Webinar WebRTC HTML5 (english)

Introduction to WebRTCNovember 20th, 2012

In a few words

WebRTC (Real Time Communications) will be the next big thing in terms of unified communications during the next years. Web browser will be able to manage voice and video in a native way.

During 60 minutes, Elías Pérez, Antón Román and Iago Soto will talk about their vision about this technology and how is the best way to implement WebRTC in the legacy and VOIP networks of corporations and telcos.

Iago Soto MataCMO

@iagosotoiago.soto@quobis.com

Antón RománCTO

@antonromananton.roman@quobis.com

Elías Pérez CarreraCEO

@epcarreraelias.perez@quobis.com

Agenda

- Introduction to HTML5 and WebRTC- Network architecture- Tech challenges- Application cases- Demos- Identity Management- Questions and answers

HTML5 and WebRTC. Short intro

... is an opensource project that makes possible to manage multimedia communications in the web browsers, using simples API's in Javascript, in a native way.

It is promoted by the team of Google Chrome

... is the fifth version of the HTML language, that offers new capabilities to create web pages, specially in terms of dynamic elements.

Defines WebSockets as the new transport protocol between browser and servers

It is promoted by W3C

HTML5 and WebRTC. Standards

Two main players:

● IETF RTCWeb WG ( Internet world)● W3C WebRTC WG (web world)

Drafts:

WebRTC 1.0 Real-time Communication Between BrowsersWebRTC MediaCaptureThe WebSocket APIdraft-ietf-rtcweb-data-channeldraft-ietf-rtcweb-jsepdraft-ietf-rtcweb-rtp-usage

... etc ...

Just a few days ago !!!

HTML5 and WebRTC. Increasing interest

HTML5 and WebRTC. Advantages

Opensystems, with no propietary implementations

¡No plugins!

Multi-platform... and multi-device!

HTML5 and WebRTC. Application cases

Software is going to migrateto WEbRTC like collab apps

HTML5 and WebRTC. Application cases

And Click2call will be a trend in CRM or marketing websites

Architectures. Browser to browser

Any browser will have capabilities to call any other with RTC.

Architectures. Multiconference

Architectures. SIP interconnection

SIP to WebRTC gateway will be implemented for interconnection.

Architectures. PABX interconnection

Tech challeges. Codecs

G711a/u (RFC 3551): supported by all the devices. Needs to use a lot of bandwidth.

DTMF tones (RFC 4733, updates RFC 2833): needed for interactions with several systems (for instance IVRs).

Opus (RFC 6716): bitrate variable, low latence and high quality for human voice and music. Specially designed for real time communications.

In order to interact with VoIP systems, in several scenarios, it will be needed transcoding or interworking of DTMFs (RFC 4733-> INFO, RFC 4733-> in-band, etc).

And this only for audio... ... battle in vídeo VP8 vs H264

Tech challeges. SRTP vs DTLS-SRTP

VoIP devices implement normally RTP encryption using SRTP. They share the key in the SDP protocol.

DTLS-SRTP implements a new method to manage the SRTP key. This is done using DTLS, a version of TLS based on datagrams.

Tech challeges. NAT. STUN and ICE

Browsers are allways behind NAT and Firewalls, so it's difficult to exchange packets in real time.

The user want a seamless communication, with no problems (similar to Skype).

It is needed to found a "way" for RTP/RTCP traffic, independent to the architecture..

ICE, makes possible (using STUN and TURN, protocols) to discover y to choose the addresses that are going to be used to exchange packets.

ICE was a difficult protocol to implemnt and had to be defined during some years.

Tech challeges. Media negotiation

Uses SDP for negotiating media.Classic negotiation offer-answer

There is a draft IETF: SDP for the WebRTC / draft-nandakumar-rtcweb-sdp-00

Example SDP: | v=0

| o=bob 16833 0 IN IP4 0.0.0.0 | s= | t=0 0 | a=ice-ufrag:c300d85b | a=ice-pwd:de4e99bd291c325921d5d47efbabd9 | a2 | a=fingerprint:sha-1 | 99:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:7 | 0:9d:1f:66:79:a8:07 | | m=audio 49203 RTP/AVP 109

Tech challeges. Signalling. Options

WebRTC does not define how to manage signalling

There are some possibilities● XMPP / Jingle● SIP● Protocol ad-hoc that manages SDP's

Quobis chooses SIP:● "Standard": NGN networks are supported by SIP● "Interoperable": it's possible to connect to everything!!!● "Powerful": allows to build apps over it

Exisiting SIP implementations:● sipML5: first implementation available● JsSIP: authors of the draft-ibc-sipcore-sip-websocket

Introduction to QoffeeSIP

Complete implementation of SIP protocol using Javascript

It runs directly in the browser

Focused on developers,written using CoffeeScript

Easy to extend.

Light application (5 KB)

Will be opensource in the nexts weeks.

Introduction to QoffeeSIP

+ SIP =

Interconnection of browsers with SIP legacy devices.

Tech challeges. Adoption. Browsers

Really involved in WebRTC

First stable versions (for desktop)

Implements PeerConnection, GetUserMedia, etc..

Involved, working more slowly than Chrome

Implements part of the API

Iniciative Firefox OS for mobiles

Involved in WebRTC

First stable versions (for desktop)

Implements PeerConnection, GetUserMedia, etc..

Tech challeges. Adoption. Browsers

Unknown roadmap, patents involved

Important for smartphones and tablets

Involved in WebRTC

Suggestion and proposal that do not fit with the standard

There is a plugin developed by Google (Chrome Frame)

Developed by Ericsson labs

First browser with WebRTC in mobiles (without Websockets)

Beta version for developers.

Tech challeges. Security.

Media access: must be allowed by the user

Models of allowance:

● Access to cam and mic in a unique session● continuous access to cam and mic● Accesses based on users:

○ Allow calls just to an user○ Allow calls to known users

LED to show that camera o mic are been used

Automatic off in case of not attendance (change of window)

Defined in draft-ietf-rtcweb-security

Tech challeges. Monitoring

It is possible to monitor all the traffic, similar to standard SIP.

Similar to SIP over TLS, if WSS is used (secure Websockets) monitorization should be done at the edges (most usually in the server).

We have modified the monitoring tools in order to see websocket traffic.

Case studies. Applications

Telcos Corporaciones

SP / Social media Fabricantes

Case studies. Applications

Telcos

● Webphone for existing customers● Integration in NGN and IMS● Inbound channel for corporate

customers● Alternative to calls from mobiles

Case studies. Applications

● Inbound Click to call. New channel.● Can be deploy by ourtsourcers● Calls in internal directories● Multiconferences.● Integration with legacy systems

Corporations

Case studies. Applications

● Social networks -> next MVNOs● New services for webs managing IDs● ISPs will offer UC modules/widgets● Entertainment: chats, etc...

SP / Social media

Case studies. Applications

● Online gaming● Set-top-boxes and web TV● Gateways WebRTC to IPBX● Mobile device manufacturers

Manufacturers

Demo. TalkSetup

http://TalkSetup.quobis.comJoin a demo of WebRTC at

Demo. TalkSetup backend

Easy to manage WebRTC calls and users.

Tech challenges. Identity management

Makes possible to be sure of the identity using a thirdparty

Adds a second factor of authentications because we validate the device (smartphone or PC) and the credentials are introduced ciphered in a SIP signalling packet.

Tech challenges. Identity management

Tech challenges. Identity management

Tech challenges. Identity management

Agents can be sure of the identity of the person who is calling.

Demo. IdentityCall

Demo using national electronic ID card.

What we have learned

● Voice traffic is going to be thru web● Browsers are the new endpoints● A website of a company can be the call

center● Telephone number is not important● Security, identity a privacy is very

important● New business opportunities

Iago Soto MataCMO

@iagosotoiago.soto@quobis.com

Antón RománCTO

@antonromananton.roman@quobis.com

Elías Pérez CarreraCEO

@epcarreraelias.perez@quobis.com

Any questions?