DeveloperWeek 2015 - WebRTC - Where to start and how to scale

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DeveloperWeek 2015
WebRTC – Where to
start and how to scale
Vince Puglia
Developer Advocate

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Agenda
Looking into the WebRTC API
Adding WebRTC to your first app
Considerations for scaling WebRTC
Dealing (and not dealing) with WebRTC media

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Looking into the WebRTC API

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WebRTC is world-class VoIP engine for free
OPUS
On2
Global IP
Solutions
.org
Video Codec
Google acquires On2 for $133
million
VoIP Media Engine
Google acquires Global IP
Solutions for $68.2M
Audio codec
Opus - open, royalty-free, highly
versatile standardized audio codec
Open Source Community

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WebRTC is a Web and Internet Standard
draft-ietf-rtcweb-audio-05
draft-ietf-rtcweb-data-channel-07
draft-ietf-rtcweb-data-protocol-03
draft-ietf-rtcweb-jsep-06
draft-ietf-rtcweb-overview-09
draft-ietf-rtcweb-rtp-usage-12
draft-ietf-rtcweb-security-06
draft-ietf-rtcweb-security-arch-09
draft-ietf-rtcweb-stun-consent-freshness-01
draft-ietf-rtcweb-transports-03
draft-ietf-rtcweb-use-cases-and-requirements-14
WebRTC 1.0:
Real-time Communication Between Browsers
•Editors draft
•First Public Working Draft: 27 October 2011
•Feb 9 2012 Working Draft
•Aug 21 2012 Working Draft
•Last Call Working Draft: re-scheduled for Q1 2014
•Candidate Recommendation: initially scheduled for
Q4 2012, but delayed probably until Q2 2014
webrtc rtcweb

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WebRTC Technologies
NAT Traversal
• ICE
• STUN
• TURN
Audio Codecs
• Opus
• G.711
Video Codecs
• VP8
• H.264
Encryption
• DTLS
Port reduction
• RTCP Multiplex
• RTP Bundle
Signaling
• Session Description Protocol

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WebRTC is 3 JavaScript APIs (& Java)
getUserMedia
CreatePeer
Connection
Data
Channel
Creates a media connection
(RTP) between 2 peers
Grants access to local microphone,
camera, or screen
Allows the user to send arbitrary
data between peers

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WebRTC API Examples
http://dialogic.github.io/webrtc_samples/samples/web/content/getusermedi
a/gum/
http://dialogic.github.io/webrtc_samples/samples/web/content/peerconnecti
on/pc1/
http://dialogic.github.io/webrtc_samples/samples/web/content/datachannel/
index.html

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Adding WebRTC to your first app

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Many ways to work with WebRTC
Level of effort
Low-level
source
Frameworks,
Tools, and
SDKs
Cloud API’s
Amount of control
Website Plug-
ins & widgets

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Website Plug-ins & widgets – super simple but limited
Advantages
Easy for a web-admin to use
Often integrated with your CMS
Disadvantages
Only works on your website/CMS
Limited functionality
Limited control
Does not have your branding

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Cloud-API’s – easy but can be inflexible
Advantages
Rapid setup / Often very easy prototyping
Good platforms - have most of you need
Often easy to swap with other Cloud-API
vendors for basic calling
Disadvantages
Usage based cost
Vendor dependency
Feature evolution risk – may not always
have what you need
Platform lock-in on advanced features

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Low-level source - not for the faint of heart
Advantages
Full control
Many application-specific optimization
opportunities
Native mobile support
Ability to incorporate select components
Active community support
Disadvantages
Steep learning curve
Big on-going commitment to maintain
Standards not done & codebase continually in flux
Project was not designed for your application
Abiding by license terms

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Frameworks, Tools, and SDKs – many, many options
Advantages
Many options to choose from
Ability to choose best-of-breed
components
Optimized selections based on
application needs & team skills
Disadvantages
Figuring out which option is best
Interworking various toolsets
Inconsistent feature evolution
Potential platform lock-in

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Considerations for scaling
WebRTC

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What do you care more about?

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CPU & processing power is expensive
Client view Provider View

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What about Signaling?
Image source: Flickr user Tuomo Tammenpää – http://www.flickr.com/photos/misuser/5846728262/

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NAT - Deploying STUN & TURN servers
Network Address Translation (NAT)
Interactive Connectivity Establishment
(ICE)
Session Traversal Utilities for NAT (STUN)
Traversal Using Relay NAT (TURN)
Almost every WebRTC deployment should
have a TURN server
The open source ones are very good
rfc-5766-turn-server
Coturn

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Networking: Points of presence & latency
Inherent VoIP latency:
100-150 ms
÷ 2 for
one-way
AWS round-trip latencies
Transcoding latency:
0-200 ms

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Testing and reliability

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Dealing (and not dealing) with
WebRTC

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Traditional WebRTC is P2P
P2P
Minimal server cost
Lowest latency
Server
Alice Bob
media

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Easy & cheap approach to multi-party: Mesh
Full Mesh
Works for a few parties
Minimal server cost
Lowest latency

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Mesh does not scale for many video conferees
Full Mesh
Clients get overloaded
Encode costs more than decode
Limited uplink bandwidth
Inconsistent performance
across participants

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Challenge: Multi-party topology
Desktop SIP video client
Video stream
Android tablet with SIP client
WebRTC browser
iPad with SIP client
Android smartphone with SIP client

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PowerMedia XMS
Virtualized
Cloud-ready
Audio
Video
WebRTC
IMS & VoLTE Compliant
Superior Performance &
Scalability Robust Developer APIs Standardized Production
Interfaces
Software-based real-time media processing platform

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PowerMedia XMS
Multi-party mixing InterworkingTranscoding
Stream processing Person-to-machineRecording
PowerMedia™ XMS
Media Platform for Developers
RESTful
Java (JSR 309)
VXML
MSML
NETANN
Javascript
Powerful Media APIs

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Selective Forwarding Unit
SFU
Selective Forwarding Unit
(SFU) routing
Clients send one & receive
many
Client can instruct SFU which
streams to send
High throughput
Can be lots of downlink
bandwidth
Low latency

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Selective Forwarding Unit w/ Simulcast
SFU
High bitrate
Low bitrate
Selective Forwarding Unit
(SFU) with Simulcast
Clients send multiple streams
to SFU
one high-bit rate
one or more lower-bit
Client directs SFU which
streams to receive
Reduces bandwidth vs. SFU
Simulcast in WebRTC coming

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Multipoint Control Unit
Multipoint Control Unit (MCU)
Good at manipulating all media
Transcoding
Transizing
Transrating
Interworking
Server-side CPU intensive
Client can request different
conference mixes
Usually client friendly
Downsizing & mixing can reduce
bandwidth
Fewer client streams to process
MCU

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Multi-party video summary comparison
http://networkfuel.dialogic.com/webrtc-whitepaper

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WebRTC Servers Summary
Server Type Why/when you need it
Signaling Server Required to setup a WebRTC session
STUN Server Provides NAT traversal as part of ICE protocol
TURN Server
Relays media when direct connection cannot be
established
Media Server
Provides conferencing, recording, media
interworking, transcoding, stream processing
WebRTC Gateway
Interworks signaling and media with existing VoIP
networks

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PowerMedia XMS: Available Resources
PowerMedia XMS trial software download and 4-port evaluation license
(registration required)
PowerMedia XMS 2.3 documentation - Release Notes, Quick-start Guides
and API User Guides
PowerMedia XMS GitHub – JSR309, RESTFul (Java, C#, C++), VXML, MSML
and HTML/JS demo code
Dialogic YouTube Channel - PowerMedia XMS on-demand training and A
Journey to Media Server Enlightenment
PowerMedia XMS Community Forums
DEN Developer’s Blog

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My contact info
http://www.slideshare.net/Dialogic
linkedin.com/in/vincentpuglia
Vincent.Puglia@dialogic.com
@vfpuglia
12/10/2014

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