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Quality over WebRTC - Amir Zmora - AudioCodes - Upperside 2013
 

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 ...

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.

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

    • Amir Zmora VP Alliances & Partnerships @AmirZmora Quality Over WebRTC
    • Pre-VoIP PSTN E1/T1 Enterprise/Cont act Center
    • Enterprise VoIP PSTN SIP Trunk Enterprise/Cont act Center
    • Going VoIP All The Way Residential GW IMS Backbone Home user SIP Trunk Enterprise/Cont act Center
    • In Comparison to OTT Free Best Effort
    • WebRTC, Simplified Version Browser Public Internet Public Internet Web Server HTML & JavaScript Media OPUS/G.711 Signaling over WebSocket Enterprise Enterprise SBC/GW SIP IP Phone
    • 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
    • The Carriers’ Preferred Architecture USA Carrier’s Network WebRTC GW SIP Trunk Signaling over WebSocket Media OPUS/G.711 Browser SBC/GW Public Internet Enterprise/ Enterprise Paris Web Server Los Angeles Contact Center SIP HTML & JavaScript IP Phone
    • WebRTC Media Routing Federation Carrier A VoIP Backbone Media Carrier B VoIP Backbone Media Relay SIP Trunk Media OPUS/G.711 Browser Public Internet Signaling over WebSocket SBC/GW Enterprise/Cont Enterprise act Center Web Server SIP HTML & JavaScript IP Phone
    • Optimized WebRTC Media Routing Federation Session Experience Manager: ”You have quality issues” Browser Public Internet Signaling over WebSocket SBC/GW Enterprise/Cont Enterprise act Center Web Server SIP HTML & JavaScript Media OPUS/G.711 IP Phone
    • The SBC changes media route
    • Optimized WebRTC Media Routing Federation Carrier B VoIP Backbone Carrier A VoIP Backbone Media Relay Media Relay Media OPUS/G.711 Browser Public Internet Dynamic media routing decisions Signaling over WebSocket SBC/GW Enterprise/Cont Enterprise act Center Web Server SIP HTML & JavaScript Media OPUS/G.711 IP Phone
    • 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)
    • Amir Zmora VP Alliances & Partnerships @AmirZmora Thank You