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WebRTC Standards  &  
Implementation  Q&A
Amir	
  Zmora
TheNewDialTone
Dan	
  Burnett
StandardsPlay
Alex	
  Gouaillard
WebRTC by	
  Dr	
  Alex	
  /	
  Citrix
Watch  video  recording  of  this  session
http://ccst.io/e/webrtcstandards9
Session  sponsored  by
WebRTC.ventures	
  is	
  a	
  custom	
  design	
  and	
  development	
  shop	
  dedicated	
  to	
  building	
  WebRTC based	
  applications	
  
for	
  web	
  and	
  mobile.	
  We	
  have	
  built	
  end-­‐to-­‐end	
  broadcast	
  solutions	
  for	
  events	
  and	
  entertainment	
  clients,	
  
telehealth solutions	
  for	
  multiple	
  clients,	
  live	
  support	
  tools,	
  as	
  well	
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  communication	
  tools	
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  variety	
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applications. WebRTC.ventures is	
  a	
  recognized	
  development	
  partner	
  of	
  TokBox and	
  has	
  also	
  built	
  native	
  
WebRTC solutions	
  
Session  sponsored  by
Blacc	
  Spot	
  Media is	
  comprised	
  of	
  a	
  collaborative	
  team	
  of	
  designers,	
  developers	
  and	
  thought	
  leaders	
  specializing	
  
in	
  Web-­‐Real	
  Time	
  Communications	
  (WebRTC).	
  Blacc Spot	
  Media	
  was	
  founded	
  in	
  April	
  2011	
  with	
  the	
  goal	
  of	
  
creating	
  innovative	
  solutions	
  using	
  new	
  technologies	
  such	
  as	
  WebRTC.	
  Our	
  team	
  works	
  on	
  strategic	
  projects	
  
developing	
  custom	
  applications,	
  enterprise	
  platforms	
  and	
  mobile	
  applications	
  for	
  a	
  wide	
  array	
  of	
  clients.
We  use  CrowdCast….It’s  WebRTC
WebRTCStandards.info
Sponsored  by
About  Us
• Amir  Zmora • Dan  Burnett • Alex  Gouaillard
Save  The  Date:  Nov  21st
Register  Now:  http://ccst.io/e/webrtcstandards10
Next  Session
On  The  Agenda:
W3C  WebRTC JS  API  Test  Platform  &  
Updates  from  W3C  Lisbon  Meeting
Updates  from  W3C  Lisbon  Meeting
The  meeting
• 2  days
• 20  active  participants,  many  observers  (yes,  Apple  folks  were  there!)
• Mostly  steady,  uncontroversial  discussion  and  progress
• A  few  contentious  issues  
Major  discussion  topics
• Queuing  and  order  of  operations  -­-­ tricky
• Errors  – we  need  to  define  new  error  objects  beyond  what  WebIDL supports,  both  for  
createOffer/Answer  and  IdP
• Permissions  – Lots  of  new  Media  Capture  and  Streams  text  for  how  indicators  will  work
• Senders/Receivers/Transceivers  – fixing  the  bugs  and  edge  cases  we  keep  finding
• Certificates  – getting  fingerprints  of  certificates
• ICE  issues  – cleanup  of  naming  and  definitions,  particularly  for  RTCIceGatheringState.    
Also  tricky  Trickle  ICE  end  of  candidates  issue
• Largely  status  checks:  Stats,  DTMF,  Test  Suite(s),  Mediacapture-­depth,  Mediacapture-­
output
• Agenda,  slides,  and  minutes  at  https://www.w3.org/2011/04/webrtc/wiki/September_22_-­
_23_2016
Tricky  stuff
• Queuing  and  order  of  operations
pc.addTrack(trackX, stream);
pc.createOffer().then(offer => { ... })
pc.addTrack(trackY, stream);
• Trickle  ICE  end-­of-­candidates  indication
• Many  options,  none  perfect
• Two  primary  options  discussed,  but  no  resolution
EoC issue:    the  way  it  is  today
pc.onicecandidate =  function  (e)  {
signalingChannel.send(JSON.stringify({  "candidate":  e.candidate }));;
};;
signalingChannel.onmessage =  function  (e)  {
var message  =  JSON.parse(e.data);;
pc.addIceCandidate(message.candidate);;
};;
Null	
  value	
  means	
  end	
  of	
  candidates
…	
  but we	
  need	
  to	
  give	
  the	
  ufrag
EoC issue:    one  option  (modify  candidate)
Regular  candidate:
{sdpMid:  "audio",
candidate:  "candidate:...  ufrag 1111"}
End  of  candidates:
{sdpMid:  "audio",
candidate:  "candidate:...  ufrag 1111  trickle  done"}
Some	
  impls.	
  do	
  this	
  already
So	
  we	
  could	
  just	
  add	
  to	
  it
…	
  but it	
  might	
  break	
  current	
  code	
  that	
  parses	
  
the	
  candidate
EoC issue:    another  option  (new  event  &  method)
Regular  candidate  event
{candidate:  {
sdpMid:  "audio",
ufrag:  "1111",
candidate:  "candidate:..."},
}
.addIceCandidate(
{sdpMid:  "audio",  ufrag:  "1111",
candidate:  "..."});;
End  of  candidates  event
{sdpMid:  "audio",
ufrag:  "1111"}
.endIceCandidate(
{sdpMid:  "audio",  ufrag:  "1111"});;
Status  updates
• Media  Capture  and  Streams  spec
• Permissions  stuff  is  main  outstanding  topic  before  another  CR,  then  a  rapid  PR.    We  currently  
now  have  a  normative  dependency  on  the  separate  Permissions  Specification  work  L
• WebRTC  spec
• Want  a  first  CR  before  end  of  year
• More  likely  is  1st quarter  2017
Beyond  WebRTC  1.0
• Background  PeerConnection processing  (likely  in  ServiceWorkers),  needed  for
• keeping  a  connection  alive  even  when  the  user  is  not  directly  engaging  in  the  current  tab
• being  ready  to  accept  calls  on  short  notice  without  requiring  an  open  browser  tab
• interacting  with  IoT ‘things’  that  may  only  need  a  simple  data  channel,  for  example,  where  there  
is  no  need  for  audio  and  video  at  all
• Better  control  over  more  "types"  of  video
• spherical  video,  stereo  video
• augmented  reality
• multicast,  simulcast,  and  pluggable  codecs
• far-­end  camera  control
• Improved  network  handling
• better  network  interface  handoff  without  call  drop,  e.g.  mobile  to  WiFi and  vice  versa
• support  for  QUIC  for  faster  overall  setup
• fuller  support  for  ICE,  and  support  for  the  improved  version  of  ICE  being  developed  at  the  IETF.    
This  may  even  allow  ICE  to  be  implemented  via  a  JS  library
W3C  WebRTC JS  API  Test  Platform
Satellite  view
Specification
TR
Working	
  draft
Stable	
  draft
(Last	
  Call)
Implementors	
  feedback	
  (CR) Standard	
  (Rec) Test	
  Suite
WebRTC	
  1.0:	
  Real-­‐time	
  Communication	
  Between	
  Browsers 28-­‐Feb-­‐16 N/A Q2	
  2016? Y
Media	
  Capture	
  and	
  Streams	
  (getUserMedia) 12-­‐Feb-­‐15 14-­‐Apr-­‐15 19-­‐May-­‐16 Y
MediaStream	
  Recording 8-­‐Sep-­‐15 N/A N
MediaStream	
  Image	
  Capture 27-­‐Jan-­‐15 N/A N
Media	
  Capture	
  Depth	
  Stream	
  Extensions 26-­‐Feb-­‐16 N/A N
Media	
  Capture	
  from	
  DOM	
  Elements 19-­‐Feb-­‐15 N/A N
Audio	
  output	
  devices	
  API 10-­‐Feb-­‐15 N/A N
Identifiers	
  for	
  WebRTC's	
  Statistics	
  API 6-­‐Feb-­‐15 N/A N
Screen	
  Capture 10-­‐Feb-­‐15 N/A N
APIs  View
Media	
  Capture	
  and	
  Streams	
  (getUserMedia) isTested Size Coverage
MediaStream Y 179 95%
MediaStreamTrack Y 512 34%
Media	
  stream	
  in	
  Media	
  element Y 77 100%
Enumerating	
  Local	
  Media	
  Devices Y 159 45%
GUM Y 281 68%
Constraints N 136 0%
Error	
  Handling N 100 0%
Where  do  we  stand?
(march  2016)
WebRTC Testing  -­ DIY  1/3  – Small  Tools
• Here  is  a  presentation  about  the  state  of  the  art  of  webrtc  testing:
• http://www.slideshare.net/alexpiwi5/2016-­q1-­webrtc-­testing-­state-­of-­the-­art
• you  have  a  couple  of  diagnostic  tools  available  that  can  be  used/reproduced  (always  good  
to  have)
• -­ https://www.netscan.co/
• -­ https://test.webrtc.org/ (source  code  is  here)
• adapter.js has  its  own  testing  framework
• -­ https://github.com/webrtc/adapter
WebRTC Testing  – DIY  2/3  – Interop  &  system  
• the  most  advanced  work,  sponsored  by  IMTC.  It  allows  to  tests  you  app,  running  with  your  
infrastructure,  across  browsers,  browser  versions,  OS,  OS  versions,  .....  The  README  is  
nicely  written.
• -­ https://github.com/fippo/testbed
WebRTC Testing  – DIY  3/3  -­ Native
• For  native,  everybody  use  the  Google  native  stack  (webrtc.org)  with  its  binding,  and  the  
testing  is  explained  in  detail  on  my  blog:
• 1.  The  build  system  (here)
• 2.  Automatic  build  with  CMake (here)
• 3.  How  to  test  lib  webrtc  (here)
• 4.  How  to  Package  /  Install  lib  webrtc  (here)
• 5.  How  to  set  up  build  bots  for  WebRTC  (here)
• 6.  How  to  patch  automatically  lib  webrtc  (here)
• It  is  also  resumed  here  (some  slides  are  in  japanese):
• http://www.slideshare.net/alexpiwi5/testing-­and-­packaging-­webrtc-­stack
?
Save  The  Date:  Nov  21st
Register  Now:  http://ccst.io/e/webrtcstandards10
Next  Session
Session  sponsored  by
Thank  You
Amir	
  Zmora
TheNewDialTone
Dan	
  Burnett
StandardsPlay
Alex	
  Gouaillard
WebRTC by	
  Dr	
  Alex

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WebRTC Webinar & Q&A - W3C WebRTC JS API Test Platform & Updates from W3C Lisbon Meeting

  • 1. WebRTC Standards  &   Implementation  Q&A Amir  Zmora TheNewDialTone Dan  Burnett StandardsPlay Alex  Gouaillard WebRTC by  Dr  Alex  /  Citrix
  • 2. Watch  video  recording  of  this  session http://ccst.io/e/webrtcstandards9
  • 3. Session  sponsored  by WebRTC.ventures  is  a  custom  design  and  development  shop  dedicated  to  building  WebRTC based  applications   for  web  and  mobile.  We  have  built  end-­‐to-­‐end  broadcast  solutions  for  events  and  entertainment  clients,   telehealth solutions  for  multiple  clients,  live  support  tools,  as  well  as  communication  tools  for  a  variety  of  other   applications. WebRTC.ventures is  a  recognized  development  partner  of  TokBox and  has  also  built  native   WebRTC solutions  
  • 4. Session  sponsored  by Blacc  Spot  Media is  comprised  of  a  collaborative  team  of  designers,  developers  and  thought  leaders  specializing   in  Web-­‐Real  Time  Communications  (WebRTC).  Blacc Spot  Media  was  founded  in  April  2011  with  the  goal  of   creating  innovative  solutions  using  new  technologies  such  as  WebRTC.  Our  team  works  on  strategic  projects   developing  custom  applications,  enterprise  platforms  and  mobile  applications  for  a  wide  array  of  clients.
  • 7. About  Us • Amir  Zmora • Dan  Burnett • Alex  Gouaillard
  • 8. Save  The  Date:  Nov  21st Register  Now:  http://ccst.io/e/webrtcstandards10 Next  Session
  • 9. On  The  Agenda: W3C  WebRTC JS  API  Test  Platform  &   Updates  from  W3C  Lisbon  Meeting
  • 10. Updates  from  W3C  Lisbon  Meeting
  • 11. The  meeting • 2  days • 20  active  participants,  many  observers  (yes,  Apple  folks  were  there!) • Mostly  steady,  uncontroversial  discussion  and  progress • A  few  contentious  issues  
  • 12. Major  discussion  topics • Queuing  and  order  of  operations  -­-­ tricky • Errors  – we  need  to  define  new  error  objects  beyond  what  WebIDL supports,  both  for   createOffer/Answer  and  IdP • Permissions  – Lots  of  new  Media  Capture  and  Streams  text  for  how  indicators  will  work • Senders/Receivers/Transceivers  – fixing  the  bugs  and  edge  cases  we  keep  finding • Certificates  – getting  fingerprints  of  certificates • ICE  issues  – cleanup  of  naming  and  definitions,  particularly  for  RTCIceGatheringState.     Also  tricky  Trickle  ICE  end  of  candidates  issue • Largely  status  checks:  Stats,  DTMF,  Test  Suite(s),  Mediacapture-­depth,  Mediacapture-­ output • Agenda,  slides,  and  minutes  at  https://www.w3.org/2011/04/webrtc/wiki/September_22_-­ _23_2016
  • 13. Tricky  stuff • Queuing  and  order  of  operations pc.addTrack(trackX, stream); pc.createOffer().then(offer => { ... }) pc.addTrack(trackY, stream); • Trickle  ICE  end-­of-­candidates  indication • Many  options,  none  perfect • Two  primary  options  discussed,  but  no  resolution
  • 14. EoC issue:    the  way  it  is  today pc.onicecandidate =  function  (e)  { signalingChannel.send(JSON.stringify({  "candidate":  e.candidate }));; };; signalingChannel.onmessage =  function  (e)  { var message  =  JSON.parse(e.data);; pc.addIceCandidate(message.candidate);; };; Null  value  means  end  of  candidates …  but we  need  to  give  the  ufrag
  • 15. EoC issue:    one  option  (modify  candidate) Regular  candidate: {sdpMid:  "audio", candidate:  "candidate:...  ufrag 1111"} End  of  candidates: {sdpMid:  "audio", candidate:  "candidate:...  ufrag 1111  trickle  done"} Some  impls.  do  this  already So  we  could  just  add  to  it …  but it  might  break  current  code  that  parses   the  candidate
  • 16. EoC issue:    another  option  (new  event  &  method) Regular  candidate  event {candidate:  { sdpMid:  "audio", ufrag:  "1111", candidate:  "candidate:..."}, } .addIceCandidate( {sdpMid:  "audio",  ufrag:  "1111", candidate:  "..."});; End  of  candidates  event {sdpMid:  "audio", ufrag:  "1111"} .endIceCandidate( {sdpMid:  "audio",  ufrag:  "1111"});;
  • 17. Status  updates • Media  Capture  and  Streams  spec • Permissions  stuff  is  main  outstanding  topic  before  another  CR,  then  a  rapid  PR.    We  currently   now  have  a  normative  dependency  on  the  separate  Permissions  Specification  work  L • WebRTC  spec • Want  a  first  CR  before  end  of  year • More  likely  is  1st quarter  2017
  • 18. Beyond  WebRTC  1.0 • Background  PeerConnection processing  (likely  in  ServiceWorkers),  needed  for • keeping  a  connection  alive  even  when  the  user  is  not  directly  engaging  in  the  current  tab • being  ready  to  accept  calls  on  short  notice  without  requiring  an  open  browser  tab • interacting  with  IoT ‘things’  that  may  only  need  a  simple  data  channel,  for  example,  where  there   is  no  need  for  audio  and  video  at  all • Better  control  over  more  "types"  of  video • spherical  video,  stereo  video • augmented  reality • multicast,  simulcast,  and  pluggable  codecs • far-­end  camera  control • Improved  network  handling • better  network  interface  handoff  without  call  drop,  e.g.  mobile  to  WiFi and  vice  versa • support  for  QUIC  for  faster  overall  setup • fuller  support  for  ICE,  and  support  for  the  improved  version  of  ICE  being  developed  at  the  IETF.     This  may  even  allow  ICE  to  be  implemented  via  a  JS  library
  • 19. W3C  WebRTC JS  API  Test  Platform
  • 20. Satellite  view Specification TR Working  draft Stable  draft (Last  Call) Implementors  feedback  (CR) Standard  (Rec) Test  Suite WebRTC  1.0:  Real-­‐time  Communication  Between  Browsers 28-­‐Feb-­‐16 N/A Q2  2016? Y Media  Capture  and  Streams  (getUserMedia) 12-­‐Feb-­‐15 14-­‐Apr-­‐15 19-­‐May-­‐16 Y MediaStream  Recording 8-­‐Sep-­‐15 N/A N MediaStream  Image  Capture 27-­‐Jan-­‐15 N/A N Media  Capture  Depth  Stream  Extensions 26-­‐Feb-­‐16 N/A N Media  Capture  from  DOM  Elements 19-­‐Feb-­‐15 N/A N Audio  output  devices  API 10-­‐Feb-­‐15 N/A N Identifiers  for  WebRTC's  Statistics  API 6-­‐Feb-­‐15 N/A N Screen  Capture 10-­‐Feb-­‐15 N/A N
  • 21. APIs  View Media  Capture  and  Streams  (getUserMedia) isTested Size Coverage MediaStream Y 179 95% MediaStreamTrack Y 512 34% Media  stream  in  Media  element Y 77 100% Enumerating  Local  Media  Devices Y 159 45% GUM Y 281 68% Constraints N 136 0% Error  Handling N 100 0%
  • 22. Where  do  we  stand? (march  2016)
  • 23. WebRTC Testing  -­ DIY  1/3  – Small  Tools • Here  is  a  presentation  about  the  state  of  the  art  of  webrtc  testing: • http://www.slideshare.net/alexpiwi5/2016-­q1-­webrtc-­testing-­state-­of-­the-­art • you  have  a  couple  of  diagnostic  tools  available  that  can  be  used/reproduced  (always  good   to  have) • -­ https://www.netscan.co/ • -­ https://test.webrtc.org/ (source  code  is  here) • adapter.js has  its  own  testing  framework • -­ https://github.com/webrtc/adapter
  • 24. WebRTC Testing  – DIY  2/3  – Interop  &  system   • the  most  advanced  work,  sponsored  by  IMTC.  It  allows  to  tests  you  app,  running  with  your   infrastructure,  across  browsers,  browser  versions,  OS,  OS  versions,  .....  The  README  is   nicely  written. • -­ https://github.com/fippo/testbed
  • 25. WebRTC Testing  – DIY  3/3  -­ Native • For  native,  everybody  use  the  Google  native  stack  (webrtc.org)  with  its  binding,  and  the   testing  is  explained  in  detail  on  my  blog: • 1.  The  build  system  (here) • 2.  Automatic  build  with  CMake (here) • 3.  How  to  test  lib  webrtc  (here) • 4.  How  to  Package  /  Install  lib  webrtc  (here) • 5.  How  to  set  up  build  bots  for  WebRTC  (here) • 6.  How  to  patch  automatically  lib  webrtc  (here) • It  is  also  resumed  here  (some  slides  are  in  japanese): • http://www.slideshare.net/alexpiwi5/testing-­and-­packaging-­webrtc-­stack
  • 26. ?
  • 27. Save  The  Date:  Nov  21st Register  Now:  http://ccst.io/e/webrtcstandards10 Next  Session
  • 29. Thank  You Amir  Zmora TheNewDialTone Dan  Burnett StandardsPlay Alex  Gouaillard WebRTC by  Dr  Alex