WebRTC is Bringing "Customer Service of the Future" to the Present!
Today we’re bringing back one of our previous guest bloggers, Val Matula! Val is the Senior Director of Multimedia Research for Avaya Labs. His research is focused on combining dialog and speech technology, acoustics, video, messaging, and graphics in a way that helps businesses to improve their effectiveness. He currently holds 27 patents ranging from video customer service and social media interactions to speech and natural language processing.
WebRTC is promising to be the technology that will move many customer service sessions from toll-free calling to an all-Internet model. By embedding voice (and even video) communication components into browsers, and with the industry creating WebRTC software for mobile applications, customers who ‘start on the Internet’ will be able to stay there when they need agent-based assistance.
Yes, WebRTC supports person-to-person sessions, video conferencing, group sessions, and collaboration. However, it will be especially valuable in customer service applications, because it allows Internet consumers to reach agents without leaving a web or mobile application–enabling the agent to get the exact context of the customer’s issue, then provide specific help to the customer while still in the web app. Say goodbye to having to click-to-call an agent, going through voice menus, and then working to have an agent complete a transaction!
Furthermore, while still in the app the agent can also start a ‘co-browsing’ session, and thus see what the consumer sees in order to better help the consumer. If desired, the consumer can then continue on their way–allowing the agent to drop and serve another customer immediately!
While WebRTC’s initial design was for native browser-to-browser communication, I predict that many enterprises will use it as an on-ramp to the contact center core to unlock the value of thin client access to communications. Why? Contact centers need the ability to route a request for help (and an all-browser implementation might do this), but they also need continuous control of the media. In addition, the right codec(s) must be used to leverage existing contact center systems.
WebRTC provides the OPUS codec for its superior Internet-friendly attributes: Variable bit rate support, excellent voice and music quality, and good packet-loss concealment to ‘smooth over the cracks’ when an audio packet is lost or delayed in transit. While OPUS will be a native part of the contact center infrastructure over time, transcoding to a current common codec is likely in initial installations as part of this on-ramp process.
Digging deeper, session (call) recording requires the ability to tap into the voice or voice/video streams, and copy them to the enterprise recording platform. Furthermore, while WebRTC supports OPUS and G.711, recording systems initially will continue to support G.729 or internal codecs. During any call, a 3-way conference between customer, agent, and another agent or supervisor might be needed–so conferencing media systems must be engaged. Silent call observation is often done in a contact center for quality assurance and agent instruction–but when a supervisor ‘taps into a call’, the quality can’t change as a call is mixed, or otherwise it’s no longer silent: The last thing we want is the customer or agent saying “it sounds like someone is on the line”.
About two-thirds of the U.S. today has access to high speed internet, a virtual requirement for satisfactory WebRTC-based communication sessions. This means, one-third of the U.S. does not have this capability, and thus will likely continue to use toll-free calling for customer service. Having the ability to “blend” WebRTC and toll-free calling sessions will maximize agent utilization, while simplifying agent scheduling, infrastructure planning, not to mention reducing overall contact center costs. On-ramping WebRTC into a telephone-capable core solves this issue.
Finally, contact center agents may also find value in using WebRTC sessions as an ‘off-ramp’ from the contact center core. Agents on thin client devices, work at home agents, off-shored agents, and outsourced agents will benefit from OPUS codec support (which is possible also via SIP sessions), and from the ability to have fully browser-based agent desktops. Off-ramping permits thin client agents to take toll-free calls and WebRTC sessions like any other agent when done from the properly equipped contact center core.
In summary, WebRTC is being driven by the value of adding customer service into web service, rather than forcing a separate toll-free call. While WebRTC supports browser-to-browser communication, there is a strong story to be told for on-ramping and off-ramping WebRTC sessions into a contact center core that is capable of recording, conferencing, supporting thin client agents, and blending both WebRTC and toll-free customer sessions.
We’d love to hear what you think! For more insight you can check out the article: “The Future is WebRTC” in our 2013 Collaboration Trends Guide . AND…if you’re going to the WebRTC conference and expo next week in Atlanta, don’t miss my session “WebRTC and Customer Service “ Thursday at 2:30!