SIPWISE AND DEUTSCHE TELEKOM / TLABS

ANNOUNCE COOPERATION FOR THE

WebRTC SOLUTION

RTC:engine

Sipwise and Deutsche Telekom AG / TelekomInnovation Laboratories signed a cooperationagreement for joint development and distributionof the WebRTC based communication solutionRTC:engine. Sipwise includes the solution in theirproduct portfolio and will provide vertical productsand solutions (like call center, e-learning, …) on top.

Vienna, July 1st 2015

Executive OverviewDuring the last year, Sipwise and DeutscheTelekom have been working together inthe emerging field of web based real-timecommunication, enabling end users toconsume voice, video and chat servicesvia cloud based solutions. It is ourcommon goal to completely uncouple auser's identity from their devices, grantingaccess to services from wherever a webbrowser is available.

Sipwise as a vendor of Telco solutions hasfollowed a revolutionizing approach in theindustry since their market entry in 2008by bringing the open and modular productdevelopment approach of the Internetworld to the Telco space. Using andcreating open source software to buildcutting edge solutions for real-timecommunication has been key to rapidlylaunching stable and flexible products forTelco carriers who provide core- and OTTservices to their end customers.

Since July 2015, Sipwise offers a new real-time communication framework calledRTC:engine, jointly developed by Sipwiseand T-Labs, the Innovation Laboratories ofDeutsche Telekom. RTC:engine providesboth back-end technology for highperformance signaling and media routing,and web services and CDKs, allowingeasy front-end application development inHTML5 for rapidly building cutting edgecloud communication applications.

A complete software-only approachencapsulating different signaling protocolsand media handling in simple APIs with afocus on on-premise scenarios allowsoperators of RTC:engine to launch nextgeneration communication services withintheir own private cloud infrastructure.

RTC:engine seamlessly blends into theSipwise product portfolio and enables anew approach to communicationsolutions.

The key USP of RTC:engine is itscapability to extend traditionalcommunication infrastructure like SIP andXMPP services and enrich it with purelyweb-based scenarios in a very short time-to-market, without the need to dive intolow level APIs or protocols.

Enabling new Use CasesRTC:engine as a purely web based overlayon top of existing communicationinfrastructure enables new use cases forboth core and OTT services.

Use Case: Cloud Telco

A purely web-based communicationsolution offering voice, video and instantmessaging from within an HTML5 webapplication fully decouples the servicefrom the device. No software or app otherthan a modern web browser needs to beinstalled on a device, nor is any specifichardware phone necessary to use theservice. Yet, all typical features like callforwarding, hunting etc. are available as

usual and existing end customer devicesand apps can be used in parallel andseamlessly along with the web application.

Integration of the web app into existingcustomer portals with single-sign-onmechanisms allow for stronger customerloyalty. At the same time, third partyofferings like cloud storage, musicstreaming and more can be used alongwith the real-time communication service,thus forming a fully featured entertainmentand communication portal for endcustomers.

Use Case: Smart Video Sharing in Call Centers

RTC:engine offers the flexibility toestablish two-way communication pathsvia arbitrary signaling channels, andtherefore a side channel communication inaddition to standard voice calls e.g. viamobile phones can be established at anytime.

A compelling use case would be a callcenter agent creating an on-demand videocall by clicking a button in a CRM console,which sends a link to the calling customervia SMS . Once the customer opens thelink, a video element is shown in theagent's CRM, allowing him to instruct thecustomer to remotely show the cabling ofa router or any other device the customerneeds support for, and providing the abilityto remotely take pictures which are

automatically attached to the supportcase.

Use Case: Multi Party Video Conferencing and Collaboration

The classical use case for web basedcommunication is cost effective multi-party video conferencing solutions withoutthe need for tailor-made third partyconferencing. Since video streams aresent separately to the web browserinstead of having fixed tiles in one singlestream, the web application can takeadvantage of designing the user interfaceaccording to their use cases, likeannotating video streams and more.

Parallel instant messaging sessions inconferencing mode or in private chatsessions between participants, includingpeer to peer or public file sharing as wellas screen sharing enrich the overall videoconferencing experience.

RTC:engine and WebRTCRTC:engine is based on WebRTC, an opensource media engine for modern webbrowsers. WebRTC provides JavascriptAPIs for web applications to access theuser's microphones and cameras on theone hand and to establish peer-to-peerconnections between two browsers forexchanging real-time communicationstreams (voice and video calls) and datastreams on the other hand. WebRTCfocuses on security by enforcing strictencryption of communication channels,and on media quality by applying qualityfeedback and adaptive bandwidth controlof codecs used.

WebRTC in itself does not provide asignaling protocol to exchange mediainformation between endpoints. This partis handled transparently for the applicationdeveloper by RTC:engine by exposing aJavascript API in the RTC:engine CDK toperform the relevant communication tasksvia HTML5 websockets.

The RTC:engine back-end systemconverts the internal signaling protocol toSIP, XMPP and other protocols whichmight be introduced in the future. Thisallows an RTC:engine web application tointegrate seamlessly into existing SIP andXMPP networks. The encrypted mediastreams on the WebRTC layer aretransparently converted to plain mediastreams by RTC:engine for communicatingwith legacy devices not supporting mediaencryption.

In case of internal communicationscenarios or calls from or to SIP devicessupporting the Opus codec, the mediaquality is full HD, providing crystal clearvoice quality.

RTC:engine in a Mobile EnvironmentDue to the underlying WebRTC technologyleveraging Opus as audio codec, therequired bandwidth for calls usingRTC:engine can be reduced significantlydown to narrow band, resulting inbandwith consumption lower than AMR-NB and matching the bandwith of G.729.Transcoding between narrow-band on theRTC:engine side and G.711 for exampleon the peering side towards fixed andmobile networks is handled directly withinthe system.

This transcoding mechanism allows forclear voice calls even in scenarios withrestricted bandwith like 3G and 2G,whereas full HD quality can be deliveredover LTE.

RTC:engine and Native AppsImplementing a soft-phone with voice,video, chat and presence functionality ingeneral, and a mobile app in particular,require huge development effort due totheir complexity and platformfragmentation (e.g. Android vs. iOS). Thisresults in huge costs and slow time-to-market. Re-branding an existing solutionrequires a Telco or Enterprise to stick tothe general outline and feature set of anApp and only allows simple branding.

Alternatively, using a web based approachon top of RTC:engine gives full flexibilityon the look and feel and the feature setincluded within the app, while at the sametime it can be built to work on any platform(desktop browsers and mobile platforms).Releasing design changes and additionalfunctionalities is as easy as rolling out anew web application.

RTC:engine based applications for mobileplatforms are usually still wrapped within avery small native app in order to receiveinbound call and message notificationswhen the customer is not logged in on theparticular web page providing the cloudcommunication service. Sipwise providesmobile push mechanisms in the back-endto notify mobile devices, and a wrapperapp which starts the mobile webapplication automatically or on demand toanswer calls or replying to messages. OniOS, third party APIs can be linked into thewrapper app to expose the WebRTC APIsrequired by RTC:engine, which arecurrently not available natively on iOS.

Third Party Components - Mix and MatchRTC:engine is designed in a modular wayto allow the engagement of differentnetworks (native WebRTC, SIP and XMPPare provided out of the box) as well asdifferent third party application servers.

One application server natively integratedinto RTC:engine is Dialogic XMS, providingmulti party video conferencing services,transcoding between various codec pairsfor improved interoperability, andannouncements.

Any combination of networks and externalapplication servers is possible. New plug-ins are provided on request.

Contact InformationFor more information on RTC:engine or theSipwise product portfolio, please contactour Sales team at <sales@sipwise.com>.