WebRTC Media Challenges

COMPANY CONFIDENTIAL © COPYRIGHT 2013 DIALOGIC INC. ALL RIGHTS RESERVED.

WebRTC Media Challenges

April 2014

COMPANY CONFIDENTIAL © COPYRIGHT 2013 DIALOGIC INC. ALL RIGHTS RESERVED. 2

Agenda

WebRTC Introduction

Why Server-side media?

WebRTC Gateway Options

WebRTC in Contact Centers

Multi-party Confernecing

Security Considerations

PowerMedia XMS


WHAT IS WEBRTC?


What is WebRTC – world-class VoIP engine for free

OPUS

On2

Global IP Solutions

.org

Video CodecGoogle acquired video compression

company On2 in February 2010 for $133 million

VoIP Media EngineGoogle acquires Global IP Solutions for $68.2M for real-time video and audio

Audio codecOpus is a totally open, royalty-free, highly versatile

audio codec standardized by the Internet Engineering Task Force (IETF) as RFC 6716 which incorporated technology from Skype's SILK codec

and Xiph.Org's CELT codec.

Open Source Community


What is WebRTC – pending Web and Internet Standard

draft-ietf-rtcweb-audio-05

draft-ietf-rtcweb-data-channel-07

draft-ietf-rtcweb-data-protocol-03

draft-ietf-rtcweb-jsep-06

draft-ietf-rtcweb-overview-09

draft-ietf-rtcweb-rtp-usage-12

draft-ietf-rtcweb-security-06

draft-ietf-rtcweb-security-arch-09

draft-ietf-rtcweb-stun-consent-freshness-01

draft-ietf-rtcweb-transports-03

draft-ietf-rtcweb-use-cases-and-requirements-14

WebRTC 1.0:Real-time Communication Between Browsers•Editors draft

•First Public Working Draft: 27 October 2011

•Feb 9 2012 Working Draft

•Aug 21 2012 Working Draft

•Last Call Working Draft: re-scheduled for Q1 2014

•Candidate Recommendation: initially scheduled for

Q4 2012, but delayed probably until Q2 2014

webrtc rtcweb

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/

http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/

http://dev.w3.org/2011/webrtc/editor/webrtc.html

http://www.w3.org/TR/2011/WD-webrtc-20111027/




What is WebRTC - 3 Javascript APIs used in the browser

CreatePeerConnection

GetUserMedia

Data Channel

Creates a media connection (RTP) between 2 peers

Grants access to local microphone and/or camera

Allows the user to send arbitrary data between peers


Why Enterprises & Contact Centers like WebRTC

User benefitsNo context switching

No minute usage

Video option: high value customers expect a premium experience

Faster time to delight

Contact center benefitsIntegrated in with lower-cost web CRM systems

Faster resolution = decreased agent time

Customer delight = return customers


WebRTC browers are growing to be ubiquitous

1 Billion Endpoints!


Why Developers Like WebRTC

HardcoreTelco SoftwareInfrastructure

(10k’s)

Source: TADS, Alan Quayle

Telecom App Developer Hierarchy

Robust & Reliable

Cheap & Flexible

Easy & Integrated

Expectations

IT/Web Programmers building on FOSS, telecom app platforms and

telecom API’s(1M’s)

‘Cut and Paste’ web developers using web-scripting and graphical tools on app platforms

(10M’s)


Basic call scenario

1. Alice connects to website & downloads Javascript application

2. Alice’s browser asks if it is ok to share media

3. Bob connects to website & downloads Javascript application

4. Bob’s browser asks if it is ok to share media

5. Alice clicks “call”

6. The Server tells Alice’s browser how to call Bob (signaling)

7. Alice’s browser calls Bob

8. Bob accepts

9. Media is routed direcly between Alice & Bob

Server

Alice Bob

media


WebRTC Technologies

NAT Traversal• ICE• STUN• TURN

Audio Codecs• Opus• G.711

Video Codecs (TBD)• VP8• H.264

Encryption• DTLS

Port reduction• RTCP Multiplex• RTP Bundle

Signaling• Session Description Protocol


WHY SERVER-SIDE MEDIA?


Typical WebRTC Models – P2P media

Server

client client

media

triangle

Server

client client

signaling signaling

media

trapazoid

Serversignaling

Lowest server & network cost


P2P with TURN

Server

client client

triangle

Server

client client

signaling signaling

trapezoid

Serversignaling

TURN Server

TURN Server

more server & network cost


WebRTC with Server-side Media

client client

triangle

Server

client

signaling media

trapazoid

signalingServer

client

signaling media

more server & network cost

Server


Design Trade-offs – Cost vs. Performance vs. Location

Application Performance

Server Performance

Client Performance

Higher infrastructure

costs

Battery lifeSpeed

Quality expectations


When does server-side media make sense?

When bandwidth is limited

When the user is paying per GB

When power is limited

When there is regulation


PowerMedia™ XMS

Multi-party mixing InterworkingTranscoding

Stream processing Person-to-machineRecording

WebRTC Media Processing Use Cases


WEBRTC GATEWAY MODELS


“Legacy” Telephony Interworking Functions

HTTP-to-SIP(H2S)

Media GW

Transcoder

NetworkWebRTC Users Phone User

SRTP-DTLS

UDP/TCP

RTP/SRTP-SDES

UDP

OPUS/VP8 G.7XX/H.264


WebRTC Gateways Come in Many Packages

HTTP-to-SIP(H2S)

Media GW

Transcoder

HTTP-to-SIP(H2S)

Media GW

Transcoder

HTTP-to-SIP(H2S)

Media GW

Transcoder

HTTP-to-SIP(H2S)

Media GW

Transcoder


Will One H2S Gateway be Sufficient?

Retail Click-to-Call

Enterprise Click-to-Call

Residential Web Phone

RCS Web Client

Developer API exposure

Remote Agent

Remote Worker

HTTP-to-SIP(H2S)

IMS/RCS

RFC3261 SIP

Enterprise SIP

Application specific APIs work best SIP is not really that standard


Media Requirements More Uniform than H2S

H2S

WebRTCMedia GW

Transcoder

NetworkWebRTC Users SIP/IMS Networks

Web Signaling

SRTP-DTLS

UDP/TCP

RTP/SRTP-SDES

UDP

OPUS/VP8 G.7XX/H.264

H2S

H2SSIP

H2S

Control Interfaces


3GPP TR23.701 WebRTC Gateway Architecture for IMS

H2S

WebRTCMedia GW

Transcoder

JavascriptClient

Webserver


WEBRTC IN CONTACT CENTERS


State-of-the-Art in Conferencing & Collaboration

Voice, video, text

2 or multi-party

Web or mobile app

PSTN calling

Share screen, desktop, docs

Webcasting

Real-time video augmentation

APIs for 3rd parties26


State-of-the-Antiquated?

99.999% reliability

Universal interoperability

IVR with TextSpeech

Regulatory compliant

Supervisor modes

Recording & playback

API’s for systems integration

27


Contact Center Integration

28

click

Application Server

Web Apps

SBC Media ServerAgent

Media Interface

Media Control

SIPSIP

RTP RTP

HTTP

Location

• Traditional click-to-callback accelerated by WebRTC

Web-based Application integration

PSTN

Contact Center



29

Web-based callers• Eliminate PSTN charges• Faster time-to-resolution• Improved customer experience

Web Apps

Agent

Media Control

WebRTC Signaling

SIP

SRTP RTP

HTTP

WebRTC Browsers

Application Server

Media Interface

HTTP

Contact Center

Media Server



30

Web-based agents• WebRTC reduce cost of

supporting remote agents• Optional video integration

Web Apps

Agent

Media Control

WebRTC Signaling

HTTP

Location

WebRTC Browser

Application Server

Media Interface

SBC Media Server

SIP

RTPPSTN

Contact Center


Case Study – the Airline

31


Video-enabling a Contact Center

Web App Server

Media Control

HTTP/REST

Media Interface

SBC Media Server

RTP

Contact CenterSRTP

PSTN

PSTN Customers

Agents

ACD

Kiosk Customer

RTP

SIP

HTTP/REST


Case Study – the Airline

33

Lessons Learned:• Applications are viable today when you have

end-to-end control• Leveraging existing infrastructure shortens

time-to-market• Assume you will have poor connectivity• Be careful with variable cloud latencies


MULTI-PARTY CONFERENCING


Challenge: Multi-party topology

35

Desktop SIP video client

Video stream

Android tablet with SIP client

WebRTC browser

iPad with SIP client

Android smartphone with SIP client


Peer-to-Peer Mesh with a Few Parties

36

• Works for a few parties

• No server cost• Lowest latency


Peer-to-Peer Mesh with Many Parties

37

• Doesn’t scale for many parties• Heavy

processing burden on all parties

• Massive bandwidth


Centralized Routing

38

• Some server cost• Lots of bandwidth• Lots of client

processing

Video Router


Centralized Routing with SVC

39

• Some server cost• Reduced bandwidth• Reduced client

processing• Requires specialized

clients

Video Router


Streamline with Media Server

40

Multipoint Control Unit (MCU)• Provides scalability• Improves mobile

performance• Dynamic back-end

control• Least bandwidth &

client processing

MCU


Challenge: mixed capabilities

41

Desktop SIP video clientAndroid tablet with SIP client

WebRTC browser

iPad with SIP client

Android smartphone with SIP client

1920x108030 FPSH.264

640x48015 FPSH.261

352X28830 FPS

VP8

1280x72025 FPS

VP8

1280x72030 FPS

VP8

1280x72030 FPSH.264


0

240

480

720

960

1200

1440

1680

1920

2160

0 320 640 960 1280 1600 1920 2240 2560 2880 3200 3520 3840

Video Resolutions will Continue to Expand

720P.9MP

1080P2.1MP

VGA/.3MP

4K8.3MP

WebRTCResolutions

today


SECURITY CONSIDERATIONS


Scope All Traffic APIs VoIP Traffic

Identifier Address:Port API Key Phone #, URI

Controls Bandwidth, Address,Ports

API Calls VoIP Messages, Bandwidth, Ports

Topology Hiding

No N/A Yes

WebRTC Security - Looks More like Web than Telco

API Management

Session Border Control

Firewalls


Basic, Telco-Centric Model – Trust the SBC, Not the Web

Web server

SBC

SIP/IMS Network

SIP

JavascriptApp

H2S

WebMG

SIP

RTP

Firewall


Mature, Web-Centric Model – Re-use Web Security

Web server

SIP/IMS Network

SIP

WebMGFirewall

AP

I Man

ager

API Calls

H2S Server

Identity Server

App Server

App Server

WSS server

STUN/ TURN

Internal REST APIs


POWERMEDIA XMS


PowerMedia XMS

Virtualized

Cloud-ready

Audio

Video

WebRTC

Robust Developer APIs

Standardized Production Interfaces

real-time media processing platform


SIP Web Java Speech

Audio Video WebRTC CPA

PowerMedia™ XMS

Media Platform for Developers & IT

RESTful

Java (JSR.309)

VXML

MSML

NETANN

MRCPv2

JavascriptPow

erfu

l Med

ia A

PIs

PowerMedia XMS

Diverse Developer Audiences

Extensive Capabilities


PowerMedia™ XMS

Multi-party mixing InterworkingTranscoding

Stream processing Person-to-machineRecording

PowerMedia XMS – Many Use Cases

RESTful

Java (JSR.309)

VXML

MSML

NETANN

MRCPv2

Javascript

Powerful Media APIs


PowerMedia XMS Media Resource Function (MRF)

IMS compliant MRF interfaces

XMS controlled via SIP from ASMSML or VXML embedded inside SIP signaling directed to XMS

All core SIP traffic brokered by S/I-CSCF

Offloads media processing from ASMulti-party conferencing

Announcements & Prompts

IVR

Ring back tones

Voicemail

Transcoding

Future ready - comprehensive video & WebRTC support

Starting point for NFV – 100% virtualized software

Hypervisors: VMWare, KVM, XEN, OVM

TAS

MGCF/ MGW

SBC/ P-CSCF

CSCF

PowerMedia XMS MRF

SIP

RTP

Mr

Mb Mb

ISC

Mg/MiMw


PowerMedia XMS WebRTC Media Gateway

Converts WebRTC media to standard VoIP media

Encryption interworking

Media plan interworking

NAT traversal negotiation

Rapidly integrates with applicationsWeb apps - RESTful & Javascript API

Java apps – JSR.309

SIP – MSML, NetAnn

Scripting interface – VXML

Proven integration with many ecosystem partners

Streamlined media planeLess infrastructure -reduced CAPEX & OPEX

Offloads media functions from applications'

SIP/IMS Networks

H2S H2S

H2S H2S

Media Control Interfaces

PowerMedia XMSWebRTC Media Gateway

Developer & Partner Applications with Signaling


z

PowerMedia XMS Software Transcoder

Virtualized software-based transcodingLeverage cloud assets instead of expensive DSPs

Inline transcoderActs as a B2BUA

No signaling control required

Transcoding gatewayUse media server control interfaces

RESTful API

RFC4117*

Comprehensive & rapid codec supportAudio - G.711u/a, G.723, G.726, G.729a, G.729b, AMR-NB, iLBC*, Wideband: G.722, AMR-WB** (G.722.2), Opus

Video - VP8, H.264, MPEG 4, H.263, H.263+, H.263++, VP9*

MGW/SBC

SIP/IMS Core

MGW/ SBC

Codec A Codec B

Inline Transcoder

PowerMediaXMS

MGW/SBC

SIP/IMS Core

MGW/ SBC

Codec A Codec B

Transcoding Gateway

PowerMedia XMS

Signaling

Media

** - Using the AMR-WB resource in connection with one or more Dialogic products mentioned herein does not grant the right to practice the AMR-WB standard. To seek a patent license agreement to practice the standard, contact the VoiceAgeCorporation at http://www.voiceage.com/licensing.php.

http://www.voiceage.com/licensing.php


• Audio Codecs• G.711u/a, G.723, G.726, G.729a, G.729b, AMR-NB, iLBC*

• Wideband: G.722, AMR-WB** (G.722.2), Opus

• Video Codecs and Processing• VP8, H.264, MPEG 4, H.263, H.263+, H.263++

• HD720p, VGA, QVGA,4CIF, CIF, QCIF

• Frame rates: Up to 30 FPS

• Bit rates: Up to 2Mbps

• MSML (SIP based)

• RESTful API (http based)

• JSR309 (Java based)

• VXML (script based)

• NETANN (SIP)

State of the art rich media processing

• Streaming and Recording

• Advanced Conferencing

• Transcoding, Trans-rating, Trans-sizing

• Video text and image overlays, scrolling

• Tone Generation / Detection

• Call Progress Analysis

• Positive Voice Detection (PVD)

• Positive Answering Machine Detection (PAMD)

• SIP (RFC 3261)

• RTP, RTCP, RTCP-XR, RTCP-HR, SRTP

• RTSP (client support, streaming media)

• MRCPv2 (ASR/TTS server support)

• ICE Lite / STUN* / TURN*

• IPv4, IPv6*** - Using the AMR-WB resource in connection with one or more Dialogic products mentioned herein does not grant the right to practice the AMR-WB standard. To seek a patent license agreement to practice the standard, contact the VoiceAge Corporation at http://www.voiceage.com/licensing.php.*planned for 2014

SIP / WebRTC / RTP Specifications

Audio / Video Media Functions Audio / Video Media Functions

Media Control Protocols

http://www.voiceage.com/licensing.php

Technology

WebRTC Media Challenges