29 Sep 2014

Capturing the WebRTC Opportunity

Ray Adensamer, Senior Product Marketing Manager at Radisys, explains how operators can successfully seize the opportunities that WebRTC offers.

Web Real Time Communications, WebRTC is a technology that delivers interoperable browser-to-browser applications for voice calling, video chat and peer-to-peer (P2P) file sharing – without the requirement for any additional plugins. WebRTC is poised to become compatible with popular smartphone browsers.

However, WebRTC technology does allow end users to bypass the traditional telecom voice and data services provided by mobile operators. This has prompted many in the mobile industry to condemn the technology as a threat to operators’ revenues.

In reality, it is quite the opposite. The technology actually poses a considerable opportunity for operators. By supporting WebRTC endpoints (PCs, laptops and even mobile phones), an operator could have access to millions of new end points through which consumers can potentially access its services.

A typical cellphone handset that can operate on 4G LTE

Before WebRTC, the process of delivering a broad set of real-time services was incredibly complex. Specialised Session Initiation Protocol (SIP) endpoints or browser plug-ins were needed to support real-time service delivery. However, before operators can leverage WebRTC to deliver real time services to all these terminals, they must first solve service quality and experience issues, technical interoperability problems, and challenges around service reliability.

WebRTC Beginning

In May 2011, Google released an open source project for browser-based real-time communication known as WebRTC. Now, WebRTC capabilities are included in HTML 5 - the latest markup language of modern Internet browsers - used for structuring and presenting web content. HTML 5 has been adopted by several leading browsers and has been championed by Google Chrome and Mozilla Firefox.

WebRTC can allow users to bypass traditional IP Multimedia Subsystem (IMS) telecom services, providing a powerful ability for web developers to set up real-time audio and video connections as part of a low cost web browser. For example, if an end-user is on the same subnet, they can set up a peer-to-peer voice or video call between browsers without any centralised media processing systems. An example would be Facetime – which enables two iPhone users on a WiFi connection to make a video call to each other. What WebRTC does is similar, simpler, and more ubiquitous. Rather than needing a Facetime application, inherently limiting communications capabilities to an Apple device, WebRTC runs in a standard browser, with no special plugins, and is more ubiquitous than just Apple devices.

Before WebRTC was developed, real-time services required a specialised OTT or Session Initiation Protocol (SIP) client to be downloaded, installed and configured. But these services suffer from significant market fragmentation. They are, in effect, closed islands of communication as one service user cannot communicate with the customer of another service.

For example, a user sending a message on Skype could only send that message to another Skype user – not a WhatsApp user. WebRTC acts as a vehicle to remove this barrier and liberates closed communication. An end user requires just an end point to initiate a call. WebRTC makes the process of communicating online easy, convenient and highly interoperable. A browser is opened, one click to make the call and then the connection is made. Video calls, voice calls, file transfer, downloads – all completed through the browser.

The vision for WebRTC is that any HTML 5 capable browser will be able to support WebRTC. Yet despite significant support from the browser community and many technical benefits, WebRTC is not yet ubiquitous. Microsoft’s Internet Explorer and Apple’s Safari browsers do not support it. But assuming that ubiquity eventually materialises, the sheer number of browser endpoints has led many in the mobile industry to become nervous that WebRTC-based voice and video communication services will heavily impact operator revenues, as end-users may opt to initiate browser to browser calls, bypassing traditional telecom service platforms in the core network.

However, mobile network operators still have a significant role to play. In solving service quality and experience issues, technological interoperability problems, and challenges around service reliability, operators stand to recapture an important role in managing WebRTC services which can be monetised, and also help speed up ubiquitous WebRTC use.

The Challenge Ahead Before the WebRTC opportunity can be seized by operators, there are a range of quality, interoperability and reliability challenges that must be overcome. To provide an example, let’s take a scenario in which a user is on the same enterprise subnet and is connecting, via WebRTC, from his own browser to another. That type of connection is relatively straightforward. However, when the browser is trying to communicate with the core mobile network, or another enterprise, this connection can become very challenging. Early WebRTC users are already experiencing the challenges of bypassing network firewalls, interconnecting different codecs, and variable bandwidth.

Interoperability is not the only issue effecting WebRTC. If consumers are using browsers for real-time communication, they will inherently be using the public internet. The availability of public network bandwidth does not allow for any quality of service. A user can only hope for ‘best effort’. This is the complete opposite of a telecom network – which is designed for five nines high availability, low latency and jitter, and provision for regulatory requirements.

An additional challenge comes in conducting these real-time audio and video calls, over the public internet, to mobile telephone numbers or Public Switched Telephone Network (PSTN) lines. These WebRTC calls are going to have to be successfully routed and connected through an array of differing 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.

The biggest challenge is improving the quality of these WebRTC media sessions and evolving them beyond a ‘best effort’ OTT connection in an unmanaged network. Customers are going to expect QoS which is comparable to what mobile operators can provide.

Despite these challenges, the WebRTC space will be a fiercely fought market space.

Operators Ideally Placed to Capture WebRTC Opportunity

There is one group of companies that is ideally placed to solve the service quality, reliability and technological interoperability issues posed by WebRTC. Mobile operators are able to provide a superior WebRTC experience due to one main factor – they own their own network. These architectures are capable of providing, and controlling, levels of service with telco-grade service reliability, delivering a superior customer experience that OTT brands can only dream of. Many of the challenges of WebRTC and SIP interoperability can be met by a piece of equipment which can be deployed in an operators’ IMS core – the Media Resource Function (MRF). The MRF provides a mobile operator with the ability to process, transcode and transrate media travelling from one diverse endpoint to another. For example, a basic voice and video P2P call between endpoints using different technologies often requires some transcoding and transrating between the two endpoints.

This is because each endpoint works to different codecs and varying bandwidths. However, this process becomes a lot more complex if a three- or four-way call is being conducted between end points in a mesh formation. If the target for this type of communication is to be scalable and use the bandwidth efficiently – then a centralised MRF for mixing and switching of a single media stream to each end point is crucial for successful interoperability. Bridging the gap between the telco and IP worlds is complex, but operators are prepared for this challenge by virtue of the MRF.

Over and above the media processing abilities of the mobile operator, their network architecture enables them to adapt bit rates and apply policies which can guarantee bandwidth for improved WebRTC sessions, improving the user’s quality of experience, and overcoming the ‘best effort’ nature of a public internet connection.

The mobile operator is also best placed to deal with the regulatory considerations that will arise for WebRTC. This technology must provide a mechanism by which calls can be recorded if and when required. This is particularly pertinent for calls to emergency services, to corporate contact centres and for the purpose of legal interception. An operator can ensure this regulatory compliance through its network infrastructure, in which there is equipment capable of recording calls using an MRF. An OTT service with no centralised service delivery infrastructure cannot perform this kind of capability.

By leveraging the abilities of its network architecture, including the MRF, operators can transcode and transrate media from WebRTC to SIP, and vice versa. Communication services can then be extended from the WebRTC endpoints to a mobile phone, to the PSTN, an IP phone and many more.

The operator has the gateways to deliver the interworking, the successful connectivity and the service quality. These areas just cannot be provided in a P2P environment. These centralised telecom network capabilities mean that operators can look forward to delivering a full suite of communications services via WebRTC. These comprise audio and video mail, ringback tones, video conferencing and many more.

WebRTC Ecosystem Develops

An entire range of very different businesses are set to benefit from the successful launch of WebRTC. For example, vertical companies in any sphere of customer service can leverage WebRTC in their websites to expand browser and chat services to now include voice and video calling. Social media players will also be interested in leveraging WebRTC to add extensions to their existing communications functions – in order to broaden their reach into the consumer space. App development companies are also entering the fray with the provision of environments that bridge the telco and IP world, enabling apps for WebRTC to be nurtured.

However, it is important for infrastructure providers and mobile operators to also realise the importance and significance of supporting WebRTC applications, driving interworking and interoperation between WebRTC clients and IMS, wireline and wireless. Admittedly, operators are much more tentative when it comes to WebRTC. Some players still see the technology as a threat, but mostly because they are large organisations that move slowly on developing new technology.

But the majority of the early market activity is around innovation and gearing towards how WebRTC can be monetised. Early adopters are developing some very cutting-edge services, but they are still OTT style offerings which don’t generate great financial returns. These OTT style early adopters can be more nimble, as they don’t own their own networks, and can focus more on marketing and rapid service delivery through innovation. The advantage for these early movers is that they will be the first companies to prove out WebRTC innovations. However, it will be the mobile operators that adopt the leading ideas and improve upon them with their QoS and interoperability solutions.

Getting to Mainstream

It will take time for WebRTC to go mainstream, and become a part of operators’ strategies moving forward. The WebRTC opportunity is vast – but there are currently no companies generating significant revenues just yet. It is going to improve as the technical hurdles are overcome, the technology starts to mature, the business models develop and market traction kicks in.

And if those inherent challenges posed by WebRTC are to be solved by operators – then the answer lies within a highly scalable, high density MRF in the core of their IMS-based architectures. An MRF provides an operator with the crucial ability to deliver media transcoding and mixing between WebRTC and SIP end points for scalable and quality real time services. In tandem, the mobile network provides the policy management and bit rate adaptation to make WebRTC work as it was intended – even in a ‘best effort’ public internet environment.

Mobile operators can target the ability to deliver high-quality, bundled real-time communication services which interconnect differing systems and WebRTC-enabled browsers. Doing this not only ensures the long-term viability of WebRTC, it also delivers on its promise of ubiquity and has the potential to unlock new revenue opportunities for operators.

Moreover, WebRTC could also provide that valuable route back into the hearts and minds of the consumers that have been captivated by the OTT onslaught.

Page 1 of 1

About the author

Ray Adensamer is a Senior Product Marketing Manager with Radisys and is part of the senior leadership team that defines overall business unit strategy, product enhancements and development priorities. Ray has been with Radisys for over seven years and has also worked in the telecommunications industry with companies including Convedia (now Radisys), Abatis (now Redback) and Nortel, along with system integration firms Deloitte Consulting and Accenture.

Radisys (NASDAQ: RSYS) is a market leader enabling wireless infrastructure solutions for telecom, aerospace and defense applications. Radisys' market-leading MRF (Media Resource Function) and T-Series Virtualized Platforms coupled with Trillium software, services and market expertise enable customers to bring their products to market faster with lower investment and risk. Radisys technology is used in a wide variety of 3G & 4G / LTE mobile network applications including: small cell Radio Access Networks (RAN), wireless core network applications including SDN and NFV, deep packet inspection (DPI) and policy management equipment; conferencing and media services including voice, video and data, as well as commercial offerings for network applications that support the aerospace and defense markets.

Most popular articles in Cellular telecoms

  • Carrier Aggregation – How to Test the Key Enabler for LTE Advanced
  • Cellular Base Station Installation & Maintenance Challenges
  • End-to-End Assessment of Mobile Video Services
  • Realizing the Promise of 5G: utilizing the technologies
  • 4G, 5G & IoT Predictions for 2016
  • Share this page

    Want more like this? Register for our newsletter

    The benefits of replacing plain old paper with e-paper displays in automotive assembly plants HD Lee | Pervasive Displays
    The benefits of replacing plain old paper with e-paper displays in automotive assembly plants
    Efficiency is at the heart of automation, and that is nowhere more apparent than in the manufacture of automobiles. The Ford Motor Company is widely credited with inventing the moving assembly line, but the concept of moving the assembly, rather than the worker, dates back centuries.

    Radio-Electronics.com is operated and owned by Adrio Communications Ltd and edited by Ian Poole. All information is © Adrio Communications Ltd and may not be copied except for individual personal use. This includes copying material in whatever form into website pages. While every effort is made to ensure the accuracy of the information on Radio-Electronics.com, no liability is accepted for any consequences of using it. This site uses cookies. By using this site, these terms including the use of cookies are accepted. More explanation can be found in our Privacy Policy