WebRTC. Sonus Special Edition. by Mohan Palat and Justin Hart

WebRTC Sonus Special Edition by Mohan Palat and Justin Hart

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Foreword WebRTC started at Google in 2009 to enable browsers to have the elements of Real-Time Communications (RTC). Now, WebRTC is defining a transformation in communications that will enable 10 to 20 million JavaScript developers to build RTC into millions of websites. With WebRTC, two browsers connected to a server can have a voice or video real-time session with as little as 40 lines of JavaScript. With WebRTC, one of the peers can be a media server, or a SIP gateway enabling complex communications scenarios. The standard includes the data channel, enabling peers to send data to each other for a wide variety of uses from IM to screen and file sharing and application synchronization. With WebRTC, RTC moves from the traditional server to server model that has existed from the beginning of the PSTN to the web model where each communication event can be hosted and managed by a single server. This mirrors the transformation in information distribution that the browser, WWW, and URLs ushered in 20 years ago. The resulting web has transformed industries and created trillions of dollars of business and market value. The Web was a new model where end-users visit the site where information is and then go on to the next site. Each web event is independent, as will be the WebRTC world where a variety of very different RTC experiences and events will occur through the day. With WebRTC, communication will be added to virtually all applications and will create new applications and capabilities that will amaze all of us. WebRTC will impact enterprises, service providers, websites, gaming, and creators of new applications and services. Communications will no longer be a phone call or a separate event, but integrated into your use of the web and applications. This will change the market, business, social, and political systems in dramatic ways. This book is an introduction into WebRTC and the technology. For the corporate or service provider reader, understanding WebRTC in your environment

may be critical to your business success. For the innovator, WebRTC opens the door to new and wonderful capabilities; the next Google of Facebook is just around the corner. I look forward to talking or having a video call with WebRTC on the web with you soon. Phil Edholm WebRTC Conference & Expo, PKE Consulting LLC & UCStrategies.com pedholm@pkeconsulting.com www.pkeconsulting.com www.ucstrategies.com Twitter @PEdholm Office: 925-264-9420 Cell: 408-832-5618

Table of Contents Introduction... 1 About This Book... 1 Icons Used in This Book... 2 Chapter 1: Getting Acquainted with WebRTC.... 3 What Is WebRTC All About, Anyway?... 4 Looking into WebRTC History... 5 Why Is WebRTC Important?... 6 Understanding the Elements that Make Up WebRTC... 7 Browsers... 7 Browser API... 8 Web servers... 8 Codecs... 9 Web applications... 9 Gateways... 10 Session border controller... 10 Understanding the Data Channel... 12 Chapter 2: Integrating WebRTC into Other Networks... 13 Learning Where WebRTC Can Fit into Your Network... 14 Understanding the Need for an SBC... 15 Interworking for interoperability... 16 Securing the SIP network... 17 Navigating the firewall... 18 Managing policies... 19 Figuring out the Role of WebRTC Gateways... 20 Chapter 3: What Can WebRTC Do For Me?... 23 Calling Enterprise Contact Centers... 23 Conferencing through Video... 24 Bringing a Carrier s IMS Network to the Web... 25 Extending Over-The-Top Services... 26 Games... 27 Chapter 4: Where s The Money in WebRTC?... 29 Tapping into Licensing Revenues... 29 Selling Tools for Apps... 30

viii WebRTC For Dummies, Sonus Special Edition Making Money for Telcos and Other Service Providers... 31 Building the Infrastructure... 31 Benefiting Enterprise Users... 32 Chapter 5: Ten Must Haves When Deploying WebRTC... 33 Making a Business Case... 33 Getting Equipped with a Mic and Cam... 34 Choosing a Compatible Browser... 34 Leveraging JavaScript... 35 Connecting to a Web Server... 35 Authenticating Sessions... 36 TURNing Your Sessions Around... 36 Utilizing a WebRTC Gateway... 37 Interpreting Signals with Transcoding... 38 Connecting via SIP Trunks... 38

Introduction WebRTC the RTC part stands for real-time communications is a brand spanking new and potentially transformative technology that enables web browsers to participate in audio, video, and data communications without any plug-ins, applications, or downloads of any kind. With a WebRTC-compatible web browser, you can do things like place a call to a contact center, participate in a multiparty video conference, or engage in a screen sharing collaboration with colleagues and it just works, without requiring special hardware or client software on your computer or mobile device. WebRTC-based services open up the whole world of UC (Unified Communications, which is a suite of integrated voice, video, data, and text communications delivered via the VoIP protocol known as SIP) to anyone with a web browser and an Internet connection. About This Book WebRTC For Dummies, Sonus Special Edition, is here to help you understand what WebRTC is all about, what it takes to deploy it, and the benefits you ll gain when you make your move to WebRTC. This book isn t written at a down-in-theweeds level for telecom engineers looking for deep technical knowledge or tips for developing WebRTC applications there is a nearly limitless supply of source code and technical documentation online. Instead, you ll find that it s designed for the nontechnical folks marketers, sales professionals, finance wizards, and so on in mid- and large-sized enterprises or for service providers who want to understand WebRTC and figure out how it can help their businesses. We re going to talk a lot about voice and video communications throughout this book, but WebRTC isn t limited to just those kinds of communications. Instead, WebRTC is equally suited to communications such as screen sharing and collaboration or even to real-time gaming.

2 WebRTC For Dummies, Sonus Special Edition Icons Used in This Book This book calls out important bits of information with icons on the left margins of the page. You ll find two such icons in the book: The Tip icon points out a bit of information that aids in your understanding of a topic or provides a little bit of extra information that perhaps isn t 100 percent necessary, but which may broaden your understanding of what you ve just read. The Remember icon points out a crucial part of the text a point that you should lock away in your memory because you ll need to know it again in the future. Watch out! This information tells you to steer clear of things that may cost you big bucks, suck your time, or be bad practices.

Chapter 1 Getting Acquainted with WebRTC In This Chapter And in this corner... WebRTC Understanding why WebRTC is important Learning the components of WebRTC Meeting the Holy Grail: Data channel WebRTC Communicating with other people used to be simple; you picked up the phone and dialed their number. If you didn t know their number, you looked it up in a phone book. Then things got both better and worse. Now, you can call people from your mobile phone by finding names in a contact list and clicking them. You can email them, also from your mobile phone, or even send them a text message. At the office, you have unified communication (UC) systems that do similar feats. They enable you to see the presence (availability) of a co-worker, click on her image, and call, IM, or video with her. One thing that hasn t been simple is the ability to talk with people when you re on a website. When you visit websites, you can send people an email message (think Linked-In or Facebook). However, if you want to talk with people live, you have to do something else go to a phone or open up a Skype connection. Often those types of communications involve downloading special software to your PC or mobile phone. This practice is now changing. A new technology and group of technological standards called WebRTC lets you use phone, video, or text right from the web page you go to. You can also share screens (see the same web

4 WebRTC For Dummies, Sonus Special Edition pages or files) and all sorts of new things. You won t have to download new software. You will need the latest versions of your web browser (the versions that support WebRTC) and to be on a website that has communications programmed into it, but you won t know or care about any of that. You just go to the website and decide to communicate. Perhaps you ll be shopping and want to talk to someone about the size, or you want to show your friend and talk to her about what you re thinking about buying. Maybe you have a technical question about some home improvement project and you need to ask an expert. With WebRTC, not only will you be able to talk to the expert, but also you could show them via video exactly what you re doing. WebRTC makes communication a lot easier for businesses, customers, co-workers, and colleagues. No longer will you have to download and install a special application or download a browser plug-in. Instead, you ll just go to your communications website and communicate. What Is WebRTC All About, Anyway? WebRTC is built using javascript. This means that millions of web developers can now include communications in their websites or web-based applications. How does this happen? Check out these steps: 1. The web developer uses WebRTC codes and creates a communications button on a website. Maybe the button says click here to talk to a plumbing expert, or talk to sales or show a friend. Whatever the wording, it should make sense for the website. 2. The user clicks the button (image, icon, link) on the website. 3. The link connects the user to the web server, which then sends a webpage to the person you re calling or trying to communicate with. 4. The receiver then accepts the communication, and an audio or video chat is started between the two browser sessions.

Chapter 1: Getting Acquainted with WebRTC 5 That s it. No applications have to be downloaded by either user, no plug-ins need to be installed. WebRTC makes it possible for two users with web browsers to communicate with each other even if they re using different devices or different operating systems (MAC versus PS) or even different browsers. Looking into WebRTC History In 2011, Google was one of the main originators of WebRTC with an initial standard that was open sourced. This means that it was made available for others to freely build-on. Since then, Internet standards groups like the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C) have taken charge of the effort to advance WebRTC. A host of other software and hardware companies have joined into the effort, notably Microsoft and other browser vendors like Mozilla (the maker of Firefox). See Figure 1-1 for a view of how the Web has evolved and how WebRTC plays a part in this evolution. Figure 1-1: Evolution of the Web.

6 WebRTC For Dummies, Sonus Special Edition Why Is WebRTC Important? Businesses are constantly striving to improve the productivity of their workers and to reduce their costs. Businesses depend on communication tools to enable worker productivity the more tools and the more connected the tools are to each other, the better the productivity benefit. For example, when you can see that people are available (not busy) on instant messaging (IM) and you can send them a quick IM to ask if they can talk, it saves a lot of back and forth. One quick call may save multiple emails and other types of communication. All the different communication tools, which include voice, video, email, text, and presence, have been developed over time and often independent from one another. This means that many companies have different products to do different things. UC is the idea that all these communication tools should work together from the same interfaces and enable you to go from one mode of communication to another without switching screens and preferably at the click of button. For example, the smartphone is a UC device because you can do all these communications from the same device. Companies are now moving to upgrade or replace siloed communication tools for integrated or unified tools. Companies want to deliver to the employees and their customers the same simplified tools like you get on your smartphone. But how does WebRTC fit into this equation? Well, the beauty of WebRTC is that it takes away the requirement for specialized client hardware (devices) and software. It even removes the need for browser plug-ins. Now the browser, which all Internet connected devices already have, can be the interface for UC. Within an enterprise, this is useful and handy, but it becomes a really big deal when you think of how an enterprise communicates outside of its walls. For example, what if employees of an enterprise need to have a videoconference with key suppliers or with partners who are consulting with the company on a big project? With WebRTC and UC, they can simply send those outside parties a website URL and have them click a link and voila! they re part of the videoconference. Similarly, WebRTC makes it easier for an enterprise to communicate with customers and clients. For example, with one

Chapter 1: Getting Acquainted with WebRTC 7 click, a retail business could connect a web-browsing potential customer to a customer service agent. Or a bank could provide a link to a high-value customer s private banker right on the customer s online account page and connect the customer to the private banker immediately with one click. These instances are already possible today with UC, but they require either some client software or special browser plugins to make them work. With WebRTC, none of that s needed. As long as the user is on a compliant web browser, he s ready to connect now. And, by the way, some of the most popular browser software already supports WebRTC today. WebRTC makes accessing UC services easy, painless, and foolproof. Understanding the Elements that Make Up WebRTC Although WebRTC is easy, painless, and foolproof from the end-user s perspective, a fair amount of in the background stuff goes into making it work. In this section, we discuss what the major elements of WebRTC are all about. Browsers The foremost element of WebRTC at least from the user s perspective is the web browser. Organizations that create web browser software need to incorporate certain elements (the WebRTC API) into that browser code to allow the browser to control the elements of the computer s hardware and software required for WebRTC communications. Several of the most popular web browsers are already supporting WebRTC, including the latest versions of the following: Google Chrome Mozilla Firefox Opera Not yet on this list, but worth mentioning is Microsoft Internet Explorer (IE). Microsoft has publicly declared that it will be supporting WebRTC, so you can safely expect that a notso- distant future version of IE will incorporate WebRTC support.

8 WebRTC For Dummies, Sonus Special Edition WebRTC is built on top of HTML5 the latest version of the HTML markup language used to present web content and JavaScript, the programming language used to create many client-side scripts allowing interactive web apps to be used by web browsers. HTML5 and JavaScript both together and separately allow browsers to present things such as audio and video interactively, much like the ubiquitous Adobe Flash browser plug-in did before the advent of HTML5. WebRTC extends the capabilities of the HTML5/JavaScript duo into real-time communications. Browser API Web browsers support WebRTC by implementing specific WebRTC application programming interfaces (API) that enable hardware control and communications through the browser. The key APIs for WebRTC-compliant browsers are the following: getusermedia (or gum): Allows the browser to control video and audio hardware on the computer, such as the webcam and microphone. PeerConnection: Helps to setup the WebRTC communications, much like touchtone dialing did for traditional analog phone calls. PeerConnection goes beyond making the connection, however; it s also involved in elements of the call like choosing codecs, applying encryption, and more. DataChannels: DataChannels is the data portion of the WebRTC communication. DataChannels provides a peer-to-peer, decentralized data connection between the browsers over which data can flow directly between the points in real-time. We talk more about DataChannels in the final section of this chapter. Web servers There s another server element to WebRTC and that s just a web server which can host the WebRTC APIs. This server will (ahem) serve up the clickable URLs in a web page saying call now or click here for a video chat that a WebRTC user interacts with when placing a call. This web server could be

Chapter 1: Getting Acquainted with WebRTC 9 integrated into a gateway or it could be part of an enterprise or service provider s existing web server. Codecs Codecs (or coder-decoders) are the computer programs used to encode a signal for storage and transmission and to then decode that same signal at the far end for display or playback. Specifically, for WebRTC, codecs are the software that encodes audio and video for transmission at one end and then decodes that data at the far end so that the user can listen to and view it on his or her computer. The digital audio or video content you interact with be it a call on your mobile phone, a YouTube video you re watching on your ipad, or the digital HDTV signal that s displayed on your big screen TV uses a codec for the encoding and decoding process. In WebRTC, several official codecs for audio and video are involved: Two audio codecs: Opus G.711 A video codec known as VP8 These codecs were defined by Google when they released WebRTC as an open source project. There s always a bit of controversy around codecs in the industry (regarding royalties and permissions, and so on), so you should fully expect to see other vendors, such as Microsoft, proposing different codecs when they release their own implementations of WebRTC. Web applications WebRTC includes a Web API designed to allow developers to create their own WebRTC-enabled applications. An organization implementing WebRTC can create its own applications or purchase them from software vendors. There isn t just one WebRTC videoconferencing application or one WebRTC collaboration application. Instead, third-party organizations can create their own WebRTC applications by using the Web API and WebRTC specifications.

10 WebRTC For Dummies, Sonus Special Edition Gateways When a company wants to connect a user on its webpage to an employee working from his desk, it can set up a WebRTC to WebRTC (or peer to peer) communication or integrate the WebRTC communication with its existing backend communication system. Most corporate communication systems use session initiation protocol (SIP) to connect to their audio and video systems. Therefore, to connect a WebRTC communication to a corporate PBX or video system, the WebRTC communication has to be translated into SIP. The device that makes the connection between WebRTC and an organization s UC network is called a gateway. The gateway bridges the divide between WebRTC and the VoIP network inside an organization (or inside a service provider, if the VoIP network is provided by a third party). The gateway does this by performing several functions, including the following: Translating between SIP and WebRTC Transcoding and transrating traffic Interworking We talk about these functions and those of the SBC, discussed in the next section, in more detail in Chapter 2. Session border controller Another component of corporate communication systems is a session border controller (SBC). The SBC is a device deployed on the border of enterprise and service provider VoIP networks where the internal part of the network connects to outside networks. The SBC has several key functions in the network, including the following: Securing the network: SBCs are designed to protect enterprise and service provider networks from a variety of potentially malicious attacks (like Denial of Service attacks) as well as service theft and spoofing (where a user deliberately modifies her identification on the network). In a WebRTC environment, this feature secures

Chapter 1: Getting Acquainted with WebRTC 11 an organization s internal VoIP network from web-based attacks coming from WebRTC users on the network. Providing interworking: SBCs provide substantial interworking capabilities, which smooth over protocol differences and vendor-to-vendor incompatibilities so communications simply work. For WebRTC, this means that the SBC can help in the process of converting WebRTC VoIP traffic to the SIP or other VoIP system used within an enterprise or service provider network. Controlling call admissions to the network: The SBC is the gate guard for an enterprise or service provider network determining which calls and communications sessions are allowed onto the network and which aren t. This can include the use of blacklists of calls to always be denied and whitelists of calls that are to always be allowed. When traffic volumes are particularly high, this functionality additionally helps to make the network more secure and reliable. An enterprise can therefore use an SBC to help ensure that only legitimate calls from WebRTC users are entering their VoIP network. Transcoding and transrating: The SBC can transcode between codecs and adjust the bit rate as required by the network capabilities of both ends of a WebRTC call. WebRTC may use different codecs than an enterprise or service provider VoIP network. The SBC can do on-the-fly conversion between codecs so both sides can talk to each other. Dealing with network topology: SBCs are really good at getting calls through the actual endpoint within an enterprise network that they re supposed to reach. This is a pretty big deal when you look at networks that use things like Network Address Translation (NAT), which uses private IP addresses for all the computers inside the network and a single public IP address facing the outside world. SBCs can provide NAT Traversal so calls can easily reach the right computer in the network. The SBC can help connect calls from WebRTC users within an enterprise or service provider VoIP network, no matter how that network is set up.

12 WebRTC For Dummies, Sonus Special Edition SBCs can also hide the topology of an enterprise network from the outside world. The result is a network that s easily accessible to clients for making and receiving calls, but where the innards of the network are effectively invisible, which makes them less vulnerable to attack. Some of this list of features overlaps with the features of a gateway device (see the preceding section). That s okay sometimes the gateway may even be integrated with the SBC. In other cases, some functions may be assigned to the gateway and others to the SBC depending on the exact configuration, topology, and services supported by the network. Understanding the Data Channel A particularly neat feature supported by WebRTC is the DataChannel API. DataChannel provides a peer-to-peer connection meaning that there s no centralized server controlling the flow of data between browsers for data beyond just audio and video. A big advantage of cutting out the middleman the server and going peer-to-peer is that connections can be extremely fast and low-latency (meaning that there s very little delay in the transmission of data between two browers). This opens up all sorts of possibilities for WebRTC, in terms of services like web collaboration and screensharing, gaming, file and document sharing, and more. In other words, WebRTC, through the DataChannel, supports more than just voice and video, right in the web browser. DataChannel opens up a huge range of new applications without requiring client software. The DataChannel supports two types of connections: Reliable channels: These channels provide the highest level of assurance assuring that data is received at the far end of the connection and retransmitting it when it isn t. With a reliable channel connection, data will be received in the order it was sent and potentially delayed if a retransmission is required. Unreliable channels: These channels limit the retransmissions and don t necessarily ensure that data is received in the order it was transmitted.

Chapter 2 Integrating WebRTC into Other Networks In This Chapter Learning where you can connect with WebRTC Leveraging an SBC for interworking and security Utilizing a WebRTC gateway In Chapter 1, we introduce WebRTC and cover some of its basic building blocks. These building blocks help you understand how WebRTC provides voice, video, and other communication services directly from a standard web browser. In this chapter, we talk about how WebRTC integrates with the existing VoIP or Unified Communications (UC) network that an enterprise may already have in place, or with the network that a service provider may offer its customers. It is, of course, possible for WebRTC to be used for browserto-browser communications bypassing the existing networks of an enterprise or service provider but it s very likely that an organization will choose to integrate WebRTC into its existing infrastructure. Doing so integrating WebRTC with an existing network can provide a host of benefits by greatly simplifying the ability of people outside the network to call, conference, and video chat with people inside the network, using nothing more than a standard web browser. Integrating WebRTC with existing networks isn t just a useful way to expand the reach of those existing networks; it s also a way to extend their service life and keep them from becoming obsolete. Companies that have invested huge amounts of

14 WebRTC For Dummies, Sonus Special Edition money in their VoIP networks can easily extend the useful life of those networks by integrating with WebRTC as it becomes more prominent. There s no need to rip and replace. Instead, you just augment existing networks to support WebRTC calls. Learning Where WebRTC Can Fit into Your Network Before we begin digging into the bits and pieces required to integrate WebRTC into an existing VoIP network, it s worth looking at what services this integration can provide. An enterprise or service provider can choose to integrate WebRTC services into its existing networks to do the following: Integrate into an existing SIP-based network within an enterprise: The most obvious benefit is to add WebRTC to an existing call center infrastructure. Integration allows anyone with a browser and an Internet connection to contact a company s customer service or sales department without needing to know phone numbers or having to download a specific application or plug-in. Integrate with the PSTN: Another good use case is to enable employees who aren t at a work location to use a WebRTC interface to make phone and video calls. This practice saves on roaming charges from cellphones or at the very least allows these calls to be made from the less expensive SIP trunks used by the company. Using a WebRTC interface means that remote workers don t have to download and install softphone clients on their PCs or mobile phones and worry about computer settings and compatibility. Instead, it allows the enterprise to save money not only on the cost of the calls but also on the cost of providing phone lines to remote workers. Providing easy external access to a UC system: Enterprises that have deployed a UC platform (like Microsoft Lync) can easily expand the reach of that platform by integrating it with WebRTC. This allows remote users, casual users (for example, employees who are away from their normal devices), as well as partners/

Chapter 2: Integrating WebRTC into Other Networks 15 customers/clients to access a rich range of UC services (like video, collaboration, and screen sharing) through a simple web browser interface. This can greatly reduce the cost and complexity of providing access for these types of connections. Providing access to a telco services: A telco service provider can leverage (and integrate) WebRTC to provide access to its portfolio of voice, video, and related services to a broader range of customers who aren t on the telco s network. For instance, a telco could provide special calling or video services (like discounted international services) from WebRTC clients to phones on the remote end of the call without requiring the customer to be a current subscriber to the telco s services. Additionally, a telco could provide WebRTC access as part of a hosted and managed cloud UC or VoIP service for enterprises. Providing access to hosted UC systems: Service providers could use WebRTC to extend access to the applications and services they run in their cloud so users can take advantage of those communication features from anywhere. Understanding the Need for an SBC The session border controller (SBC, covered in Chapter 1) is a mainstay in enterprise and service provider VoIP and UC networks, providing a differentiation between internal and external networks by performing the following tasks: Securing the network by Protecting against Denial of Service attacks Hiding the topology of the internal network from external view Encrypting traffic so it can t be intercepted Guarding against spoofing attacks

16 WebRTC For Dummies, Sonus Special Edition Providing interoperability by Transcoding and transrating media Normalizing (or translating between) different variants of SIP Translating between VoIP protocols like SIP and H.323 Helping IPv4 and IPv6 networks work together seamlessly Implementing and enforcing network policies (we talk about this more below, in the section titled Managing Policies ) Providing a platform for creating new services Enterprises and service providers are very likely already using an SBC for these functions. The good news is that SBCs can bring a lot of these functionalities over to the WebRTC world by acting at the point where WebRTC calls interface with the VoIP or UC network being served by the SBC. In the remainder of this section, we discuss where and how WebRTC and SBCs specifically work together in this environment. You don t need an SBC for a WebRTC service that s just being served over the web between two end points that are both using WebRTC via their web browsers (for example, a gaming application between two PCs). In the enterprise and service provider world, however, where WebRTC is being integrated with an existing SIP or UC network, the SBC has a vital role to play. Interworking for interoperability One of the biggest roles that an SBC plays in a WebRTC integration into an existing SIP or UC network is in facilitating the interworking allowing disparate systems or protocols to communicate, connect, and exchange data. WebRTC and enterprise or service provider VoIP/UC networks are both technically VoIP, but they re not the same. Although most UC networks utilize the session initiation protocol (SIP), WebRTC uses well, WebRTC. And most SIP UC systems use different audio and video codecs (defined in Chapter 1 if you need a refresher) than those used by WebRTC.

Chapter 2: Integrating WebRTC into Other Networks 17 For a WebRTC client (browser) to talk to a client (like an IP phone, or a smartphone/pc client software system) of a UC system, something needs to make them speak the same language. The SBC can do that, by providing Protocol interworking/translation: The SBC can convert between the signaling protocols of WebRTC and SIP allowing calls to be seamlessly connected and disconnected and can even provide connectivity to the TDM PSTN (public switched telephone network). This functionality can also be provided by the WebRTC gateway device, which we discuss in the final part of this chapter. The gateway can be a separate device or integrated into the SBC. Transcoding: On-the-fly codec conversion between WebRTC and SIP system audio and video codecs for example, translating the WebRTC Opus codec to a standard SIP UC coded like G.711. Transrating: The SBC can also transrate (or modify the bit rate and required bandwidth) of calls, as dictated by network demands. So if a WebRTC call is placed from a computer on a gigabit Ethernet connection and ultimately ends up on a mobile device with limited 3G bandwidth, the SBC can automatically reduce the bandwidth of the call to fit the available pipe on both ends of the call. Securing the SIP network WebRTC voice and video calls are placed and received from web browsers on PCs and other devices (like tablets or smartphones) connected to the web. We suspect that you re aware that there are constant security threats involved in such an environment there s a reason you have to update your various virus and malware programs on a weekly or greater basis, after all. There exists, therefore, a possibility that a compromised system will attempt to connect to an enterprise or service provider s SIP network via WebRTC. This scares the dickens out of IT security professionals who are considering scenarios like opening up their SIP-based call centers to WebRTC calls and in the process, imagining scenarios where a compromised PC is connecting with their contact center.

18 WebRTC For Dummies, Sonus Special Edition The SBC can minimize this risk by doing the following: Protection against Denial of Service (DoS) attacks, where malicious actors attempt to overwhelm a SIP system with a sheer overload of connection attempts Providing call entry control via whitelists and blacklists and policies (discussed in the next section) to allow only authorized calls to be placed within the SIP network Denying access to rogue WebRTC requests Hiding the topology of the SIP network (the internal network being protected by the enterprise or service provider s SBC) from WebRTC clients and the wider web Navigating the firewall Another network component that must be added to a WebRTC/non-WebRTC communications path is technology to enable the signaling information to pass through the firewall. Firewalls provide Internet and data security (not unlike the role an SBC plays in a SIP communications environment). If a person on the outside of a corporate firewall needs to communicate with someone inside the firewall, he has to exchange information, such as the port address of the firewall and the IP address. This exchange allows the media to move from one location to another. Interactive connectivity establishment (ICE) is a standard, secure procedure that defines the method for determining the public IP address and PORT information for external and internal users. To do this, ICE uses two protocols: Session transversal utilities for NAT (STUN) Traversal using relay NAT (TURN) If neither WebRTC user is sitting behind a firewall, ICE isn t required. However, wherever a firewall exists, ICE/STUN/TURN is required to enable communications. Many free or shareware based ICE/STUN/TURN applications are available on the market today. These capabilities are also frequently built into gateways and SBCs. Users don t have to worry about this, though. The process is done behind the

Chapter 2: Integrating WebRTC into Other Networks 19 scenes by the communications and network teams at the company or service provider/telco when the WebRTC-based services are set up. Managing policies Enterprise and service provider VoIP/UC networks are built on top of policies. What we mean by that is that these networks aren t free for alls where anything goes and every client can use every service whenever they want to. Like any network, VoIP/UC networks aren t limitless resources; policies exist to determine who can access what resources and when. One of the primary guardians of network policy is the SBC. The SBC is the traffic cop of a SIP/UC network, providing call access control determining which calls are allowed on and off the network in the first place. There s more to policy than this, however. Policy also includes decisions on things like Bandwidth utilization and rate limiting Least cost routing for toll calls Application availability Media paths and routing WebRTC, because of its low-latency peer-to-peer nature, can allow rich, high-bandwidth multimedia communications. Policy is what determines how that demanding media will be handled when it crosses the border into an enterprise or service provider s VoIP network. Policy also provides the ability to maintain SLAs (service level agreements), which determine the minimum quality levels that should be maintained in a VoIP network. Finally, policy helps determine what happens when the SIP network gets congested for example, when there s a great surge in calls (maybe your marketing campaign was too successful, or there s a run on the bank!). Policy, as implemented through the SBC, can determine if certain services need to be throttled or turned off or if overages of calls need to be directed elsewhere.

20 WebRTC For Dummies, Sonus Special Edition Figuring out the Role of WebRTC Gateways If the SBC is the traffic cop securing the border between an enterprise or service provider VoIP network and WebRTC clients, the WebRTC gateway is the bridge between the two network segments allowing communications to flow between the two. There s really only one role for a WebRTC gateway though it is a big one: interworking. The WebRTC gateway can be a standalone device or it can be integrated into an SBC. There s a bit of a thin line between the two the way we suggest you remember it is to think of the SBC as having a superset of capabilities, while the gateway has a subset focused mainly on interworking. Figure 2-1 shows the configuration of an enterprise network with the SBC and the WebRTC gateway intermediating between WebRTC and an enterprise s internal networks. Figure 2-1: The gateway and SBC in an enterprise network. A WebRTC gateway will, bi-directionally, convert between WebRTC signals coming from the web to SIP (or other VoIP protocols) so calls can be connected, carried, and discontinued when they re over. This entails converting between

Chapter 2: Integrating WebRTC into Other Networks 21 WebRTC and SIP and dealing with the basic routing of the signaling and media parts of calls across these disparate VoIP network segments. If this is at all confusing, perhaps a more familiar analogue would work better. Where VoIP networks interact with the TDM PSTN, there is a device known as a VoIP gateway. The VoIP gateway s main function is to convert TDM calls into SIP calls by manipulating the signaling (touchtone dialing to SIP) and media (TDM voice to VoIP packets). The WebRTC gateway does the same thing, only it converts between WebRTC and SIP.

22 WebRTC For Dummies, Sonus Special Edition

Chapter 3 What Can WebRTC Do For Me? In This Chapter Connecting contact centers to customers Making video conferencing easy Extending IMS networks for service providers Expanding the reach of OTT services Getting into gaming This chapter is the fun one. That s because we talk about what you can do with WebRTC. We dig into actual use cases that show the power of WebRTC. We think that you ll get excited we know we did writing it. WebRTC has the potential to be a real game changer. Calling Enterprise Contact Centers As far as enterprise applications for WebRTC go, this one is really a no-brainer. Almost all mid- and large-sized enterprises (as well as government organizations and the like) have long ago adopted SIP-based VoIP systems for their contact centers. Whether it be customer service, inbound or outbound sales, or any customer-facing activity, chances are good that it s currently going to a SIP-connected call center. The reason why is that it s simply much cheaper to operate a call center in this environment (compared to a traditional TDM-based environment), and it s much easier to create a geographically dispersed yet unified call center with VoIP.

24 WebRTC For Dummies, Sonus Special Edition For the past few years, enterprises have begun experimenting with click to call features that expand the reach of call centers to people who are browsing the enterprise s website. Before WebRTC, however, this approach wasn t always the smoothest process for the end-user requiring steps like entering a call back number or downloading a browser plugin (and then authorizing that plug-in permission to access PC resources). WebRTC takes that complexity away. A customer or potential customer with a compliant browser (see Chapter 1 for a refresher on this, if needed) can simply click on a link and be instantly audio/video connected to an enterprise. No extra steps, no software to download. Yep, we said audio/video connected. WebRTC makes it easy to add a video chat component to any contact center, which may have an impact on a business dress code but could also have a (positive) impact on its bottom line and customer satisfaction. Conferencing through Video Many enterprises have adopted videoconferencing as a way to deal with the increasingly distributed workforce they face. Whether it s supporting teleworkers, remote offices, roadwarrior sales teams, or the newly acquired division in Tokyo, videoconferencing allows a far richer, more immersive collaborative experience. (We ve even seen a company have some fun with this, taping a video conferencing-equipped ipad to the head of a dummy seated at the conference table!) Video conferencing has come a long way in a short time after many years of being the next big thing, it s now commonplace in many businesses. There s been a process that s taken video mainstream from the ultra-expensive (often $100,000 or more) conferencing systems to integrated webcams in just about every PC, smartphone, or tablet on the market. But there s still a snag with videoconferencing compatibility. It s easy to set up everyone in a company on a standard UC platform that supports video like Microsoft Lync. It s hard, however, when you realize that you need to videoconference with consultants, partners, suppliers, and others who

Chapter 3: What Can WebRTC Do For Me? 25 have their own IT infrastructure and their own unified communications platforms. WebRTC takes that problem right out of the picture: Internally, you can use whatever platform you want (including, we should add, WebRTC video conferencing web apps), while you can provide external parties with a clickable URL to join a conference right from their compliant browsers. WebRTC also reduces the costs by enabling users to video conference with everyone else because companies don t have to buy special devices or systems. Bringing a Carrier s IMS Network to the Web Telephone service providers (or telcos) have spent a good part of the past decade developing an architectural framework for their networks known as IP Multimedia Subsystem (IMS). (Yeah, it s not the sexiest name ever, but it s important.) IMS is designed to facilitate the provisioning and delivery of IP services (like VoIP) and future services (whether that be mixed media like messaging and voice combined in a single session) for telcos using standards-based protocols such as SIP and DIAMETER. IMS is the basis of many of the 4G LTE services being offered to telcos mobile customers today, as well as IP-based services to wireline broadband customers. IMS-based services utilize IP standards like SIP, so in a lot of ways, they re similar to enterprise UC services. IMS is designed to control SIP sessions and to intermediate between users attached to the access network and the applications servers that deliver the services they want to use. With an IMS-WebRTC integration, many of the services offered to telco endpoints (like IP phones in a home or office or smartphones and tablets in the mobile environment) can be extended to any web browser anywhere in the world. For example, 3G and 4G LTE phones can use SIP-based IP calling (as opposed to GSM or CDMA calling on older generations

26 WebRTC For Dummies, Sonus Special Edition of mobile phones). With a WebRTC dialing browser API, you could place calls to mobile phones directly from a browser, without any PC application or plug-in to download. Extending Over-The-Top Services One of the more interesting telecommunications trends of the past five or ten years has been the proliferation of over-thetop (OTT) services. OTT services, if you re not familiar with the term, are those that are provided to end-users over their broadband Internet connection (wireline or mobile) but not provided by their Internet service provider. For example, a mobile or fixed line voice service offered by a telco or ISP is not an OTT service; Skype is. Similarly, broadcast and on-demand video services offered by a cable or telephone company aren t OTT services; Netflix streaming video is. There are hundreds and hundreds of over-the-top services on the market today, serving both consumer and business marketplaces. Many perhaps even most OTT services are voice-, audio-, or video-centric. One thing they all tend to have in common (in addition, of course, to being delivered over-the-top of an existing Internet connection) is that they require some sort of software download. Skype requires an app download to a PC or smartphone/tablet. Netflix requires an app on mobile devices or an installation of Microsoft Silverlight to work on PC web browsers. And once you require downloads and installations, you also require users to install updates and keep doing so over time. What a drag. With WebRTC, there s no need for extra applications and there s no need for upgrades and keeping systems up-to-date. Nothing to install, nothing to update, nothing to download it just works. This lets existing OTT service providers improve their offerings making them accessible to a wider audience. It also provides (perhaps incorporating the WebRTC to IMS extension) an opportunity for telcos to effectively compete with the OTT service providers who ve been competing for their revenue. In a WebRTC environment, a telco could quickly and easily build OTT-like applications, deliver them via existing web