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WebRTC on Mobile
Buşra Deniz
@busradeniz
#webrtc
About Me
Buşra Deniz busradeniz@facebook
busradeniz@twitter
busradeniz89@gmail.com
busradeniz.com
Certified Scrum Master @Netaş 

Mobile Software Developer @Netaş
Co-Organizer @WTM Istanbul
Co-Organizer @GDG Istanbul
Student in SWE @Bogazici University
#webrtc
Agenda
01
02
03
04
What is WebRTC ?
What exactly is WebRTC and what does it solve ?
WebRTC Architecture & API’s
Three Main API
Servers & Protocols
WebRTC needs servers
05
Building a WebRTC App
How can we include webRTC to our mobile apps?
Useful Links
WebRTC
#webrtc
How can I make multimedia call 

within my web apps?
#webrtc
How can I make multimedia call 

within my web apps?
Use Session Initiation Protocol (SIP) ??!…
- Has to use a browser plugin for SIP stack thus there is no out of the box
SIP support inside web browsers
- My website will not function properly unless web users install my plugin
- Have to utilize SIP equipment on server side

- Have to deal with highly complex SIP signaling during development cycle
#webrtc
WebRTC
Plugin Free !!
Abstract Signaling
- WebRTC is embedded in browsers no plugin required
- Security is provided by browser vendors web users do not have
to trust third party plugins
- Web users do not have to deal with plugin installation, web
apps will function properly with the embedded WebRTC
support
- No sophisticated telecom expertise is necessary to be able to
provide multimedia communication functionality WebRTC
- No need for sophisticated telecom equipment for backend
#webrtc
Wait !! Need Browser to Mobile device
communication for a complete solution ???
#webrtc
WebRTC on Mobile
Native iOS & Android library
Chrome for Android
Firefox for Android
Opera for Android
#webrtc
Why WebRTC on mobile ?
Tango
WeChat
Line
Skype
Facebook
WhatsApp
LinkedIn
Twitter
Facebook
Android
iOS
0 200 400 600 800 1000 1200 1400 1600 Millions
Statistics based on PrioriData, Oct 2014
Social and Messaging
Application
Downloads
#webrtc
WebRTC on > 6bn devices by 2019
0
500
1000
1500
2000
2500
3000
3500
4000
4500
Q4’12
Q1’13
Q2’13
Q3’13
Q4’13
Q1’14
Q2’14
Q3’14
Q4’14
Q1’15
Q2’15
Q3’15
Q4’15
Q1’16
Q2’16
Q3’16
Q4’16
Smart Phones
Tablets
PCs
#webrtc
Free & Open Source
WebRTC is an open source project.

WebRTC can be used by anyone for
anything without any payment
Plugin Free
WebRTC is embedded in browsers
no plugin required
For non-VoIP developers
WebRTC hides complexity of media
streaming, encoding/decoding media
processes. Any web/mobile developer who
has no telco background can develop RTC
app.
Low Cost
It enables peer to peer communication
and you can add multi media to your
application without any need to media
server
What makes WebRTC different ?
#webrtc
WebRTC means that …


“ WebRTC is a new front in the long war
for an open and unencumbered web”



Brendan Eich

- Mozilla CTO and inventor of Javascript
History of WebRTC
#webrtc
History of WebRTC
May 2011
Nov 2011
Jan 2013
Feb 2013
May 2013
Jun 2013
Source BlogGeek.me
WebRTC Announced
Google releases WebRTC source
code for the first time under a
permissive BSD license
Chrome 23 adds
No optional flag is required. Data
channel capabilities not supported
Firefox 20 adds WebRTC
First release of Firefox
supporting WebRTC. Comes
with getUserMedia support
only, which gives access to the
local camera
Interoperability
Initial interoperability between Chrome and
Firefox browsers achieved. This is still early
on in the process, so things still don’t work as
expected, but this is an indication of things to
come
Firefox 22 released
First Firefox release that
officially supports the ability to
make video calls as well as use
the Data channel API
ObjC&Java Bindings
Objective C and Java bindings for
WebRTC announced.
#webrtc
History of WebRTC
Source BlogGeek.me
Aug 2013
Sep 2013
Oct 2013
Mar 2014
May 2014
Oct 2014
Microsoft announced ORTC
support
Microsoft officially announced
plans to support ORTC
(WebRTC 1.1) in a future
release of IE
Chrome for Android
Chrome 29 for Android now fully
supports WebRTC
Firefox for Android
Firefox for Android supports WebRTC
Opera 18 Beta intros
First Opera released based on
Chromium, providing immediate
WebRTC support
Opera for Android with

WebRTC
Opera 20 for Android has WebRTC in
GA
Microsoft promises to
support GUM
Microsoft indicates in its IE status
page that it plans to support
GetUserMedia APIs in its next version
of Internet Explorer
#webrtc
2015 - Year of WebRTC ??
#webrtc
2014 WebRTC Solutions
Source BlogGeek.me
#webrtc
2015 WebRTC Solutions
Source BlogGeek.me
#webrtc
2015 WebRTC News
36
Ignored real time communications ( and still ignoring it)
Aquired ScreenHero, who use WebRTC for screen sharing
Has integration with WebRTC vendors
appear.in, Hangouts, Room etc.
Rebranding of Lync to Skype for Business
Announced Skype for Web
Uses WebRTC as a browser access point to Skype
Anounces WebRTC usage in Jan5, 2015
First U.S Carrier to Launch Commercial Support WebRTC
Provides its own WebRTC API with several enhancements
Introduced video calling using WebRTC PaaS
Didn’t want users to “migrate” to Skype for interactions
In first 3 months 150,000 calls
with 2,500,000 minutes of video calls
Facebook Announces in April, 2015
Messenger will use WebRTC for voice and video call
Before WebRTC, Messenger uses Skype as VoIP
#webrtc
Popular Verticals
Source BlogGeek.me
Healthcare
Experts Market
Education
Gaming
Job interviews
Surveillance
Financial
#webrtc
WebRTC Global Developer Statistics
Source WebRTC Global Submit London 2015
2,000+ visitors per day to the developer docs

1,000+ visitors per day to the sample code
4,000+ developers subscribed to the
WebRTC mailing list
300K+ views of Webrtc.org developer
videos
600+ companies building on WebRTC

40+ external contributors
3B+ total downloads of WebRTC-
powered mobile apps
Supported Platforms
#webrtc
Supported Platforms
#webrtc
1.500.000.000+ 

WebRTC Endpoints
Architecture & API’s
#webrtc
WebRTC is both an open source project and a
standard specification
WebRTC 1.0 webrtc.org
the standart specification
not yet completed
handled by the IETF and W3C
the open source project by Google
implementation of WebRTC
spesification
can be used by anyone for anything
#webrtc
WebRTC
Session Management / Abstract signalling
WebRTC C++ API
Audio Capture/Render
Voice Engine
Echo Cancler/Noise
Reduction
NetEQ for voice
Codecs (ISAC /Opus)
Video Engine
Image enhancements
Video jitter buffer
Codecs ( VP8)
Video Capture
Transport
P2P 

STUN + TURN + ICE
Multiplexing
SRTP
Network I/O
WebRTC Architecture
JavaScript / ObjC / Java API
#webrtc
WebRTC API’s
MediaStream
PeerConnection
DataChannel
Means audio and/or video stream
Can contain multiple track
Can access camera and microphone Peer to peer multi media communication
Encoding&Decoding media (codecs)
Sends media over the network
Security
Bandwidth management
P2P communication of arbitrary data
Low latency
Unreliable or reliable
Secure
Servers & Protocols
#webrtc
Abstract Signaling
Signaling methods and protocols are not specified by WebRTC

- To avoid redundancy
- To maximize compatibility with established technologies
Signaling is the process of coordinating communication. In order to set up a
call, clients need to exchange some information :
- Session control messages used to open or close communication
- Media metadata such as codecs, bandwidth and media types
- Key data, used to establish secure connections
- Network data, such as host IP address and port
#webrtc
Signaling
Caller Callee
Media
Session Description Session Description
WebRTC
App
WebRTC
App
Offer Answer
OfferAnswer
#webrtc
A call scenario (offer/answer mechanism)
create offer
set offer
send offer set remote offer
create answer
send answerset remote answer
SDP
SDP
Caller Callee
#webrtc
SDP (Session Description Protocol)
SDP
Session Description
Time Description
Media Description
Purely a format for session description, intended to use different transport protocols as SIP, RTSP, HTTP
v=0

o=- 7614219274584779017 2 IN IP4 127.0.0.1
t=0 0
m=audio 1 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:W2TGCZw2NZHuwlnf
a=ice-pwd:xdQEccP40E+P0L5qTyzDgfmW
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9c1AHz27dZ9xPI91YNfSlI67/
EMkjHHIHORiClQe
a=rtpmap:111 opus/48000/2
m=video 0 RTP/AVP 31
a=rtpmap:31 H261/90000

…
#webrtc
Signaling
Signaling
Peer Peer
After signaling…
needs more servers :/
It’s ideal world
media
#webrtc
Signaling
Signaling
Peer Peer
But the real world !!
? SignSign
NAT
NAT
STUN
media
STUN
- What is my public IP address
- Media peer to peer
- Cheap
- It always works but some cases ??STUN
#webrtc
Signaling
Signaling
Peer Peer
Cases STUN does not work
? SignSign
NAT
STUN STUN
TURN TURN
NAT
media
- Used to relay audio/video/data
streaming between peers
- Data sent through server,
uses server bandwidth
TURN - The call works in almost all
environments with using TURN
#webrtc
ICE Framework
- WebRTC uses ICE to overcome the complexitiy of real-world networking.
- Tries to find best path to connect peers in parallel
Peer Peer
STUN
TURN
Sign
#webrtc
TURN vs P2P
14% 86%
- According to Google I/O 2013
Relayed P2P
8% 92%
- According to Bistri report
Relayed P2P
Building WebRTC

App
#webrtc
WebRTC for iOS and Android
#webrtc
WebRTC for iOS and Android
https://chromium.googlesource.com/external/webrtc/+/master
https://github.com/webrtc
#webrtc
Set up a call
Remote Peer Peer Connection Factory Peer ConnectionApplication
Create
Connection Factory
Create Peer Connection
Create Local Media Stream
Create Local Video Track
Create Local Audio Track
Add Stream
Commit Stream Changes
On Signalling Message (offer)
Send Offer To Remote Peer
Get Answer From Remote Peer
Process Signalling Message (answer)
Media
#webrtc
Receive Call
Remote Peer Peer Connection Factory Peer ConnectionApplication
Create
Connection Factory
Create Peer Connection
Create Local Media Stream
Create Local Video Track
Create Local Audio Track
Add Stream
Commit Stream Changes
On Signalling Message (answer)
Send Offer To Remote Peer
Media
On Add Stream
Receive Answer From Remote Peer
Process Signalling Message (offer)
Useful Links
#webrtc
Useful Links
http://www.webrtc.org/
http://www.html5rocks.com/en/tutorials/webrtc/basics/
http://bloggeek.me/
discuss-webrtc
@webrtc
@juberti & @sw12
Thanks
See You Next Time !!!

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WebRTC on Mobile: A Guide to Building Multimedia Apps

  • 1. WebRTC on Mobile Buşra Deniz @busradeniz
  • 2. #webrtc About Me Buşra Deniz busradeniz@facebook busradeniz@twitter busradeniz89@gmail.com busradeniz.com Certified Scrum Master @Netaş 
 Mobile Software Developer @Netaş Co-Organizer @WTM Istanbul Co-Organizer @GDG Istanbul Student in SWE @Bogazici University
  • 3. #webrtc Agenda 01 02 03 04 What is WebRTC ? What exactly is WebRTC and what does it solve ? WebRTC Architecture & API’s Three Main API Servers & Protocols WebRTC needs servers 05 Building a WebRTC App How can we include webRTC to our mobile apps? Useful Links
  • 5. #webrtc How can I make multimedia call 
 within my web apps?
  • 6. #webrtc How can I make multimedia call 
 within my web apps? Use Session Initiation Protocol (SIP) ??!… - Has to use a browser plugin for SIP stack thus there is no out of the box SIP support inside web browsers - My website will not function properly unless web users install my plugin - Have to utilize SIP equipment on server side
 - Have to deal with highly complex SIP signaling during development cycle
  • 7. #webrtc WebRTC Plugin Free !! Abstract Signaling - WebRTC is embedded in browsers no plugin required - Security is provided by browser vendors web users do not have to trust third party plugins - Web users do not have to deal with plugin installation, web apps will function properly with the embedded WebRTC support - No sophisticated telecom expertise is necessary to be able to provide multimedia communication functionality WebRTC - No need for sophisticated telecom equipment for backend
  • 8. #webrtc Wait !! Need Browser to Mobile device communication for a complete solution ???
  • 9. #webrtc WebRTC on Mobile Native iOS & Android library Chrome for Android Firefox for Android Opera for Android
  • 10. #webrtc Why WebRTC on mobile ? Tango WeChat Line Skype Facebook WhatsApp LinkedIn Twitter Facebook Android iOS 0 200 400 600 800 1000 1200 1400 1600 Millions Statistics based on PrioriData, Oct 2014 Social and Messaging Application Downloads
  • 11. #webrtc WebRTC on > 6bn devices by 2019 0 500 1000 1500 2000 2500 3000 3500 4000 4500 Q4’12 Q1’13 Q2’13 Q3’13 Q4’13 Q1’14 Q2’14 Q3’14 Q4’14 Q1’15 Q2’15 Q3’15 Q4’15 Q1’16 Q2’16 Q3’16 Q4’16 Smart Phones Tablets PCs
  • 12. #webrtc Free & Open Source WebRTC is an open source project.
 WebRTC can be used by anyone for anything without any payment Plugin Free WebRTC is embedded in browsers no plugin required For non-VoIP developers WebRTC hides complexity of media streaming, encoding/decoding media processes. Any web/mobile developer who has no telco background can develop RTC app. Low Cost It enables peer to peer communication and you can add multi media to your application without any need to media server What makes WebRTC different ?
  • 13. #webrtc WebRTC means that … 
 “ WebRTC is a new front in the long war for an open and unencumbered web”
 
 Brendan Eich
 - Mozilla CTO and inventor of Javascript
  • 15. #webrtc History of WebRTC May 2011 Nov 2011 Jan 2013 Feb 2013 May 2013 Jun 2013 Source BlogGeek.me WebRTC Announced Google releases WebRTC source code for the first time under a permissive BSD license Chrome 23 adds No optional flag is required. Data channel capabilities not supported Firefox 20 adds WebRTC First release of Firefox supporting WebRTC. Comes with getUserMedia support only, which gives access to the local camera Interoperability Initial interoperability between Chrome and Firefox browsers achieved. This is still early on in the process, so things still don’t work as expected, but this is an indication of things to come Firefox 22 released First Firefox release that officially supports the ability to make video calls as well as use the Data channel API ObjC&Java Bindings Objective C and Java bindings for WebRTC announced.
  • 16. #webrtc History of WebRTC Source BlogGeek.me Aug 2013 Sep 2013 Oct 2013 Mar 2014 May 2014 Oct 2014 Microsoft announced ORTC support Microsoft officially announced plans to support ORTC (WebRTC 1.1) in a future release of IE Chrome for Android Chrome 29 for Android now fully supports WebRTC Firefox for Android Firefox for Android supports WebRTC Opera 18 Beta intros First Opera released based on Chromium, providing immediate WebRTC support Opera for Android with
 WebRTC Opera 20 for Android has WebRTC in GA Microsoft promises to support GUM Microsoft indicates in its IE status page that it plans to support GetUserMedia APIs in its next version of Internet Explorer
  • 17. #webrtc 2015 - Year of WebRTC ??
  • 20. #webrtc 2015 WebRTC News 36 Ignored real time communications ( and still ignoring it) Aquired ScreenHero, who use WebRTC for screen sharing Has integration with WebRTC vendors appear.in, Hangouts, Room etc. Rebranding of Lync to Skype for Business Announced Skype for Web Uses WebRTC as a browser access point to Skype Anounces WebRTC usage in Jan5, 2015 First U.S Carrier to Launch Commercial Support WebRTC Provides its own WebRTC API with several enhancements Introduced video calling using WebRTC PaaS Didn’t want users to “migrate” to Skype for interactions In first 3 months 150,000 calls with 2,500,000 minutes of video calls Facebook Announces in April, 2015 Messenger will use WebRTC for voice and video call Before WebRTC, Messenger uses Skype as VoIP
  • 21. #webrtc Popular Verticals Source BlogGeek.me Healthcare Experts Market Education Gaming Job interviews Surveillance Financial
  • 22. #webrtc WebRTC Global Developer Statistics Source WebRTC Global Submit London 2015 2,000+ visitors per day to the developer docs
 1,000+ visitors per day to the sample code 4,000+ developers subscribed to the WebRTC mailing list 300K+ views of Webrtc.org developer videos 600+ companies building on WebRTC
 40+ external contributors 3B+ total downloads of WebRTC- powered mobile apps
  • 27. #webrtc WebRTC is both an open source project and a standard specification WebRTC 1.0 webrtc.org the standart specification not yet completed handled by the IETF and W3C the open source project by Google implementation of WebRTC spesification can be used by anyone for anything
  • 28. #webrtc WebRTC Session Management / Abstract signalling WebRTC C++ API Audio Capture/Render Voice Engine Echo Cancler/Noise Reduction NetEQ for voice Codecs (ISAC /Opus) Video Engine Image enhancements Video jitter buffer Codecs ( VP8) Video Capture Transport P2P 
 STUN + TURN + ICE Multiplexing SRTP Network I/O WebRTC Architecture JavaScript / ObjC / Java API
  • 29. #webrtc WebRTC API’s MediaStream PeerConnection DataChannel Means audio and/or video stream Can contain multiple track Can access camera and microphone Peer to peer multi media communication Encoding&Decoding media (codecs) Sends media over the network Security Bandwidth management P2P communication of arbitrary data Low latency Unreliable or reliable Secure
  • 31. #webrtc Abstract Signaling Signaling methods and protocols are not specified by WebRTC
 - To avoid redundancy - To maximize compatibility with established technologies Signaling is the process of coordinating communication. In order to set up a call, clients need to exchange some information : - Session control messages used to open or close communication - Media metadata such as codecs, bandwidth and media types - Key data, used to establish secure connections - Network data, such as host IP address and port
  • 32. #webrtc Signaling Caller Callee Media Session Description Session Description WebRTC App WebRTC App Offer Answer OfferAnswer
  • 33. #webrtc A call scenario (offer/answer mechanism) create offer set offer send offer set remote offer create answer send answerset remote answer SDP SDP Caller Callee
  • 34. #webrtc SDP (Session Description Protocol) SDP Session Description Time Description Media Description Purely a format for session description, intended to use different transport protocols as SIP, RTSP, HTTP v=0
 o=- 7614219274584779017 2 IN IP4 127.0.0.1 t=0 0 m=audio 1 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126 c=IN IP4 0.0.0.0 a=rtcp:1 IN IP4 0.0.0.0 a=ice-ufrag:W2TGCZw2NZHuwlnf a=ice-pwd:xdQEccP40E+P0L5qTyzDgfmW a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9c1AHz27dZ9xPI91YNfSlI67/ EMkjHHIHORiClQe a=rtpmap:111 opus/48000/2 m=video 0 RTP/AVP 31 a=rtpmap:31 H261/90000
 …
  • 35. #webrtc Signaling Signaling Peer Peer After signaling… needs more servers :/ It’s ideal world media
  • 36. #webrtc Signaling Signaling Peer Peer But the real world !! ? SignSign NAT NAT STUN media STUN - What is my public IP address - Media peer to peer - Cheap - It always works but some cases ??STUN
  • 37. #webrtc Signaling Signaling Peer Peer Cases STUN does not work ? SignSign NAT STUN STUN TURN TURN NAT media - Used to relay audio/video/data streaming between peers - Data sent through server, uses server bandwidth TURN - The call works in almost all environments with using TURN
  • 38. #webrtc ICE Framework - WebRTC uses ICE to overcome the complexitiy of real-world networking. - Tries to find best path to connect peers in parallel Peer Peer STUN TURN Sign
  • 39. #webrtc TURN vs P2P 14% 86% - According to Google I/O 2013 Relayed P2P 8% 92% - According to Bistri report Relayed P2P
  • 41. #webrtc WebRTC for iOS and Android
  • 42. #webrtc WebRTC for iOS and Android https://chromium.googlesource.com/external/webrtc/+/master https://github.com/webrtc
  • 43. #webrtc Set up a call Remote Peer Peer Connection Factory Peer ConnectionApplication Create Connection Factory Create Peer Connection Create Local Media Stream Create Local Video Track Create Local Audio Track Add Stream Commit Stream Changes On Signalling Message (offer) Send Offer To Remote Peer Get Answer From Remote Peer Process Signalling Message (answer) Media
  • 44. #webrtc Receive Call Remote Peer Peer Connection Factory Peer ConnectionApplication Create Connection Factory Create Peer Connection Create Local Media Stream Create Local Video Track Create Local Audio Track Add Stream Commit Stream Changes On Signalling Message (answer) Send Offer To Remote Peer Media On Add Stream Receive Answer From Remote Peer Process Signalling Message (offer)