As discussed in the September cover story of this magazine, the sun is setting on the PSTN and narrowband voice networks, while the WebRTC ecosystem and the number of solutions based on the new technology are growing.
Already, WebRTC is supported on more than 1 billion endpoints, says Google, one of a raft of important tech companies driving this new technology. And Disruptive Analysis expects that to grow to 3.9 billion by 2016.
Here’s our roundup of some of the companies offering solutions in the WebRTC space.
AddLive is a WebRTC platform that allows developers to add live video and voice communications to their applications. AddLive offers a full RTC stack that includes SDKs for all major platforms including WebRTC browsers, non-WebRTC browsers (via a plugin), native mobile applications and native desktop applications.The AddLive cloud infrastructure provides reliable and secure multipoint control units for multiparty sessions; and firewall and NAT traversal infrastructure. AddLive clients include Citrix, Genband and Bratton Technologies. It is deployed to more than 20,000 businesses through our customers’ applications.
ACONYX Unify is a mobile contextual communications and collaboration solution built on the latest WebRTC and mobile technologies that empowers service providers to support enterprise mobility and BYOD for enterprise customers. ACONYX Unify functions as a cloud-based enterprise communication and telephony application server with a rich set of components: Business Mobile Connect (IP Centrex-based voice and video calling, auto attendant, and FMC), Converged Messaging (single unified mailbox for text, voicemail, videomail, fax), and Multimedia Conferencing & Collaboration (audio, HD video and text conferencing for LTE (News - Alert) operators). The solution provides a customizable rich communications client for iOS/Android (News - Alert)-based mobile devices and a web client for PCs and Macs that utilizes HTML5 and WebRTC technology to provide SIP-based multimedia communication. Contextual communications are delivered through rich in-call media sharing capabilities as well as through integration with external systems such as Salesforce CRM, Dropbox, Google Drive, etc. Developed on the proven ACONYX CAS platform, ACONYX Unify delivers carrier-grade high availability and scalability. ACONYX CAS is a multi-protocol next generation converged platform which has a robust and flexible architecture with extension points to integrate with other subsystems and network components. With ACONYX Unify, service providers are able to monetize their network investments by offering revenue-generating multimedia services that increase ARPU and enhance service differentiation.
Today, web applications often run over the top of the carrier network and do not take into account its capabilities. Alcatel-Lucent WebRTC helps service providers deliver carrier-grade communications as a web experience and makes it easy to embed real-time communications into applications, websites and browsers. Building on the WebRTC standard, the Alcatel-Lucent WebRTC border controller is implemented on its own field-proven IP Border Controller. Purpose-built for multimedia and naturally supporting the web, the IBC scales readily to support growing web and VoLTE traffic volumes. Additionally, Alcatel-Lucent WebRTC-verified IMS provides a cloud communication platform that includes critical functions like session logic, identity management, routing, security, transcoding, interconnection, operations and regulatory support. The fundamental openness and reach of WebRTC helps service providers gain traction with web developers. To capitalize on the full potential of WebRTC and fast-track innovation on device and application sides, the solution provides complementary (WebRTC) client and (OMA Network) server New Conversation APIs, allowing the creation and delivery of new web-scaled communication services and clients faster and cheaper. Carriers and developers can freely join Alcatel-Lucent’s web developer portal to develop, test and taste the new flavors of cloud communication services.
ANYMEETING is an all-in-one conferencing tool designed and priced for small business. It serves all the conferencing needs of small businesses – from a web, video and phone conference to a 200-person webinar. ANYMEETING is also one of the first companies to apply WebRTC to a full-featured web conferencing product. With ANYMEETING SaaS-based service, meeting hosts are able to invite up to 200 attendees per meeting with no time limits, and enjoy a full range of features including integrated videoconferencing, phone conferencing, screen sharing, presentation sharing, recording, file sharing, custom branding, meeting notes, registration forms, social media integration and more. The ANYMEETING WebRTC-enabled platform delivers exceptional audio performance through best in class echo cancellation, low latency response times, and high quality audio codecs. It is currently available to users of Chrome 27+ (with Firefox coming soon), while supporting older browsers through Flash technology.
Apizee develops WebRTC instant messaging, audio and video cloud communications solutions for integration in applications and websites. Telephony and videoconferencing are available from a web browser on any devices. Apizee real-time communications SaaS platform allows developers to add audio and video calling functionality without any knowledge on VoIP and without investment in any specific equipment. This is a simple way for our customers to take advantage of real-time communication and to focus on their business application specificities. Apizee solutions are available through simple API to accelerate integration by our partners (web agencies, SaaS applications developers, web stores, integrators and operators). In addition to the API, Apizee provides packaged modules for easy integration without any knowledge requirements on software development. IzeeChat is a customer's relationship module to add chat, audio and video on websites to improve sales conversion rates. Izeelink is a ready-to-use module to add unified communication between connected users in intranet or web applications.
WebRTC brings voice, video and data communication to the browser and provides web developers with no VoIP experience easy access to VoIP communication technology. When required to bridge WebRTC communication with existing enterprise communication platforms there is a need for native support for WebRTC media technology in the end user equipment in the enterprise. Support for the WebRTC voice Opus codec and encryption algorithms natively on the enterprise IP phone yield better call quality, privacy, scalability, reduced cost and an easier migration to multi-purpose cloud platforms. In support of this strategy, AudioCodes recently demonstrated two product offerings. The demonstration showed a click-to-call web browser to IP phone contact center simulation using the wid-band Opus codec. Product demonstrated included the AudioCodes 440HD SIP Phone, a multi-line executive IP phone supporting the Opus Codec, and the AudioCodes Mediant 800 Session Border Controller with WebSockets, terminating the WebRTC call control and integrating SIP session management with the above IP phone.
Bistri is the new way to make video calls. Bistri provides everyone with their own link, like an online phone number. Your friends can chat, send some files, and make a video call with you for free, even if they are on a web browser on a mobile with no Bistri account. Based on the WebRTC standard, part of HTML5, Bistri is available on the web, on Android, on Chrome OS, on Surface Tablets, on iOS and Linux. Bistri Platform is now open to help any developer or company to implement WebRTC features. Bistri SDKs provide an easy way to the developers to integrate video calling, videoconferencing, screen sharing, chat, and P2P data in any kind of website, game, application, even on Google Glass. Bistri Widgets for Wordpress make the integration as simple as a few clicks on a form.
Blue Jeans Network offers a cloud-based video collaboration service requiring no additional hardware or software. All parties can connect to the same Blue Jeans meeting using their platform of choice. This includes mobile phones or tablets using the Blue Jeans iOS, the Android app, or third-party solutions on Android. Users can also join through their browser, meaning that all they require is a computer and a webcam. Dual-stream support ensures the quality of both the video feed and shared content. Blue Jeans meetings can also be recorded including the audio, video, and content being shared. All recordings are stored in the cloud so they can be accessed from anywhere. Using Firewall/NAT traversal and encrypted meetings, even in a multi-vendor environment, participants can meet with privacy and confidence. Blue Jeans also supports single sign-on and integrates into Outlook or Google Calendars for easy scheduling. Blue Jeans began using WebRTC in June 2012, when it released a WebRTC-based browser connection to the Blue Jeans videoconferencing service. To overcome current WebRTC limitations (WebRTC was still in beta at the time), Blue Jeans made several enhancements allowing participants to join high quality video meetings from any browser including IE, Safari, Firefox and Chrome.
Brook-Pro offers a complete line of products to support spontaneous team collaboration. A family of scalable WorkPoints, from laptops to oversized multi-screen room systems, facilitates HD team collaboration regardless of distance or endpoints, by providing HD collaboration meeting rooms in-a-box functionality much like working in the same physical conference room. Video images, documents, files or applications originating from local PCs, or streamed by auxiliary cameras, whiteboards, and video recordings can be shared by WorkPoint users with other participants without delay nor any need for IT facilitation. Separately, subscriptions to Virtual Meeting Rooms from the BPRO-Invite series provide individuals the freedom of hosting HD video meetings at any time and any place, even while travelling. The BPRO.Net cloud provides participants global reach and flexibility of choice: WebRTC calls, as well as Skype, Lync, Jabber, H.264, H.363, are accepted simultaneously in Brook-Pro VMRs and WorkPoint sessions. Brook-Pro.net, a global public/private video exchange, allows free registration for WebRTC end user clients as well as multi-corporate tenant hosting. Customer relationship management augmented by integrated HD Brook-Pro WorkPoint and VMR support, provide the highest quality customer engagement in a number of verticals.
Commodisee is a web platform that serves as a retail store extension, enabling shoppers to enter a retail store from home and visually interact with the store's staff. Commodisee utilizes Google's WebRTC technology + SIP over web socket to enable high quality audio-video communication between sellers and shoppers. What makes this platform different from other third-party video chat software is privacy – no registration or logging in is needed for the shopper to enter a store and shop. And Commodisee does not stream the video from the shopper's end, keeps his privacy, and makes him even more comfortable to shop through this platform. Because no extra software is needed, this platform is very spontaneous for browsing and shopping, just like in a live mall. With Google's VPX codec and under normal conditions, you get a very nice, high quality video stream (up to 720p).
Dialogic’s PowerMedia XMS is a software media server that addresses many of the greatest challenges facing WebRTC deployment for application developers, enterprises, integrators and telecom carriers. It efficiently mixes diverse media streams, enabling any-to-any network connectivity and cloud-based services. It connects legacy networks, mobile and Internet endpoints and WebRTC to simplify multi-party communications and collaboration. Companies leverage PowerMedia XMS to energize application delivery by boosting virtualization performance and cloud delivery through a software-based model, offering proven scalability in telco environments and a supercharged integration to WebRTC. Business logic of applications deployed on SIP application servers and web application servers control PowerMedia XMS to execute high-density, multimedia functions including inbound and outbound session and call control, audio and video play and record, transcoding, transrating, transizing of video streams, conferencing, content streaming and advanced supporting functions for communication sessions. Supporting standard media control interfaces such as MSML, VXML, NetAnn, MRCP, JSR 309 and RESTful API, PowerMedia XMS does the legwork in both mobile and broadband environments.
Asterisk (News - Alert), the world’s most widely deployed open source communications platform, is used as a media server, PBX, and protocol gateway. For many years Asterisk has provided the foundation for everything from enterprise call centers to carrier applications. Since 2012 WebRTC support within Asterisk has included SIP transport over WebSockets; SRTP; ICE, STUN, and TURN for NAT transversal; G.711a/u transcoding; H.264 pass-through; and a built-in mini HTTP server to provide web content. In 2013, pass-through support for Opus and VP8 was also added to Asterisk. Three key Asterisk WebRTC roles are acting as a gateway between WebRTC and other technologies like SIP, analog, PRI, BRI, or IAX2; providing media services such as prompts, IVR, and conferencing functionality; and routing of traffic based on customizable criteria. Free to download and deploy, Asterisk provides a low barrier to entry. A proven track record and large global community has made Asterisk the platform of choice for a growing number of WebRTC deployments.
Meet your new intelligent digital agent. She can engage, educate, evaluate and help customers in real time. Through Q&A, she can deliver the right message at the right time through your videos. DilogR is your automated companion to WebRTC that allows you to engage customers in a virtual conversation. It’s the modern version of the automated phone system, but with video. It can be used in conjunction with WebRTC in multiple ways including before getting to a live person or during a call to show videos and slides that help communicate information to a caller. With DilogR, you can field sales inquiries with a virtual salesman, support inquiries with a video FAQ, and then give the viewer the option to immediately connect to a live person through WebRTC. And the agent can get all of the information communicated through the virtual agent so that she can immediately jump into the sale, support issue, etc. While you’re solving your customer’s needs, you’re gathering real-time analytics and creating up-sell opportunities. With DilogR, you can reduce sales and support expenses, equalize call center load, and keep current on complex issues without constant training.
Flashphoner Web Call Server 3 is a middleware platform for WebRTC, Flash, SIP audio and video calls, and instant messaging. The developer is the company Flashphoner LLC.
Main features include voice and video calls, browser-to-browser applications, and browser-SIP. Browser-PSTN and browser-GSM calls are also available if a SIP provider or VoIP equipment allow calls to PSTN and GSM phones. Two-way calls are available; you can call to a webpage open in a browser directly from a landline phone or SIP phone.
Additional features include holding and transferring calls, instant messaging, DTMF, and additional codecs. For calls from browsers that do not support WebRTC, Web Call Server is compatible with Adobe Flash Player, which is installed by default in most web browsers. The product includes two browser applications Webphone and Click-to-Call. It has open source code to allow users to easily change its look and feel while integrating the Webphone into an existing web interface. The license is for a lifetime. The cost depends on the number of simultaneous calls and the set of features, and varies between $26 to $500 for one audio line or from $28 to $900 for one audio/video line with additional options included.
FreeCRM.com is the first cloud CRM to offer click-to-call powered by WebRTC. Simply right-click on any phone number in FreeCRM.com and you can call any U.S. phone number instantly using just your computer and a WebRTC-powered browser. FreeCRM.com keeps track of the call and pops up a call information screen and has powerful call scripting and follow-up management. FreeCRM.com with click-to-call makes it simple to quickly make outbound calls to any phone number and track it with a simple click. By combining WebRTC with CRM, users can quickly build targeted call lists and automate the calling process as well as taking advantage of the huge cost savings using this new VoIP-style technology. FreeCRM.com uses WebRTC to make it easier to call any U.S. phone number straight from your browser using your speakers and microphone.
IceLink is a collection of libraries that enable developers to create reliable UDP media streams between peers, regardless of the peer's network configuration and environment. IceLink traverses every possible firewall/NAT combination to guarantee connection establishment. It uses IETF/IANA standards to provide the broadest compatibility with third-party components. IceLink includes a WebRTC extension for select platforms that implements the WebRTC standards for communication, including stream formatting, RTP/RTCP packet processing, DTLS key exchange, audio/video capturing/rendering, audio/video encoding/decoding, data channels, a full MediaStream API, and more. IceLink works in conjunction with any third-party signaling library (like WebSync, XMPP, or SIP) to perform an initial offer/answer exchange when setting up peer connections.
In April 2013 GENBAND introduced SPiDR, a WebRTC gateway. SPiDR sits at the edge of the network and provides open, web-centric APIs that allow application developers to leverage the rich communications services of the telecommunications network – including voice, video, presence, shared address book, call history, instant messaging, and collaboration. Leveraging SpiDR’s advanced technology service providers can now quickly extend the reach and breadth of their offerings to include rich multimedia directly from any web browser – including voice, video, IM and presence. The OS-agnostic SpiDR (Windows, Android, iOS, BlackBerry) makes it possible to offer rich unified voice, video and shared-data services to all subscribers, via desktop, laptop, smartphones, tablets or mobile phone browsers. Additionally, GENBAND’s SPiDR investment enables the integration of WebRTC into all areas of the service provider network – from the core, to the edge, to the experience – and across any network architecture. Earlier this year GENBAND introduced SMART OFFICE 2.0, a WebRTC-enabled unified communications platform. Designed in HTML5, the SMART OFFICE 2.0 soft-client is the first WebRTC compliant multimedia user experience that delivers the premium suite of voice, video, conferencing, chat, presence and collaboration features through a browser.
The Ingate WebRTC & SIP PBX Companion is an OEM product for PBX and call center vendors, bringing all the benefits and features of WebRTC to the enterprise SIP PBX and UC solution. It includes a WebRTC/SIP gateway, a SIP E-SBC, a firewall for security and Ingate’s Q-TURN technology for quality assured videoconferencing. Everyone’s web browser becomes the soft client, both locally and remotely for the PBX, UC and call center infrastructure, both for enterprises and service providers. Click-to-dial buttons into the PBX are easily added to the enterprise website, and http links can be passed as invitations to call individuals or to join meetings. Ingate’s Q-TURN technology is included in Ingate’s line of session border controllers and also licensed to firewall vendors and carrier network equipment vendors. Q-TURN gives WebRTC, or any real-time traffic using the ICE/STUN/TURN standard for NAT/firewall traversal, end-to-end connectivity and priority over data traffic. That allows the telepresence capabilities in PCs, laptops, tablets and smartphones to be used with WebRTC person-to-person communication. Existing Ingate SIParator E-SBCs can be upgraded with Q-TURN and Q-TURN technology enabling carriers and network providers to offer WebRTC-ready broadband access.
LiveOps (News - Alert) Inc., which provides cloud contact center and customer service solutions, has led the charge in transforming the contact center by expanding its traditional voice support with WebRTC - the emerging standard for browser-to-browser communications. LiveOps Engage, a single integrated multichannel agent desktop, provides native integration with Twilio (News - Alert) Client to deliver a true virtual contact center. LiveOps Engage users are now able to handle voice calls directly from their browser – whether from a PC or tablet – with zero requirements for a landline, mobile phone, or any downloads. The need for expensive servers, software, landline and phones required by traditional call center technologies can now be eliminated, which dramatically reduces costs while enhancing agent experience and productivity. Furthermore, companies can continue to create sophisticated and personalized call routing strategies to increase call resolution with advanced call management and dynamic business rules. Leveraging Twilio Client, LiveOps Engage provides an option for any company looking for an opportunity to forgo its telephony infrastructure and have everything – contact center plus telephony – in the cloud.
The advantages of extending clientless multimedia service options to mass-market consumers are obvious. Complementing native client-based SIP endpoints, network operators must ensure that both their legacy and new telephony infrastructures can support browser-based real-time communications. Acting as a SIP server, terminating WebSockets from a browser and converting them to UDP or TCP, Metaswitch has implemented SIP-over-WebSockets gateway functionality, formalized within the IETF,
The company's WebRTC Telemedicine videoconferencing tool allows organizations to set up users, groups and subgroups with crystal clear, real time, face-to-face web-based connections significantly facilitating doctor-patient and hospital communications. Although designed specifically for the health care industry, other types of companies and organizations can also use the system. RTC Conference Switch and RTC Receiver can
be embedded on a website for one-to-one and multiparty calls. Features include push-to-talk, echo canceling, data channels for text and speech recognition, and medical device integration. No plug-Ins are required and the solution is compatible with existing browser technology from Firefox, Google Chrome and Opera.
NGVX or Nex Gen Video Exchange is a video calling server from Nex Gen Bits, bridging together the worlds of SIP and WebRTC. This software-based conferencing and streaming server is intended for anyone looking to deploy a solution for video collaboration, conferencing, webcasting, recording, and live video streaming. NGVX allows you to securely host a meeting, record the video and audio content while simultaneously webcasting to any number of native devices such as iPhones, Android, BlackBerry, and desktop clients. It supports WebRTC, allowing anyone with a web browser to seamlessly connect through the conferencing server to any other endpoint for real-time collaboration. Leveraging the media engine of Nex Gen Media Server, it is capable of operating as a multipoint conference server, providing video interoperability and live video streaming services to a variety of devices. It can be deployed as a stand-alone server or in conjunction with the IMS calling architecture to provide real-time media services in the role of a media resource function processor. The software is open source and customizable for integration into your own proprietary streaming solution and architecture.
Oracle Communications announced the Oracle Communications WebRTC Session Controller in September 2013. The product enables communications service providers and enterprises to offer WebRTC services – from virtually any device, across virtually any network– with carrier-grade reliability and security. The Oracle (News - Alert) Communications WebRTC Session Controller is designed to support the emerging demand for WebRTC communications and supports the development of WebRTC-based applications and services to enable simple peer-to-peer web communications without a plug-in download. Specifically, the product enables CSPs and enterprises to integrate carrier-grade signaling, policy and charging with any WebRTC application to help reliably scale WebRTC offerings to millions of subscribers, and also provides robust security and authorization capabilities not currently built-in to the open-source WebRTC API standard. The product also facilitates interworking between WebRTC clients and SIP-based, multivendor unified communications systems, enabling users to more easily access their enterprise communications environment and seamlessly switch WebRTC sessions between devices.
Pexip Infinity is a software-based, virtualized conferencing platform providing personal meeting rooms to any number of users on video, voice and mobile. It runs on standard X86 servers using hypervisors from VMware and Microsoft. It is a distributed architecture, and a voice/video/data conference can exist on one or more servers in one or more locations at any time. Infinity provides broad interoperability. A single conference may have SIP, H.323, Microsoft Lync and WebRTC endpoints connected, providing HD video, wideband audio and data sharing to and from all the different participants. All protocols are supported natively in the Pexip Infinity product – no gateways are required for interoperability with H.323, SIP, WebRTC, H.263, H.264, H.264SVC, VP8, BFCP, H.239, MPEG4 AAC, Opus, etc. WebRTC participants connect to a Pexip conference without any browser plugins. Voice, video and desktop sharing from WebRTC is transcoded on the fly to any protocol required by H.323, SIP or Lync participants. A typical deployment leverages the distributed architecture by deploying Pexip Infinity conferencing nodes in all major regions. Videoconferencing endpoints connect to the closest Pexip conferencing node. Between Pexip conferencing nodes bandwidth is preserved, only forwarding required video stream in full resolution.
PubNub offers an open source template to allow developers to quickly and easily add Skype-like video chat into their apps. The free template provides a fully functional video chat platform using WebRTC, PubNub and Google Authentication for a global, reliable collaboration solution. All the core elements needed to build and deploy a fully featured WebRTC video chat product are available as a documented, open source template. This new template leverages the browser’s built-in WebRTC API for peer-to-peer audio and video, and adds the additional components needed for a fully deployable video chat application. This includes collaborative features such as presence detection to see which friends are online, call signaling and initialization, text chat, and friend lists. This can be easily extended to use Facebook (News - Alert), LDAP or other homegrown directory services. Together, these components allow any developer to add a globally scaled video chat solution to apps running on WebRTC-compatible platforms. All the source code, a website with a working app using the template, and a tutorial are available online.
With more than 6 million ports deployed, mobile network operators, service providers and converged communication developers depend on Radisys media processing solutions to develop and deploy a growing variety of WebRTC, VoLTE, OTT and other real-time IP communication services. Radisys has added WebRTC features and codecs to its standards-based Media Resource Function for IMS architectures, enabling its media processing products to support scalable HD video transcoding and transrating for WebRTC to IMS gateway vendors. Its broad range of multimedia processing capabilities allows WebRTC service developers to rapidly integrate and deploy revenue-generating WebRTC services, such as IP contact center communications, or multimedia conferencing between IMS and WebRTC endpoints. MPX Operating Software provides the common software foundation for all of Radisys’ media processing products, allowing WebRTC customers to develop with the Radisys Software MRF for virtualized cloud deployments, and later seamlessly scale to the purpose-built Radisys MPX-12000 platform. Radisys’ MRF product portfolio delivers the real-time IP media processing required for WebRTC service revenue generation.
An independent mobile VoIP firm whose products have more than 23 million users globally, Rebtel has two products that utilize the WebRTC framework. The first is a calling app, Rebtel, available for iOS and Android. It lets users make free calls to other Rebtel users anywhere in the world, and calls to non-Rebtel users are super cheap (just 1.5 cents per minute to call the U.S. from abroad). Callers can make international calls over Wi-Fi/mobile data, or they can use their local minutes. During a call, users can switch seamlessly between Wi-Fi /mobile data and local minutes using Rebtel’s Keep Talking feature. Rebtel also features seamless contact list integration, so users can automatically see which of their contacts are Rebtel users who can be dialed for free. Rebtel also uses the WebRTC framework in the Rebtel SDK, which lets developers add free app-to-app voice and IM communication to their apps with just a few lines of code. It’s an ideal solution for developers making apps in categories like social, enterprise, dating and gaming. Developers can even use it distribute their own mobile VoIP app without a major back-end commitment.
Developed by Telenor Digital, the new service appear.in allows users to easily set up video conversations in the browser without any logins or downloads. As opposed to existing videoconference solutions such as Skype and Lync, appear.in uses your Internet browser, thus making it easy to set up a call with both friends or business associates without requiring any downloads, installations or even a login. Up to six simultaneous guests simply click on their invite link to enter the unique room. Users can even customize the background images to make their chat room more personal or representative, and can claim ownership of their personalized room name for repeat usage. Built on the new standard for real-time communication in the browser, WebRTC, the goal of appear.in is to make talking over video as natural as using a phone.
One of the most powerful and appealing features of UberConference is its use of WebRTC. UberConference was one of the first to adopt the WebRTC standard, working directly with the Google Chrome team to develop an elegant integration. UberConference was also the first audio conferencing service to offer HD audio over WebRTC. UberConference allows users of its award-winning visual conference calling solution to “dial in” directly from the Chrome browser, making calls more accessible for all. Now anyone can use the service without the need for a phone or a U.S. telephone number – just as long as they have Internet access. This works well for international callers, eliminating long-distance charges and making UberConference available from anywhere around the world. All you have to do when you’re connecting to your UberConference is choose the option Use Your Computer. You will then join the call in HD audio via your computer. It’s that simple.
XirSys is a WebRTC hosting platform, providing a cloud server network, API, developer tools, and personal hands-on support to make using WebRTC easy. As you may have found out firsthand, it's extremely difficult to develop your own backend infrastructure, especially when accounting for all the super secure NAT scenarios and complicated firewalls. Our service takes care of all that for you, so you can focus on the important stuff: your project’s functionality and creating a great application. And our simple pricing model makes it easy to see what you are paying for; we charge only for the bandwidth you use from our servers and nothing else. Although WebRTC is a more recent initiative, we are far from novices in the practice of real-time communication. We've been practicing RTC for more than 11 years under the name of Influxis, so we have a solid understanding of the requirements, expectations, and general know-how that are needed for successful WebRTC implementations.
Edited by Stefania Viscusi