The FRAFOS ABC SBC WebRTC gateway

The WebRTC Gateway

WebRTC-‐GW Browsers and Digital Telephony meet in Business Applica>ons

The

Frafos WebRTC GW in Nutshell

The telephony gateway server that connects browser telephony1 with VoIP2 and integrates

it with web applica>ons.

[1] webRTC standard support includes SIP over websockets [rfc7118], OPUS codec [rfc6716], DTLS [rfc6347] [2] VoIP standard support includes SIP (RFC3261), DTMF [rfc4733], ENUM [RFC6116]

The Frafos Key Product Features… One interoperable world: web browser users can connect to SIP users (and PSTN through SIP-‐2-‐PSTN gateways) thanks to the gateway func7onality. The poten7al for networking effect is unmatchable.

Applica7on integra7on: telephony is not just an audio call. It is web applica>on integra>on… with the Frafos product you place a fragment of HTML code in an exis7ng app and integra7on is finished. No APIs, no SDKs!

The whole solu7on. The Frafos product comes packaged with a JavaScript applica>on that runs on any WebRTC browser and can be customized for specific use-‐cases. Interoperability guaranteed.

… Key Product Features (cont.)

Security is unprecedented. Telephony borrows security from web-‐browsers and delivers for the first >me ins its history 256-‐bit and beNer encryp7on for every user.

Rapid Deployment in Clouds. Launch a service in less than 10 minutes – that’s how long it takes to start a whole cluster including load-‐balancers, firewalls and monitoring facility using Amazon Elas>c Cloud!

Example WebRTC Use Cases

•  Click-‐to-‐dial: Telephony embedded in web-‐pages is an incomparably more convenient and affordable alterna>ve to 800 free-‐calls.

•  Call-‐centers: web telephony makes integra>on with CRM easier, a_endants can seamlessly work from the comfort of their home PCs.

•  Secured office communica7on: web-‐grade security makes it easy for organiza>ons to run their own telephony service hard-‐to-‐intercept by any third-‐party along the communica>on path.

Key Features in Detail

One Interoperability World: Global Reach

INTERWORKING CONNECTS!

•  The Frafos WebRTC-‐SIP gateway interconnects web-‐browsers with any SIP-‐compliant devices: telephones, PBXs, gateways.

•  The “SIP side” can connect to PSTN through SIP/PSTN gateways and has been cer>fied against BT wholesale service.

DID YOU KNOW? The potential for networking effect is more than promising: There are about 7 billion Internet users and 7 billion mobile phone users worldwide.

DID YOU KNOW? The WebRTC service can be global and run multiple gateways in different geographic regions. Geographic diversity helps to eliminate inter-continental RTT latency of about 200ms down to 50ms for best audio experience.

One Interoperability World: Any Device, Any WebRTC Browser

Supported Client Equipment Web-‐telephony runs on any browser with WebRTC support. The JavaScript applica>on delivered with Frafos Web RTC-‐GW has been tested against Chrome, Firefox and Opera on smartphones, tablets and PCs using both audio and video. The app’s small footprint (190 kB) makes it a reliable choice for mobile devices.

Supported standards The “VoIP side” complies to the SIP RFC 3261 standard family. The “WebRTC side” supports SIPoWebsockets (RFC 7118), DTLS/SRTP and OPUS/G.711.

DID YOU KNOW? According to Gartner, worldwide shipments ratio PCs to mobile devices was 317,468 to 2,334,400 in 2013.

Applica>on Integra>on: Embedding Telephony in Web Apps

WebRTC Browser Client

Existing Business Applications

Frafos WebRTC Gateway

Customized HTML-linked WebRTC application

With a running Frafos WebRTC-GW, all one needs to embed telephony in existing web applications is to include a HTML code fragment. That’s really it: NO SDKs, NO APIs.

<button class="callme-btn callme-btn-lg" data-label-default="Call Frafos Voicemail" data-label-dialling="Dialling..." data-label-connected="Connected" data-call-to=”sip:[email protected]” data-conf-ws_servers= "ws:// tryit.areteasea.com:8080”> Call Frafos Voicemail</button> <script>!function(d,s,id){var js,fjs=d.getElementsByTagName(s)[0],p=/^http:/.test(d.location)?"http":"https";if(!d.getElementById(id)){js=d.createElement(s);js.id=id;js.src=p+"://go.areteasea.com/assets/js/callme-widget.js";fjs.parentNode.insertBefore(js,fjs);}}(document, "script", "click2dial-widget"); </script>

Whole Solu>on

•  Frafos has adopted the jssip.net applica>on and delivers a complete cer>fied client-‐server solu>on

•  The applica>on features audio (G.711, OPUS) and video (VP8, H.264 on Firefox), is wri_en 100% Javascript and is leightweight (~190 kB)

•  Customiza>on available on request

Integra>on Example: spinoco’s Call Center App Built on top of the RTCGW and JSSIP

WebRTC Security

Supported ciphers

•  TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA •  TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA •  TLS_DHE_RSA_WITH_AES_256_CBC_SHA •  TLS_RSA_WITH_AES_256_CBC_SHA •  TLS_ECDH_anon_WITH_AES_256_CBC_SHA •  TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA •  TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA •  TLS_DHE_RSA_WITH_AES_128_CBC_SHA •  TLS_RSA_WITH_AES_128_CBC_SHA •  TLS_ECDH_anon_WITH_AES_128_CBC_SHA •  TLS_RSA_WITH_3DES_EDE_CBC_SHA

DID YOU KNOW? Brute-force attack on AES/256 would require 1.1x1077 combinations. A 10.51 Pentaflops computer would require 3.31 x10 56 years to crack the cipher – more than the age of universe (13.75 billion years).

Both analog and digital telephony were inherently insecure, mobile telephony secured at least the wireless hop, yet rather weekly. SIP’s security protocols, PGP, S/MIME and Identity failed to be adopted. With WebRTC we have cryptographic security on the link from browser to the Frafos SIP RTC Gateway using the DTLS protocol. [rfc6347]

Cloud in Ten Minutes

DID YOU KNOW? Frafos WebRTC gateways can be orchestrated in a fault-tolerant, auto-scaling Amazon Elastic Cluster using Cloud Formation within ten minutes. The cloud includes gateway instances in a dedicated private network, load-balancers, firewall rules and an autoscaling strategy. Visit http://go.areteasea.com to launch your own EC2-powered cluster.

FRAFOS WebRTC GW is available as CentOS7 packages, virtual image or Amazon EC2 cloud

Informa>on Resources

WebRTC-‐GW: Brief Specifica>on

•  Protocol compliance –  VoIP: SIP (RFC3261), DTMF, –  Security: TLS, SRTP, DTLS, SDES –  WebRTC: SIP over Websockets [rfc7118] –  Media: ICE, STUN, OPUS, G.711, VP8

•  Security features: –  Policies: blacklis>ng, whitelis>ng, codec

selec>on –  Traffic shaping: parallel calls, call rate,

bandwidth –  Topology Hiding –  Registra>on off-‐load

•  Enhanced Signaling and Media Processing Features

–  Embedded Media: announcements, music-‐on-‐hold

–  SIP media>on, Transcoding •  Distribu>on available as:

–  Centos7 packages –  Virtual machines (VMWare, VirtualBox) –  Amazon/EC2 cloud

•  Cloud System Architecture –  Amazon/EC2 based –  Ac>ve-‐ac>ve auto-‐scaling cluster

•  Programmable APIs –  Via RESTful interface

•  Management and Monitoring –  Administra>on rule-‐based GUI –  Event processing –  SNMP –  PCAP collec>on –  Audio recording –  Mul>-‐domain support by virtualiza>on

•  Comes with JSSIP.net client app –  Leightweight footprint ~190 kB –  Customized versions available –  Certified against the WebRTC-GW –  Support for cloud scaling

Protocol Reference Diagram

SIP Equipment RTCWeb browsers Frafos RTCWeb-GW

Multiplexed UDP media port: STUNoUDP, DTLS, OPUS/G.711-o-SRTP-o-UDP

Signaling port: SIPoWebsockets-o-TCP

UDP media port: G.711/GSM/iLBC/…-o-RTP-o-UDP

Signaling port: SIPoTCP/UDP

Optional: Management Servers

Optional: Business Logic Servers

HTTP port: download Web App

Amazon Reference Architecture

SIP Equipment RTCWeb browsers Amazon Elastic Load-Balancers

Multiplexed UDP media port: STUNoUDP, DTLS, OPUS/G.711-o-SRTP-o-UDP

Signaling port: SIPoWebsockets-o-SSL

UDP media port: G.711/GSM/iLBC/…-o-RTP-o-UDP

Signaling port: SIPoTCP/UDP

HTTPS port: download Web App

multiple amazon-managed load-balancers behind a DNS name; Load-distribution round-robin TCP/SSL-based)

HTTP

SIPoWS

Gateway Instances

Number determined by scaling policy. All instances behind a NAT.

Autoscaling group

Components not shown: CloudWatch used for monitoring, Route 53 for DNS-based geographically dispersed clusters. Cloud Formation for launching a whole cluster. Multiple clusters can be launched in separate geographic regions for best media experience.

Roadmap Preview Events Console (Q1/2015)

•  Events screenshot

HOW IT WORKS… •  The events console collects all events related to a subscriber (call started/ended,

registra7on succeed/failed, etc) •  Administrators can narrow down the events by various filter to find out a behavior

they need to troubleshoot

Important Frafos Links

•  The demo website available at – h_p://go.areteasea.com/

•  Product web-‐page: – h_p://www.frafos.com/webrtc/

•  Online product documenta>on: – h_p://www.frafos.com/upload/html/34_b_webrtc.html

WebRTC in 2014 Industry News

•  h_ps://blog.mozilla.org/blog/2014/10/16/mozilla-‐and-‐telefonica-‐partner-‐to-‐simplify-‐voice-‐and-‐video-‐calls-‐on-‐the-‐web/

•  h_p://blogs.cisco.com/collabora>on/ciscos-‐openh264-‐now-‐part-‐of-‐firefox

•  h_p://www.pcworld.com/ar>cle/2858212/google-‐and-‐avaya-‐to-‐bring-‐chromebooks-‐and-‐webrtc-‐to-‐call-‐centers.html

Thank You Frafos GmbH, Ahoy office, Windscheidstraße 18, 10627 Berlin, Germany

h_p:// www.frafos.com mailto:[email protected]

The FRAFOS ABC SBC WebRTC gateway

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