Quality over WebRTC - Amir Zmora - AudioCodes - Upperside 2013

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The presentation "Quality over WebRTC" was presented at the WebRTC conference in Paris, http://bit.ly/19E3w5o. The presentation looks at the quality challenges in WebRTC as communication over the open internet and how this is different from current VoIP communication. Later the presentation reviews Tools that can help minimize these issues as well as a WebRTC network federation option managed by a cloud appliance that optimizes the communication for cost and/or quality.

Transcript of Quality over WebRTC - Amir Zmora - AudioCodes - Upperside 2013

Quality Over WebRTC Amir Zmora

VP Alliances & Partnerships

@AmirZmora

Pre-VoIP

PSTN

E1/T1

Enterprise/Contact Center

PSTN

SIP Trunk

Enterprise/Contact Center

Enterprise VoIP

Going VoIP All The Way

IMS Backbone

SIP Trunk

Enterprise/Contact Center

Home user

Residential GW

In Comparison to OTT

Free

Best Effort

Web Server

HTML & JavaScript

Signaling over

WebSocket

SIP

Media OPUS/G.711

IP Phone

Public Internet

SBC/GW

Public Internet

Browser

Enterprise Enterprise

WebRTC, Simplified Version

The Issues

Codecs – not matching, no HD all the way

Network impairments

Media route not controlled

IT Manager – “I have no clue what went wrong”

The Solutions

The Network Tools

Tools

Avoid transcoding

Use codes that support resiliency

Support WebRTC resiliency on the server

Minimize latency

Monitor your network

And act upon it

The Network

Consumer

Business

Public Internet

Web Server

HTML & JavaScript

Signaling over

WebSocket

SIP

Media OPUS/G.711

IP Phone

SBC/GW

Browser

Enterprise Enterprise/ Contact Center

The Carriers’ Preferred Architecture

Carrier’s Network

WebRTC GW

SIP Trunk

Paris

Los Angeles

USA

Public Internet

Web Server

HTML & JavaScript

Signaling over

WebSocket

SIP

Media OPUS/G.711

IP Phone

SBC/GW

Browser

Enterprise Enterprise/Contact Center

WebRTC Media Routing Federation

Carrier A VoIP Backbone

Carrier B VoIP Backbone

Media Relay

Media

SIP Trunk

Web Server

HTML & JavaScript

Signaling over

WebSocket

SIP

IP Phone

SBC/GW Enterprise Enterprise/Contact Center

Optimized WebRTC Media Routing Federation

Public Internet

Media OPUS/G.711

Session Experience Manager: ”You have

quality issues”

Browser

The SBC changes media route

Web Server

HTML & JavaScript

Signaling over

WebSocket

SIP

Media OPUS/G.711

IP Phone

SBC/GW Enterprise Enterprise/Contact Center

Carrier A VoIP Backbone

Carrier B VoIP Backbone

Public Internet

Media OPUS/G.711

Media Relay Media Relay

Dynamic media routing

decisions

Browser

Optimized WebRTC Media Routing Federation

What’s Needed to Make it Happen?

Federation agreements for WebRTC traffic

Cloud appliance

Quality assurance

Quality monitoring

Routing optimization

Mediation into existing deployment

End-to-End native WebRTC media support (Opus, VP8)

Thank You Amir Zmora

VP Alliances & Partnerships

@AmirZmora