WebRTC makes the process of communicating online easy, convenient and increasingly interoperable. A browser is opened, one click to make the call and the connection is made. Video calls, voice calls, file transfers and downloads are all completed through the browser, without any additional plugins. However, while WebRTC is poised to become compatible with popular smartphone browsers, the technology does allow end users to bypass traditional telecom voice and video calling services provided by mobile operators. This has prompted many in the telecom industry to view WebRTC as a threat to operators’ revenues. In reality, the opposite is true as WebRTC provides an opportunity for operators to monetize new services.
By supporting WebRTC endpoints (browsers running on PCs, laptops and even mobile phones), an operator could have access to millions of new end points through which consumers can use its services. However, before operators can seize the WebRTC opportunity and deliver real time services to all these terminals, they must overcome the challenges around WebRTC, while leveraging their own strengths to deliver differentiation.
WebRTC does come with quality, interoperability and reliability challenges that mobile operators are well positioned to address. When a WebRTC endpoint connects with another WebRTC endpoint on the same enterprise subnet, the connection is relatively straightforward. However, when the browser is trying to connect across the core mobile network, or another enterprise, this connection can become challenging. Early WebRTC users are already experiencing the difficulty of bypassing network firewalls, interconnecting different codecs, and variable bandwidth.
A secondary challenge involves using the public Internet for real-time communication. The availability of public network bandwidth doesn’t allow for quality of service, and a user can only hope for ‘best effort.’ This is the opposite of a telecom network designed for five nines high availability, low latency and jitter.
Another challenge comes in interconnecting WebRTC endpoints over the public internet, to mobile telephone numbers or Public Switched Telephone Network (PSTN) lines. These WebRTC calls need to be successfully routed and connected through an array of differing gateway infrastructure and back again. The bandwidth and quality of the public internet will have to be adapted to achieve reliable connections and improved service quality.
Seizing the Opportunity
Mobile operators are well positioned to solve these challenges as they own their own network. IP Multimedia Subsystem (IMS) architectures are capable of providing, and controlling, levels of service with telco-grade reliability to deliver a customer experience superior to OTT brands. The deployment of a Media Resource Function (MRF) in a mobile operator’s IMS core delivers the capabilities to process, transcode and transrate media travelling from one diverse endpoint to another.
A basic voice and video P2P call between two endpoints using different technologies requires some transcoding and transrating between the endpoints because each endpoint works to different codecs and varying bandwidths. This process becomes more complex if a multi-party call is conducted between WebRTC end points, necessitating a mesh formation. A centralized MRF can receive inputs from multiple WebRTC devices, complete the mixing, and then switch a single media stream to each end point, reducing complexity while using scarce wireless bandwidth more efficiently.
Operators’ network architectures also enable them to adapt bit rates and apply policies to guarantee bandwidth for improved WebRTC sessions, improving the user’s experience, and overcoming the ‘best effort’ nature of a public internet connection.
Operators can also manage WebRTC regulatory considerations. For WebRTC to extend into mainstream “lifeline” services, WebRTC must provide a mechanism by which calls can be recorded if and when required, such as for emergency services. An operator can help ensure regulatory compliance through its network infrastructure, in which there is equipment for recording calls using an MRF. Today’s OTT service offerings rarely consider regulatory requirements.
By deploying a multi-service MRF, operators can transcode and transrate media from WebRTC to SIP, and vice versa. Communication services can then extend from the WebRTC endpoints to a mobile phone, to the PSTN, an IP phone and many more. Operators have the gateways to deliver the interworking, the successful connectivity and the service quality. These centralized telecom network capabilities mean that operators can deliver a full suite of differentiated communications services – audio and video mail, ringback tones, video conferencing and more – via WebRTC.
Getting to Mainstream
It will take time for WebRTC to go mainstream and become part of operators’ strategies moving forward. While the WebRTC opportunity is vast, there are currently few companies generating significant revenues. This will change as technical hurdles are overcome, the technology matures, the business models develop and market traction kicks in.
Mobile operators that deploy highly scalable, flexible MRFs will be able to seize the WebRTC opportunity. They can target the delivery of high-quality, bundled real-time communication services which interconnect differing endpoint devices with WebRTC-enabled browsers, thereby ensuring the long-term viability of WebRTC, and more importantly, unlocking new revenue opportunities.
Ray has worked in the telecommunications industry with industry leaders including Convedia (now Radisys), Abatis (now Redback) and Nortel, along with system integration firms Deloitte Consulting and Accenture. He enjoys sharing his passion and viewpoints around IP-based telecommunication solutions with Radisys customers and partners. Ray has a B.A.Sc. in Systems Engineering from the University of Waterloo along with an M.B.A. from the University of British Columbia.
Edited by Stefania Viscusi