Contact Centers and the Voice-Enabled Web Peter J. Cornelius
Agenda Introduction WebRTC in the Contact Center LiveOps Browser VoIP Implementation Looking Ahead
LiveOps Built as a VoIP based solution for call centers PSTN only at the edge Call routing to at home agents Large, independent agent population Excellent test bed for general deployments
Why is the voice-enabled web valuable? Reduces costs Price of public Internet will always be lower than PSTN. Increases deployment reach Everyone has a browser, and all browsers will eventually support full-duplex voice. Enables new applications Web standards have a history of enabling new applications, and voice innovations within the browser are no different.
Voice-Enabled Web in the Contact Center
Barriers to Large-Scale Desktop VoIP Deployment and Maintenance complexity Upgrades dependent upon implementation Quality and consistency problems Codecs varied by implementation and configuration Lack of standards for developers No standard desktop integration API Behaviors and capabilities varied widely Infrastructure elements costly and/or difficult to manage Adobe Media Server Adobe Cirrus Service (Hosted, P2P only) openrtmfp/cumulus open source RTMFP Server
Why WebRTC? Built into the browser No other installation needed No components (e.g. Java applets) to download JavaScript API is natural and accessible for web developers Mobile browser support (phones, tablets) Automatic updates Based on open standards Unlike Flash, not under one company s control Standard specifies codecs, reducing uncertainty Voice - G.711, G.722, ilbc, Opus, and isac Video VP8 (+ H.264? IETF vote results?) Supports both peer-to-peer and intermediated communication Everyone is getting into the WebRTC business Vendors: Avaya, Cisco, GENBAND Open Source: FreeSWITCH, Asterisk
WebRTC P2P Connection Web Server Signaling Signaling Voice/Video/Data This is the WebRTC Triangle Signaling managed by Ajax/Comet or WebSockets Potential problems for the contact center No access to voice path can t record
WebRTC Intermediated Connection Caller Signaling Web Server Web Server Agent Signaling Voice/Video/Data Media Gateway Voice/Video/Data WebRTC Trapezoid Signaling still managed by Ajax/Comet or WebSockets Two triangles with media gateway, one for each party Voice/video passed through, recorded, etc. Potential problems for the contact center Latency must closely manage any audio processing delays
Implications No phone required Expensive desktop phones can be removed from the agent desktop Reduced telephony charges Fewer calls to toll-free numbers; replaced by direct connect from the browser No more extra line for home agents Application convergence Availability of new features and functionality Tight integration with the user or agent experience Faster development of custom agent applications
Challenges Standard is still settling API becoming firm, but uncertainty still exists in some areas (e.g. video codec) Browser support incomplete Internet Explorer not yet supported IE is still a factor in the enterprise Microsoft proposing alternative to WebRTC (CU-WebRTC) Safari not supported Call center agent desktops might need to be upgraded Agent workstations are not changed often Bare-minimum configurations might have audio quality issues Public Internet No control over QoS
LiveOps Browser VoIP Implementation
Capabilities Two lines of business Cloud based contact center platform and applications Bring your own agents Business Process Outsourcer (BPO) for call center agent services Independent contractor agents Features Multiple inbound/outbound media channels: PSTN or VoIP voice, email, web chat, SMS, Twitter, Facebook Interaction flow processing Cradle-to-grave reporting Voice and screen recording Agent, Admin, Supervisor applications Constraints 24/7 availability; no planned downtime Must support large, unpredictable bursts of activity Some stats: >300 Customers >40,000 registered agents >100,000,000 interactions processed per year
Agent VoIP Requirements Provide a no-install voice option Connect via Public Internet or direct connectivity infrastructure Public Internet will be adequate for most use cases Large customers may demand QoS assurances that public Internet cannot provide Maintain voice path quality Callers have low tolerance for audio quality degradation, jitter, echo, etc. Support older browsers to the extent possible Call centers are slow to upgrade; some contact centers still run IE8 Provide for easy move to newer technology Upgrade of browser should provide improvements without changes to our app
Endpoint Options Considered Softphone Examples: X-Lite, SJPhone, Broadsoft Business Communicator Our partners provide various SIP endpoint options Didn t want to introduce installable software to our platform Non-Flash browser plug-ins Examples: N-SIP, Voipfone, Gtalk, SureVoIP No real advantage over separate softphone installer -- requires just as much installation and maintenance work Generally paired with a VoIP subscription service Flash-based APIs Examples: Ribbit, Gizmo Flash is deployed broadly Version differences cause some clients to behave unpredictably Can be difficult for IT to manage Native WebRTC Preferable, but requires investment Third-party voice API providers Examples: Kandy, Twilio, Plivo, Phono Provide multiple options (Flash or WebRTC), abstracting the differences with their client APIs
LiveOps Browser VoIP Timeline 2000 LiveOps founded in Florida as a BPO POTS Phones 2001 Callcast founded in California as a contact center tech service 2003 LiveOps/Callcast companies technologies merge 2005 SaaS/PaaS business launched, larger customers added SIP Phones 2007 Fortune 500 customers added 2008 PCI Compliance 2009 Enterprise Agent product launched, continued growth Softphones 2010 REST API and Salesforce.com integration launched Ribbit Prototype 2011 Integrated multichannel tech stack, Data Exchange product launched 2012 LiveOps Application Server project started and launched WebRTC Prototype 2013 LiveOps Browser-based VoIP Support Released
Application Integration for Browser VoIP LiveOps Engage Application LiveOps Browser Phone API Third Party Connectors Twilio JS API SIP.JS API Genband JS API Browser WebRTC API Browser
LiveOps/WebRTC Integration for Agent VoIP 800 or DID inbound call terminates at LiveOps (PSTN) Call Data is sent via Agent API to LiveOps integrated agent experience via Public Internet LiveOps selects agent and sends Call to WebRTC Provider Extension (SIP or PSTN) Inbound call is received by LiveOps Application data is sent to desktop via LiveOps proprietary data channel LiveOps extends a PSTN call to a WebRTC provider to connect with agent WebRTC provider finds agent s registered client instance and opens voice path Legend SIP Data PSTN WebRTC provider sends VoIP call (WebRTC or Flash) to LiveOps Engage desktop client via Public Internet Caller is conferenced with agent by LiveOps platform In October 2013, we enabled WebRTC for a 500-agent call center We now have 20 call centers using WebRTC
Looking Ahead
Future Use Cases SIP Connections to WebRTC providers In house WebRTC capability WebRTC Call me now button Agent on mobile device Firefox and Chrome both support WebRTC on mobile browser ABI Research: 4.7 billion mobile WebRTC devices will be sold by 2018 Collaboration/Co-browsing Exploring use of TogetherJS Video support
Conclusion Cost Beats out softphone and plugin implementations Enables additional functionality and collaboration The few drawbacks have known solutions Enables an enhanced user experience