISTING and FUTURE Telecom Services

EP

C

eN

B

Web server

Web Application

Browser

Web Application

Web API’s

WebRTC

Carrier PAAS Services

Find and Connect Find and Connect

ICE Media Services

SIP/HTTPS

Browser

Web Application

Web API’s

WebRTC

Find and Connect SIP/HTTPS

SRTP/SRTCP SRTP/SRTCP

PAAS Offering

Cloud Services to an Enterprise

CEBP

Retailer case

Expert

shop

Insurance case

Agent

Insurance Company

Remote patient

monitoring Case

Doctor

Hospital

Visualcom

External web Users

Enterprise

Examples of Wholesale…

Network Impact / Challenges

• QoS for WebRTC Traffic – Identify the optimum way of using LTE QoS for SRTP/SRTCP – Using transport DSCP marking and Ethernet pBit is desired

• Large Capacity Demands on LTE Access – Impact on Cell Planning and RF Optimization – Voice/video bandwidth + security overhead – May impact scheduling, timing, transport block sizing, etc… – Work in progress to improve LTE RRM efficiency for high video BW

• Transcoding to/from OPUS in IMS Media GWs – Optimum codec selection with the least fallback for all call types

Summary

• WebRTC fits very well into 4G Telecom Infrastructure

• WebRTC is enabler for many new business opportunities in the wireless carrier space

• Existing LTE QoS concept can be used for WebRTC apps – if desired

• WebRTC brings challenges with high bandwidth demands on LTE access. Work is in progress to meet those demands but spectrum availability will also be paramount

WEBRTC IMPACT ON ENTERPRISE & INTERNET IP NETWORKS

Frederic Huvé

Industry Principal and Chief Solution Architect Communications & Media Solutions/Network Applications

HP Enterprise Services

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Agenda

• Reminder for RT QoS over IP network

• WebRTC use-cases, and impact on IP networks

– Enterprise IP networks for enterprise users

– Internet (xDSL) for consumers

• Solutions to optimize WebRTC experience

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IP Network QoS for RT Comms

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Voice/Video QoS requirements • Data & Voice packet loss < 1% • Data& Voice jitter < 30 ms • Voice Latency < 150 ms • Voice RTT < 300 ms • No QoS/ToS change allowed

At high utilization time, packet loss (up to 4%) and

levels of jitter rise (up to 60ms)

Huge daily IP network utilization during working hours

*actually leading to capacity collapse

On October, 3rd

In sync with that,

occasionally causes latency

extensions from 100 msec

to 140 msec

and RTT extends from <200 msec

to > 300msec

Impact on RT communications

WebRTC use-cases

Consumer market • Web interactive voice & video sessions

(e.g. Click-To-Call)

• Healthcare, Education, Video surveillance use-cases, …

Enterprise market • Collaboration: voice/video/desktop and

application sharing

• Web interactive voice & video sessions

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WebRTC IP requirements

Enterprise scenario - powerful laptop

• Voice communication, Opus Voice codec ~32 kbps average (~20% variation observed: 28-40 Kbps)

• Video communication, VP8 video codec, Enterprise Network

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SVGA (960x720) 720p (1280x720)

Desktop sharing (video)

Medium video changes (12-13 fps) = 590 kbps

Medium video changes (12-13 fps) = 640 kbps

Camera Frequent video changes (25 fps) = 1.8 mps

Frequent video changes (25 fps) = 2 mps

WebRTC IP requirements (end)

Consumer scenario – Laptop/Tablet/Smartphones

• Voice communication, Opus speech codec ~32 kbps average, important variation observed.

• Video communication, VP8 video codec, Internet Network (xDSL-2-xDSL)

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600 x 600 Camera Medium video quality (~25 fps) = 580 kbps Rx

Medium video quality (~25 fps) = 820 kbps Tx

Impacts on IP network

• Explosion of IP QoS bandwidth: 2Mbps full duplex video stream per browser

Chrome RT engine tries to maximize the usage the hosting devices capabilities to optimize the user experience !

• Set the right QoS marking for Signaling, voice, voice & video, data, and ensure e-2-e QoS settings

• Congestion control is required for heavy loaded IP network 11/25/2013

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Impacts on IP network For consumer market 50% of session using video will

lead to 300-500% BW increase with corresponding interactive video/voice QoS

For enterprise market Huge impact on Enterprise Network

to supports “mobile” video conferencing endpoints. Up to 15x increase ! 11/25/2013

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Voice won’t be the problem !

Co

urt

esy

PK

E C

on

sult

ing

Forecast of data usage per user / month

Solutions to manage IP network There’s strong need for Active Network

Performance Monitoring:

• For Enterprise IP network, product (like PathView product) going across LAN, WAN, MPLS networks, providing continuous Monitoring , SLA Validation and QoS validation (Figure shows QoS flag has been stripped at hop 6).

• Existing OSS monitoring solution with specific plug-in, e.g. MPLS, LTE‘s HP NNMi Plug-Ins (figure shows the EnodeB neighborhood view – ANR

usecase)

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Solutions for WebRTC Applications

When the Carrier or Large Enterprise are part of the communication chain, there is a need for WebRTC to SIP GW to implement

→The media policies, e.g. – Authorized SD or HD Voice, Audio, stereo Audio – Authorized Video codecs, definition, bit-rate and frame-rate

→ on the fly Real Time transcoding from WebRTC Enterprise UC/VoIP/VoLTE/... worlds

→ e-2-e QoS mapping depending on the mobility scenario e.g. for Enterprise IP network, Audio mapped to ‘EF’ with higher Priority than Audio/Video mapped to ‘AF1’ on LANs & WANs

→ e-2-e Congestion control from WebRTC world to SIP e.g. Need RTCP XR extensions to let the application understand that the Packet loss is caused by congested routers

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11/25/2013

22 HikingArtist