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DevCon5 (July 2014): Acision SDK 
Peter Dunkley, Technical Director, Acision 
Email: peter.dunkley@acision.com 
Twitter: @pdunkley 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 1
Acision at-a-glance 
Heritage & history 
20 year history 
270 customers globally 
2 trillion messages/ year 
Global presence & reach 
35+ offices across 6 
continents 
850+ employees 
100+ support team 
Innovation 
1st to launch commercial 
SMSC 
Cloud & virtualized 
services 
Multi-Channel messaging 
+ WebRTC 
Operational Excellence 
Data and insights 
Carrier grade security 
© Acision 2014 24/7 Support
Evolution on the Web 
Sir Tim Berners-Lee 
creates HTML. Web 
-pages are static 
W3C produces the 
DOM1 specification 
1990 1996 1998 2004 2011 
Microsoft and Netscape 
introduce different 
mechanisms for DHTML 
Google uses Ajax 
in Gmail (W3C 
releases 1st draft in 
2006) – the dawn 
of web-apps 
WebSocket and 
WebRTC 
implementations 
become available 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 3
Revolution in Telecoms 
The revolution 
Before today the operators (big and small) 
had full control over real-time 
communications because it was hard to do 
and substantial infrastructure investment 
Claude Chappe was required. 
invented the optical 
telegraph 
Alexander 
Graham 
Bell patents 
the telephone 
From the 1960s 
onwards digital 
exchanges start to 
appear 
From the 1990s onwards 
voice started to be carried 
on technologies developed 
for data networks such as 
ATM and IP 
1792 1837 1876 1919 1960s > 1990s > 2011 
WebSocket and 
WebRTC 
implementations 
become 
available 
Rotary dial 
enters 
service 
First commercial 
electrical telegraph 
created by 
Cooke and 
Wheatstone 
1963: DTMF 
enters service 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 4
RTCWeb 
There are a number of proprietary implementations that 
provide direct interactive rich communication using audio, 
video, collaboration, games, etc. between two peers' web-browsers. 
These are not interoperable, as they require 
non-standard extensions or plugins to work. There is a 
desire to standardize the basis for such communication so 
that interoperable communication can be established 
between any compatible browsers. 
Real-Time Communication in WEB-Browsers 
(rtcweb) 2013-03-13 charter 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 5
Real-time communication is hard 
10/3/2014 
Title Version No: 0.1/ Status: DRAFT 
6 
• VoIP has always suffered from a lack of standard profile 
– What QoS/feedback mechanisms should be supported 
– What codecs should be supported 
– What security mechanisms should be supported 
• This means that VoIP end-point vendors can all follow the 
specifications to the letter and still produce devices that do not 
interoperate 
• This makes the job of developing a VoIP end-point far harder than it 
should be 
• This makes large scale consumer use of VoIP difficult 
– You can’t just add real-time communications to your service - you have 
to really, really, need it for it to be worth doing 
RTCWeb provides this standard profile
WebRTC 
The mission of the Web Real-Time Communications 
Working Group, part of the Ubiquitous Web Applications 
Activity, is to define client-side APIs to enable Real-Time 
Communications in Web browsers. 
These APIs should enable building applications that can be 
run inside a browser, requiring no extra downloads or 
plugins, that allow communication between parties using 
audio, video and supplementary real-time communication, 
without having to use intervening servers (unless needed for 
firewall traversal, or for providing intermediary services). 
Web Real-Time Communications 
Working Group Charter 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 7
RTCWeb/WebRTC Architecture 
Your web 
app #1 
RTCWeb 
Voice Engine 
. . . 
WebRTC API 
Your web 
app #2 
WebRTC C++ API (PeerConnection) 
Session management / Abstract signalling (Session) 
G.711/OPUS Codec 
NetEQ for Voice 
Echo Canceller / 
Noise Reduction 
Video Engine 
H.264/VP8 Codec 
Video Jitter Buffer 
Image enhancements 
Transport 
Your web 
app #3 
SRTP 
Multiplexing 
P2P 
STUN + TURN + ICE 
Audio Capture/Render Video Capture Network I/O 
The web 
Your browser 
Based on the diagram from http://www.webrtc.org/reference/architecture 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 8
The signalling triangle (non-interoperable) 
Server 
UA Media 
UA 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 9
The signalling trapezoid (interoperable) 
Server 
Signalling Server 
UA Media 
UA 
The fact that something can interoperate does 
not mean it must, or will, interoperate 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 10
Interoperability could be bad for business 
• Good for consumers doesn’t always mean good 
for business 
– Why would, for example, Microsoft want Office 365 
users to easily collaborate with Google Docs users? 
• In other situations… 
It’s the API, stupid… 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 11
Picking the right API is the most important thing 
Personally, I’ve come around (the hard way) to a fairly pragmatic stance 
on this point, and I hesitate from declaring “it has to be” one approach 
for signaling protocol, or another. Let me explain: Initially, I must admit I 
used to obsess about what protocol the library was doing under the 
hood. In fact, full disclosure: when I first started looking into JavaScript 
WebRTC signaling, I thought SIP over WebSockets was a bad idea… 
However, after I did a few projects… I started to have a change of heart. 
I found that in using the library, I was able to accomplish the goals of my 
projects, and the JavaScript interface to the library could be just as 
simple and powerful as all the others. I couldn’t find the fatal flaw I was 
expecting, and it just worked. In all this I started to care less and less 
about what was happening on the wire between the library and the 
service, and more about what features of the service I could access 
through the API. 
… Most importantly, the best one for you is flexible enough to meet your 
security, architecture, and functional requirements (no matter what 
protocol it uses). 
webrtcH4cKS: What’s in a WebRTC JavaScript Library, 
Reid Stidolph 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 12
Acision SDK v1.0 for Android, iOS, and JavaScript 
3rd Party 
Identity 
Provider 
Single point-of-access: 
sdk.acision.com:44 
3 
Note: different identity 
providers work in different 
ways, some may require 
specific handling in the app 
itself (for example, show a 
Facebook login screen) 
Application 
Servers 
Amazon AWS 
Contact 
Management 
Library 
Auth/Bootstra 
p Library 
Messaging 
Library 
Presence 
Library 
WebRTC 
Library 
SMS In/Out through Acision Forge 
SuperNode 
REST over HTTPS (port 443) 
REST over HTTPS (port 443) 
HTTP Proxy 
and 
Load Balancer 
REST over HTTPS (port 443) 
Secure WebSockets (port 443) 
REST over HTTPS (port 443) 
Secure WebSockets (port 443) 
SIP over Secure WebSockets (port 443) 
TURN server 
SDK Wrapper 
Audio/video calls 
SIP trunk(s) 
TURN over 
TCP/UDP (port 
3478) and TLS (port 
Acision S5D34K9) 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 13
AcisionSDK Example: Android 
AcisionSdkConfiguration config = 
new AcisionSdkConfiguration("MYAPIKEY ", "user", "password"); 
config.setApplicationActivity(this); 
AcisionSdk acisionSdk = new AcisionSdk(config, new AcisionSdkCallbacks() { 
@Override 
public void onConnected(AcisionSdk acisionSdk) { 
session = acisionSdk.getWebrtc().connectTo("+441632960000", 
new WebrtcSessionCallbacks() { 
@Override 
public void onConnected() { 
session.setLocalView(localVideoView); 
session.setRemoteView(remoteVideoView); 
} 
}, new WebrtcOptions()); 
} 
}); 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 14
AcisionSDK Example: iOS 
ACSAcisionSDKConfiguration *config = 
[ACSAcisionSDKConfiguration acisionSDKConfigurationWithAPIKey:@"MYAPIKEY” 
username:@"user" password:@"password"]; 
self.acisionSDK = [ACSAcisionSDK acisionSDKWithConfig:config delegate:self]; 
- (void)onConnected:(ACSAcisionSDK *)acisionSDK { 
ACSAcisionSDKWebRTCOptions* options = 
[[ACSAcisionSDKWebRTCOptions alloc] init]; 
self.currentSession = 
[self.acisionSDK.webrtc connectTo: @"+441632960000”, 
delegate:self options:options]; 
} 
-(void)onConnected { 
self.currentSession.localView = self.localView; 
self.currentSession.remoteView = self.remoteView; 
} 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 15
AcisionSDK Example: JavaScript 
var lib = new AcisionSDK("MYAPIKEY", { 
onConnected: function() { 
var session = lib.webrtc.connect("+441632960000"); 
session.remoteVideoElement = document.getElementById("vidRemote"); 
session.remoteAudioElement = document.getElementById("audRemote"); 
} 
}, { 
username: "user", 
password: "password" 
} 
); 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 16
OAuth2 Authentication 
OAuth2 
Provider 
6 
Authentication Code 
Access Token 
8 
Request Identity 
Identity 
7 
9 
Acision SDK 
SDK Authenticate (APIKey) 
Authentication URL 
2 
Authentication Code 
1 
5 
Identity 
3 
4 
10 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 17
Acision SDK Contact Centre Demo 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 18
Acision SDK Contact Centre Demo 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 19
Some stats… 
• 60%+ of communication is non-verbal.. 
• Satisfied online customers are 70% more 
likely to purchase again. Human 
interactions matter!! 
© Acision 2014 20
Thank you 
Email: peter.dunkley@acision.com 
Twitter: @pdunkley 
10/3/2014 Title Version No: 0.1/ Status: DRAFT 21

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DevCon5 (July 2014) - Acision SDK

  • 1. DevCon5 (July 2014): Acision SDK Peter Dunkley, Technical Director, Acision Email: peter.dunkley@acision.com Twitter: @pdunkley 10/3/2014 Title Version No: 0.1/ Status: DRAFT 1
  • 2. Acision at-a-glance Heritage & history 20 year history 270 customers globally 2 trillion messages/ year Global presence & reach 35+ offices across 6 continents 850+ employees 100+ support team Innovation 1st to launch commercial SMSC Cloud & virtualized services Multi-Channel messaging + WebRTC Operational Excellence Data and insights Carrier grade security © Acision 2014 24/7 Support
  • 3. Evolution on the Web Sir Tim Berners-Lee creates HTML. Web -pages are static W3C produces the DOM1 specification 1990 1996 1998 2004 2011 Microsoft and Netscape introduce different mechanisms for DHTML Google uses Ajax in Gmail (W3C releases 1st draft in 2006) – the dawn of web-apps WebSocket and WebRTC implementations become available 10/3/2014 Title Version No: 0.1/ Status: DRAFT 3
  • 4. Revolution in Telecoms The revolution Before today the operators (big and small) had full control over real-time communications because it was hard to do and substantial infrastructure investment Claude Chappe was required. invented the optical telegraph Alexander Graham Bell patents the telephone From the 1960s onwards digital exchanges start to appear From the 1990s onwards voice started to be carried on technologies developed for data networks such as ATM and IP 1792 1837 1876 1919 1960s > 1990s > 2011 WebSocket and WebRTC implementations become available Rotary dial enters service First commercial electrical telegraph created by Cooke and Wheatstone 1963: DTMF enters service 10/3/2014 Title Version No: 0.1/ Status: DRAFT 4
  • 5. RTCWeb There are a number of proprietary implementations that provide direct interactive rich communication using audio, video, collaboration, games, etc. between two peers' web-browsers. These are not interoperable, as they require non-standard extensions or plugins to work. There is a desire to standardize the basis for such communication so that interoperable communication can be established between any compatible browsers. Real-Time Communication in WEB-Browsers (rtcweb) 2013-03-13 charter 10/3/2014 Title Version No: 0.1/ Status: DRAFT 5
  • 6. Real-time communication is hard 10/3/2014 Title Version No: 0.1/ Status: DRAFT 6 • VoIP has always suffered from a lack of standard profile – What QoS/feedback mechanisms should be supported – What codecs should be supported – What security mechanisms should be supported • This means that VoIP end-point vendors can all follow the specifications to the letter and still produce devices that do not interoperate • This makes the job of developing a VoIP end-point far harder than it should be • This makes large scale consumer use of VoIP difficult – You can’t just add real-time communications to your service - you have to really, really, need it for it to be worth doing RTCWeb provides this standard profile
  • 7. WebRTC The mission of the Web Real-Time Communications Working Group, part of the Ubiquitous Web Applications Activity, is to define client-side APIs to enable Real-Time Communications in Web browsers. These APIs should enable building applications that can be run inside a browser, requiring no extra downloads or plugins, that allow communication between parties using audio, video and supplementary real-time communication, without having to use intervening servers (unless needed for firewall traversal, or for providing intermediary services). Web Real-Time Communications Working Group Charter 10/3/2014 Title Version No: 0.1/ Status: DRAFT 7
  • 8. RTCWeb/WebRTC Architecture Your web app #1 RTCWeb Voice Engine . . . WebRTC API Your web app #2 WebRTC C++ API (PeerConnection) Session management / Abstract signalling (Session) G.711/OPUS Codec NetEQ for Voice Echo Canceller / Noise Reduction Video Engine H.264/VP8 Codec Video Jitter Buffer Image enhancements Transport Your web app #3 SRTP Multiplexing P2P STUN + TURN + ICE Audio Capture/Render Video Capture Network I/O The web Your browser Based on the diagram from http://www.webrtc.org/reference/architecture 10/3/2014 Title Version No: 0.1/ Status: DRAFT 8
  • 9. The signalling triangle (non-interoperable) Server UA Media UA 10/3/2014 Title Version No: 0.1/ Status: DRAFT 9
  • 10. The signalling trapezoid (interoperable) Server Signalling Server UA Media UA The fact that something can interoperate does not mean it must, or will, interoperate 10/3/2014 Title Version No: 0.1/ Status: DRAFT 10
  • 11. Interoperability could be bad for business • Good for consumers doesn’t always mean good for business – Why would, for example, Microsoft want Office 365 users to easily collaborate with Google Docs users? • In other situations… It’s the API, stupid… 10/3/2014 Title Version No: 0.1/ Status: DRAFT 11
  • 12. Picking the right API is the most important thing Personally, I’ve come around (the hard way) to a fairly pragmatic stance on this point, and I hesitate from declaring “it has to be” one approach for signaling protocol, or another. Let me explain: Initially, I must admit I used to obsess about what protocol the library was doing under the hood. In fact, full disclosure: when I first started looking into JavaScript WebRTC signaling, I thought SIP over WebSockets was a bad idea… However, after I did a few projects… I started to have a change of heart. I found that in using the library, I was able to accomplish the goals of my projects, and the JavaScript interface to the library could be just as simple and powerful as all the others. I couldn’t find the fatal flaw I was expecting, and it just worked. In all this I started to care less and less about what was happening on the wire between the library and the service, and more about what features of the service I could access through the API. … Most importantly, the best one for you is flexible enough to meet your security, architecture, and functional requirements (no matter what protocol it uses). webrtcH4cKS: What’s in a WebRTC JavaScript Library, Reid Stidolph 10/3/2014 Title Version No: 0.1/ Status: DRAFT 12
  • 13. Acision SDK v1.0 for Android, iOS, and JavaScript 3rd Party Identity Provider Single point-of-access: sdk.acision.com:44 3 Note: different identity providers work in different ways, some may require specific handling in the app itself (for example, show a Facebook login screen) Application Servers Amazon AWS Contact Management Library Auth/Bootstra p Library Messaging Library Presence Library WebRTC Library SMS In/Out through Acision Forge SuperNode REST over HTTPS (port 443) REST over HTTPS (port 443) HTTP Proxy and Load Balancer REST over HTTPS (port 443) Secure WebSockets (port 443) REST over HTTPS (port 443) Secure WebSockets (port 443) SIP over Secure WebSockets (port 443) TURN server SDK Wrapper Audio/video calls SIP trunk(s) TURN over TCP/UDP (port 3478) and TLS (port Acision S5D34K9) 10/3/2014 Title Version No: 0.1/ Status: DRAFT 13
  • 14. AcisionSDK Example: Android AcisionSdkConfiguration config = new AcisionSdkConfiguration("MYAPIKEY ", "user", "password"); config.setApplicationActivity(this); AcisionSdk acisionSdk = new AcisionSdk(config, new AcisionSdkCallbacks() { @Override public void onConnected(AcisionSdk acisionSdk) { session = acisionSdk.getWebrtc().connectTo("+441632960000", new WebrtcSessionCallbacks() { @Override public void onConnected() { session.setLocalView(localVideoView); session.setRemoteView(remoteVideoView); } }, new WebrtcOptions()); } }); 10/3/2014 Title Version No: 0.1/ Status: DRAFT 14
  • 15. AcisionSDK Example: iOS ACSAcisionSDKConfiguration *config = [ACSAcisionSDKConfiguration acisionSDKConfigurationWithAPIKey:@"MYAPIKEY” username:@"user" password:@"password"]; self.acisionSDK = [ACSAcisionSDK acisionSDKWithConfig:config delegate:self]; - (void)onConnected:(ACSAcisionSDK *)acisionSDK { ACSAcisionSDKWebRTCOptions* options = [[ACSAcisionSDKWebRTCOptions alloc] init]; self.currentSession = [self.acisionSDK.webrtc connectTo: @"+441632960000”, delegate:self options:options]; } -(void)onConnected { self.currentSession.localView = self.localView; self.currentSession.remoteView = self.remoteView; } 10/3/2014 Title Version No: 0.1/ Status: DRAFT 15
  • 16. AcisionSDK Example: JavaScript var lib = new AcisionSDK("MYAPIKEY", { onConnected: function() { var session = lib.webrtc.connect("+441632960000"); session.remoteVideoElement = document.getElementById("vidRemote"); session.remoteAudioElement = document.getElementById("audRemote"); } }, { username: "user", password: "password" } ); 10/3/2014 Title Version No: 0.1/ Status: DRAFT 16
  • 17. OAuth2 Authentication OAuth2 Provider 6 Authentication Code Access Token 8 Request Identity Identity 7 9 Acision SDK SDK Authenticate (APIKey) Authentication URL 2 Authentication Code 1 5 Identity 3 4 10 10/3/2014 Title Version No: 0.1/ Status: DRAFT 17
  • 18. Acision SDK Contact Centre Demo 10/3/2014 Title Version No: 0.1/ Status: DRAFT 18
  • 19. Acision SDK Contact Centre Demo 10/3/2014 Title Version No: 0.1/ Status: DRAFT 19
  • 20. Some stats… • 60%+ of communication is non-verbal.. • Satisfied online customers are 70% more likely to purchase again. Human interactions matter!! © Acision 2014 20
  • 21. Thank you Email: peter.dunkley@acision.com Twitter: @pdunkley 10/3/2014 Title Version No: 0.1/ Status: DRAFT 21