What is WebRTC?

WebRTC is an emerging standard to enable real-time communications (voice, text, video, data) directly on a web-browser running in any machine or mobile phone. It is like having ubiquitous Skype, but without the need to install any proprietary application or browser plugins.

With WebRTC, web developers can easily embed rich voice/video/data applications into web pages or apps using nothing more than  HTML5 and JavaScript. They can quickly develop new communication applications without owning media pipes (once connection is set, data flows peer-to-peer) and writing just a few lines of code.

Skype, Google Hangouts, and other stand-alone communication applications are changing how we communicate using our laptops and smartphones. WebRTC integrates that change to web services and mobile Apps.

WebRTC and Customer Service: Transforming Customer Communications

One of the greatest challenges for Customer Service today is to keep context independent of channel. A conversation that starts on the website via chat can turn into a phone call and then transition to Twitter to be eventually resolved by email.

The promisse of WebRTC is to be the single channel of real-time communications enabling the seamless transition between data, text, voice, video sharing. For example,  "Click to Call" buttons become trivial and independent of platform or type of client device.

So what we see in the horizon is a future of no more traditional phone lines, no more proprietary web widgets for chat, and a seamless, integrated channel for real time communication that will both make current problems irrelevant and create a new set of challenges for Customer Service organizations.

Where is WebRTC Today?

WebRTC is a standard driven by the IETF and W3C. It is in draft mode and is currently supported by Google Chrome, Mozilla and Opera Browsers. Safari and Internet Explorer are not yet supporting the WebRTC standards, although plug-ins are available for both browsers.

A good gateway for additional information can be found at http://www.webrtc.org/ (which is maintained by the Chrome Browser team).

Daitan Group is a pioneer in WebRTC development for business applications. Our development partners were the first in the industry to introduce call center and customer service products supporting WebRTC. If you plan to develop WebRTC services, don't start it before contacting us.

In the past decade or so the software industry has moved from waterfall to Agile Software Development methods.

In waterfall development, marketing and engineering signed a “contract” based on a fixed scope of work (the “Product Requirements Document”) and embarked in a long project based on engineering estimates of execution complexity. That provided a false sense of predictability and risk management. The problem is that requirements change and estimates are proved wrong as soon as developers start writing code, resulting in delays, budget overruns, and mismatch between product capabilities and user needs.

The idea behind Agile is that both marketing and engineering teams develop a relationship of trust and share common goals and responsibilities. Both sides agree that plans are subject to change and work collaboratively, managing schedule, scope and cost,  continuously making plan adjustments often. The result is more efficiency and better products.

Is Your Software Development Agile? Think again

Another trend in software development that has only increased in the past decade is the use of external development teams (outsourcing). It is about cost and risk management and accessing skills where they are readily available.

If you do or plan to use external teams for software development, you must transform your thinking on the relationship with them to materialize the efficiency advantages of Agile development. The interface between product developers and traditional outsourcing providers has not yet caught up with the level of transparency and collaboration required for Agile methods to work.

So, what do you need to look for when working with a software development partner?

Here are the top 3 Requirements for Agile Outsourcing:

1. Cost Efficiency.  Yes, there are other good reasons to outsource, but managing development costs is still an important driver. The common mistake is to focus in hourly cost per developer. You need to think cost of results and development organizations working in Agile with more capable developers should be able to deliver a lot more than just low cost per head.
2. Technical Capabilities.  In an Agile development team, every member must interact with others and contribute their ideas. If external developers are seen as second-class contributors, you are on the wrong path. They should be as capable as the internal team and bring their previous technical and process experience in similar projects to add value to the relationship.
3. Collaboration and Communication.  Since outsourcing often involves teams in different countries, factors such as language, time zone, cultural differences can be a challenge. Before starting a project make sure you can trust your partner and that the communication infrastructure is adequate and that there is a good match of culture, language and capabilities to communicate in real time. Joint development is not a transaction, it needs to be a trusted relationship between teams.

As we have written in previous articles, WebRTC is a communication platform and set of APIs drafted by W3C that is embedded into the common Web Browsers we use every day to browse the Internet. The intention is to add real time communication capability into the browser via simple JavaScript APIs. This is a great opportunity for innovation; it will have a major impact on the communication industry allowing many web developers to begin to more easily embed real time communications into their applications and web sites.

Rather than replacing traditional telephony or video conferencing in the short term, WebRTC will fuel the development of a whole new generation of web based communication applications and features that will be added to many of the internet and business products and services we use today on the internet and in business.

On the consumer/client side, we will be able simply to open a real time audio/video and data sharing session directly in our browser. For the web developer this will be fairly easy to achieve by using the WebRTC API's and JavaScript.

This simplicity for the developer and in the end user experience is great for new services that will run with WebRTC end-to-end, but a large problem remains to be addressed for enterprise and service providers as they are forced by their users and competition to offer new WebRTC functionality through web clients and make it work seamlessly with the communication infrastructure and processes in which they have already invested.

WebRTC InteroperabilityWebRTC takes care of media channels very well, but what it does not do is to provide or specify any signaling protocol or mechanism to do that. Instead it is left to the application developers to choose whatever protocol they want to send and to receive control messages among the peers for controlling features of the call. Luckily there is no shortage of signaling options that WebRTC applications could use: we could use an HTTP request mechanism or a more elaborate protocol like XMPP or SIP implemented in JavaScript.

Application developers also need to think about potential compatibility issues in the media stream. To interoperate with WebRTC the legacy endpoints would need to support some of the required WebRTC features, such as VP8 codec, SRTP/SRTCP for media transport, and Interactive Connectivity Establishment (ICE) for transport address negotiation. They would also need to consider the Session Description Protocol (SDP) that carries the required parameters for the call. WebRTC uses SDP but requires and implements some additions to the protocol so some form of mediation will be required.

To achieve the required interoperability, the legacy endpoints would need to be modified in order to be compatible with WebRTC, or some form of gateway solutions will need to be in place to manage the necessary protocol, signaling and media conversions.

The following is one example of how basic interoperability could be achieved.

WebRTC to PSTN InteroperabilityLet's take as an example the scenario where a WebRTC client is used to make a call to a PSTN landline.

To make a connection to the PSTN the WebRTC application would need to deal with differences in address (number) resolution, signaling, media conversion, etc. Many of these problems have already been solved in telecom systems, so we would recommend using some of the existing telecom solutions and components like SIP Servers, Session Border Controllers, and Media Gateways that already contain much of the necessary intelligence, protocols and infrastructure for interoperability with PSTN.

In our simplified example we'll assume the WebRTC client will be talking to a SIP Gateway in the network and consider what is necessary.

In the WebRTC side we need to consider that in order to create a media session the WebRTC client application will need to obtain:

• Local connection information: IPs and Ports that could be used to receive the audio and video
• Remote connection information: IPs and Ports that could be used to send the audio and video
• The list of media codecs

Fortunately this part of the problem has already been addressed in WebRTC where this functionality is provided through ICE and the use of SDP (Session Description Protocol).

In order to talk to a SIP Gateway in the network, the WebRTC client must also support SIP. As we mentioned earlier WebRTC does not specify or implement signaling, and SIP is not supported natively in the browsers. There are, however, JavaScript libraries available to implement this, including a free JavaScript library "jsSip". At this point the developer begins to delve into the depths of telecom protocols and will need to have a good background in SIP (and it's challenges) in order to establish and control calls.

The following is a possible flow for the scenario that is being discussed:

• The WebRTC client creates a connection with the SIP Gateway and registers itself. Note that the SIP Server should be modified to support WebSocket, otherwise another component (such as webrtc2sip gateway) will be needed to convert WebSocket to UDP or TCP.
• In order to make a call to a PSTN number, the WebRTC client sends a SIP INVITE request to the SIP Server. This INVITE carries the SDP local session (created earlier) containing the supported media codecs, and the ordered list of the IP and Port to be used for media transport that were established through ICE.
• The SIP Server allocates a session in the Media Gateway and sends the response to the client with the remote SDP.
• At this point the WebRTC client has all the information needed to establish a media connection between the client and the Media Gateway; it is just a matter of calling a couple of JavaScript functions to send and receive the media packages.
• The SIP Server also starts signaling with the PSTN, and as soon as the call is answered, a full duplex audio channel is established between the WebRTC client and the PSTN peer.

Conclusion

WebRTC provides great opportunity for extending many products, services and websites by allowing web developers to easily add real time communication features, but in order to realize the benefit the more challenging area of interoperability with legacy systems needs to be addressed. We can expect to get some help from off the shelf components, and we are already seeing SBCs being extended to offer support for WebRTC. In the future we may see companies coming to market with other off-the-shelf components like WebRTC to SIP gateways, which will further simplify the problem.

In our many years of experience we have found interoperability to be a continuous challenge, which in the end may put a nail in the coffin of many exciting new potential WebRTC based services. The good news is that interoperability in telecom is nothing new, and the problems and potential solutions are well understood.

If you would like to learn more, please feel free to take advantage of one of our free consultation so we can begin to help you plan your WebRTC strategy and its approach to interoperability.

Though still in its infancy, WebRTC brings great promise for transforming business communications. I recently attended the WebRTC Conference and Expo, so I thought I'd put together my top 6 take-aways for those who were not able to attend.

It was a relatively small conference attended by the industry experts who are leading the WebRTC standards, Google and Firefox engineers who are leading in the browser implementation areas, and a handful of bleeding edge vendors who have already baked WebRTC into their communications services and products. If you wanted to learn about WebRTC then this was certainly the place to be.

Among the exhibitors there was a good range of telecom and communication equipment vendors and service providers, ranging from Ericsson and Plantronics to new startups like Plivo, Drum and AddLive, together with a range of innovative product and service providers like Thrupoint, Vidtel and Twilio.

So without further ado here are my 6 key take-aways from the conference.

1. WebRTC will transform communications for businessFor a quick recap – WebRTC is a platform for communications built right into the web browser. It allows developers to develop communications applications (audio/video/data) that can be run from any website and can be used by any visitor using a standard web browser, without the need for Java or Flash plugins. For developers the application implementation for each browser, if the browser adheres to the standards, should not require any modification, meaning that one application can run across all browsers (at least as far as the WebRTC part is concerned).

Many industry experts are saying that the effect of this platform will be as game-changing for communications solutions as the WWW and web browser was to business. In reality, by embedding real-time communications into any web-based application WebRTC allows creative developers to imagine and to create many new ways for business to communicate with their customers, partners and employees. Whether WebRTC will kill the traditional carriers and even the 'traditional' OTT service providers is doubtful, but we can certainly envisage many millions of talk and video minutes being transported over WebRTC in the next few years, and unless the carriers and large service providers get creative they won't be the ones collecting the revenue on this new technology.

2. WebRTC is innovative but not CreativeWalking around the conference it was very clear that the message from the engineers was that WebRTC is definitely an innovative communication technology, and it is being provided so that creative developers can imagine and build the next generation of business communication solutions. It now requires solution providers to come up with creative new ways to engage users in innovative new business communication applications. If we look back throughout history however, we see that mankind has followed a pattern of adoption of new "medium" such as WebRTC in a rather predictable way.

The first instinct is to compare it to the thing it most closely resembles, and then to use this new medium to replicate and replace the same functionality as the old/existing medium. Only some time after that do we begin to get creative and to imagine the new applications and models that can be realized through the new medium. I like to think we have all gotten smarter, and that our learning curve is getting shorter, but as I look back at some of the recent internet history I see some of the same "mistakes" occurring now that occurred with mediums in the past (Social Media is one example). I guess my message here is that the game changing applications will be something new that we haven't seen before rather than simple replacements of existing applications.

From the exhibitors at the Conference it was clear that the creativity will definitely be present, with some exciting new solutions being shown in online meetings, unified communications and video conferencing. In the end WebRTC is a replacement of other technologies, which within their own silos of followers has already made real-time communication possible in the web (through plugins). WebRTC, with its ubiquity and standardization, should be a catalyst that will drive availability and adoption of creative new solutions in the market. This is bound to create a whole new breed of communications vendors and solutions.

3. WebRTC for Mobile is hard but worth the effortOne of the questions and one of the disappointments at the conference was the realization that WebRTC for mobile devices is not there yet. With communications solutions and business solutions in general moving very rapidly to mobile devices for delivery, it would have been natural to assume that a new disruptive communications technology would appear first in mobile devices.

There are several reasons that WebRTC for mobile is difficult to do (though not impossible). The lack of chip level support for the VP8 codec is one example of this difficulty. One of the speakers at the conference noted that running a multi-participant video conference on a mobile device, using software VP8 codec, would be a great way to melt the device. All joking aside, this is a serious problem since the standards' bodies are still not decided on the mandatory video codecs that will be specified. For mobile devices H.264 is a great choice, since it is already in virtually every mobile chipset in the market today and I don't expect to see VP8 with the same level of support in the short term.

4. WebRTC needs to integrate with existing business communication systemsAnother popular subject of interest was interoperability with existing communication standards. WebRTC will provide a great platform for many new communications applications to be embedded into websites and other web based applications, and in many cases these can be complete end-to-end solutions using WebRTC. Companies today, however, have already made a huge investment in their communication systems for customer interaction, such as call centers, CRM integration, etc., and for their own workforce.

These investments will not be thrown away, and WebRTC will not be successful if it does not integrate into these environments. From a technical perspective this isn't so much of a problem of WebRTC itself, but it is a problem to be solved in the applications and systems integration level. WebRTC does not specify any signaling protocol, but applications can implement whatever signaling they need or chose to use; the media channel is "mostly compatible" with other standards, but someone has to think about transcoding challenges to really make it work and though WebRTC uses Session Description Protocol (SDP), there will be nuances and differences as compared to other standards that will make inter working challenging.

Several solutions to this problem were present at the expo, including gateway solutions from Vidtel, as well as SBC vendors showing support for WebRTC in their devices, where such signaling and media transcoding challenges are typically addressed.

5. WebRTC is coming to a browser near youWebRTC should be available as a standard in your favorite browsers by the end of 2012, although some features such as data channels are lagging a little behind. Google Chrome, Mozilla and Opera will be the first available, and in fact all have some level of usable support for WebRTC today. Microsoft and Apple are pursuing their own paths here, but WebRTC plugins are currently available for these browsers also (although that really defeats much of the point).

6. WebRTC is not a "one size fits all"WebRTC is a great addition to the communications technology offerings list, but it's certainly not going to replace everything. In some of the sessions questions were asked about regulations and features such as 911 and lawful intercept. I'm sure these features will need to be addressed at some time int he future, but we really would be missing the point if we try to use WebRTC to replace all of our traditional communications infrastructure and services.

I believe WebRTC will have its biggest impact in customer facing web solutions that extend companies' customer service and sales capabilities. The fact that it can extend real-time communications to web browsers in a standard way will surely fuel the invention of many new and creative solutions for communication and collaboration.

SummaryFrom my perspective the WebRTC conference was one of the best investments of my time. At Daitan we've been following the emergence of this new technology and engaging in development from the beginning in the belief that it can and will have a huge impact in the development and deployment of many new communications solutions. It was great to see the early adopters at the conference (including some of our own customers) demonstrate the creative ways that WebRTC can change the world.

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Daitan Group has been helping companies realize their goals in developing and launching innovative new telecom and communications solutions. To learn more about how we might help in your initiatives please feel free to request a free consultation for us to learn more about your project and how we might help.

With the first WebRTC Conference and Expo just around the corner we thought we'd recap exactly what WebRTC is and why it's such a hot topic in the real time communications world.

So what is WebRTC?Imagine you had a real time communications engine built into every web browser that allowed any web site visitor to participate in real time audio and video communications without having to install some proprietary application that required flash, java or similar software installed in each client machine. That's the promise of WebRTC.
In the same way that the web and the browser transformed the way we do business WebRTC promises to do the same for telecom.

Who needs WebRTC?WebRTC for Enterprise
In business we all know the advantages and benefits of communication and collaboration with our employees and our customers. WebRTC will allow the enterprise to embed real time communications much more easily into all of their interfaces, both customer-facing and employee-facing.

WebRTC for Service Providers
For Service Providers, both OTT and traditional carriers, WebRTC represents a new way to reach more of their subscribers to offers them enhanced service that will drive ARPU and increase customer satisfaction. If you are providing proprietary web-based communication services today you certainly need to be planning on how to leverage WebRTC to your benefit and how to defend against an abundance of potential new market entrants.

WebRTC for Websites
With WebRTC, any website can easily provide real time communication options for its visitors. WebRTC will allow social/community sites, services sites, consumer sites, your blog and your personal sites to add rich, real time communication options that are available for use by any visitor; Imagination is the only boundary.

WebRTC for Communications Vendors
With a real time communication engine in every browser WebRTC will fuel the development of a new generation of audio, video and messaging solutions that will extend unified communications to every browser. Existing and new communication vendors and software companies will need to build expertise in WebRTC to remain competitive and to offer new solutions.

So what's the Reality of WebRTC today?
WebRTC is in active development by many companies today, both building the infrastructure into browsers and creating communications applications that leverage the new browser capabilities. WebRTC is still in draft mode, and though it is currently supported by Google Chrome, Mozilla and Opera browsers, it is not fully compatible nor fully implemented, and it is still undergoing continual change.
Safari and Internet Explorer are not yet supporting the WebRTC standards, although plug-ins are available for both browsers. Microsoft is promoting its own variations of CU-RTC-WEB.

Some things to look out for at WebRTC Conference
As we make our way around the WebRTC Conference and Expo we'll be looking for some answers that will dictate if and when WebRTC will become reality and will deliver on its promise. When will WebRTC really be available in the majority of Web Browsers? How standardized will it be? Which vendors and service providers are really going to be in the driver's seat, and more directly, how will it affect me?

If you are attending the conference we hope to get the chance to talk to you there to know how you are planning to leverage WebRTC in 2013. If not then please follow our blog for more updates from the conference.

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Daitan Group has been helping companies realize their goals in developing and launching innovative new telecom and communications solutions. To learn more about how we might help in your initiatives please feel free to request a free consultation for us to learn more about your project and how we might help.

We recently had the opportunity to talk with Paula Bernier, Executive Editor at TMC, about how Daitan has changed the outsource development model from one of lowest cost to one of highest value.

You can read her full article here but let me elaborate a little. 

So what was wrong with the old outsourcing model?Outsourcing from lowest cost countries such as India and China is certainly attractive on the surface, however several factors have led to extreme levels of dissatisfaction with this model. The major factor is the high level of competition in the IT sector that has led to high attrition rates; it's common to start a project with one team of developers only to learn that the complete team has been replaced within 6-9 months due to attrition. This has obvious implications on productivity, loss of expertise and ultimately in projects that fail to meet the goals in price and timeframe.

Daitan, located in Campinas, Brazil, has a unique advantage in this area and as a result is able to provide extreme stability to our customers that leads to high productivity and quality, hence our company tag line "Highly Reliable Outsourcing, Value Added Services Worldwide".

Why is Brazil Different?Brazil has the world's 4th largest Telecom infrastructure, and like many countries came from an environment of centralized government run telecom infrastructure (centered in Campinas). This led to Campinas becoming a center of excellence for telecom companies as well as becoming the home of the top engineering universities in Brazil. As the commercial barriers were dropped all of the large telecom vendors came to Campinas to set up development centers of excellence in order to take advantage of the large telecom opportunity that existed. While many still remain, the general global downsizing of these companies has provided a rich pool of "experienced" telecom engineers right on our doorstep.

At Daitan we take pride in our employees, all of whom are in full time employment with us. This recently led to Daitan being selected as one of the Top 100 Places to work in Brazil. Treating people well and with respect is a fundamental value to Daitan. For our customers this creates a very positive outcome of extremely low attrition rates of around 4% annually which results in higher productivity as well as continuity and alignment between our teams and with their business.

What Next?We believe this combination of highly skilled and experienced telecom engineers, along with our focus on quality, productivity and employee satisfaction delivers a unique value to our customers that results in consistent meeting of goals, high levels of satisfaction and the foundation for long-term relationships.
We are convinced that our approach offers great advantages to our current and future customers. Give us the opportunity to show you how those advantages will benefit you and your business.