WebRTC Media Challenges

COMPANY CONFIDENTIAL © COPYRIGHT 2013 DIALOGIC INC. ALL RIGHTS RESERVED.
WebRTC Media
Challenges
April 2014

COMPANY CONFIDENTIAL © COPYRIGHT 2013 DIALOGIC INC. ALL RIGHTS RESERVED. 2
Agenda
WebRTC Introduction
Why Server-side media?
WebRTC Gateway Options
WebRTC in Contact Centers
Multi-party Confernecing
Security Considerations
PowerMedia XMS

WHAT IS WEBRTC?

What is WebRTC – world-class VoIP engine for free
OPUS
On2
Global IP
Solutions
.org
Video Codec
Google acquired video compression
company On2 in February 2010 for $133
million
VoIP Media Engine
Google acquires Global IP Solutions for
$68.2M for real-time video and audio
Audio codec
Opus is a totally open, royalty-free, highly versatile
audio codec standardized by the Internet
Engineering Task Force (IETF) as RFC 6716 which
incorporated technology from Skype's SILK codec
and Xiph.Org's CELT codec.
Open Source Community

What is WebRTC – pending 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

What is WebRTC - 3 Javascript APIs used in the browser
CreatePeer
Connection
GetUser
Media
Data
Channel
Creates a media connection
(RTP) between 2 peers
Grants access to local
microphone and/or camera
Allows the user to send arbitrary
data between peers

Why Enterprises & Contact Centers like WebRTC
User benefits
No context switching
No minute usage
Video option: high value
customers expect a
premium experience
Faster time to delight
Contact center benefits
Integrated in with lower-cost
web CRM systems
Faster resolution =
decreased agent time
Customer delight = return
customers

WebRTC browers are growing to be ubiquitous
1 Billion Endpoints!

Why Developers Like WebRTC
Hardcore
Telco Software
Infrastructure
(10k’s)
Source: TADS, Alan Quayle
Telecom App Developer Hierarchy
Robust & Reliable
Cheap & Flexible
Easy & Integrated
Expectations
IT/Web Programmers building on
FOSS, telecom app platforms and
telecom API’s
(1M’s)
‘Cut and Paste’ web developers using web-
scripting and graphical tools on app platforms
(10M’s)

Basic call scenario
1. Alice connects to website & downloads
Javascript application
2. Alice’s browser asks if it is ok to share media
3. Bob connects to website & downloads
Javascript application
4. Bob’s browser asks if it is ok to share media
5. Alice clicks “call”
6. The Server tells Alice’s browser how to call Bob
(signaling)
7. Alice’s browser calls Bob
8. Bob accepts
9. Media is routed direcly between Alice & Bob
Server
Alice Bob
media

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

WHY SERVER-SIDE MEDIA?

Typical WebRTC Models – P2P media
Server
client client
media
triangle
Server
client client
signaling signaling
media
trapazoid
Server
signaling
Lowest server & network cost

P2P with TURN
Server
client client
triangle
Server
client client
signaling signaling
trapezoid
Server
signaling
TURN
Server
TURN
Server
more server & network cost

WebRTC with Server-side Media
client client
triangle
Server
client
signaling media
trapazoid
signaling
Server
client
signaling media
more server & network cost
Server

Design Trade-offs – Cost vs. Performance vs. Location
Application
Performance
Server
Performance
Client
Performance
Higher
infrastructure
costs
Battery life
Speed
Quality
expectations

When does server-side media make sense?
When bandwidth
is limited
When the user is
paying per GB
When power is
limited
When there is
regulation

PowerMedia™ XMS
Multi-party mixing InterworkingTranscoding
Stream processing Person-to-machineRecording
WebRTC Media Processing Use Cases

WEBRTC GATEWAY MODELS

“Legacy” Telephony Interworking Functions
HTTP-to-SIP
(H2S)
Media GW
Transcoder
NetworkWebRTC Users Phone User
SRTP-DTLS
UDP/TCP
RTP/SRTP-SDES
UDP
OPUS/VP8 G.7XX/H.264

WebRTC Gateways Come in Many Packages
HTTP-to-SIP
(H2S)
Media GW
Transcoder
HTTP-to-SIP
(H2S)
Media GW
Transcoder
HTTP-to-SIP
(H2S)
Media GW
Transcoder
HTTP-to-SIP
(H2S)
Media GW
Transcoder

Will One H2S Gateway be Sufficient?
Retail Click-to-Call
Enterprise Click-to-Call
Residential Web Phone
RCS Web Client
Developer API exposure
Remote Agent
Remote Worker
HTTP-to-SIP
(H2S)
IMS/RCS
RFC3261 SIP
Enterprise SIP
Application specific APIs work best SIP is not really that standard

Media Requirements More Uniform than H2S
H2S
WebRTC
Media GW
Transcoder
NetworkWebRTC Users SIP/IMS Networks
Web Signaling
SRTP-DTLS
UDP/TCP
RTP/SRTP-SDES
UDP
OPUS/VP8 G.7XX/H.264
H2S
H2S
SIP
H2S
Control Interfaces

3GPP TR23.701 WebRTC Gateway Architecture for IMS
H2S
WebRTC
Media GW
Transcoder
Javascript
Client
Webserver

WEBRTC IN CONTACT CENTERS

State-of-the-Art in Conferencing & Collaboration
Voice, video, text
2 or multi-party
Web or mobile app
PSTN calling
Share screen, desktop, docs
Webcasting
Real-time video augmentation
APIs for 3rd parties
26

State-of-the-Antiquated?
99.999% reliability
Universal interoperability
IVR with TextSpeech
Regulatory compliant
Supervisor modes
Recording & playback
API’s for systems integration
27

Contact Center Integration
28
click
Application
Server
Web Apps
SBC Media Server
Agent
Media
Interface
Media Control
SIP
SIP
RTP RTP
HTTP
Location
• Traditional click-to-callback
accelerated by WebRTC
Web-based
Application
integration
PSTN
Contact Center

29
Web-based
callers
• Eliminate PSTN charges
• Faster time-to-resolution
• Improved customer experience
Web Apps
Agent
Media Control
WebRTC Signaling
SIP
SRTP RTP
HTTP
WebRTC Browsers
Application
Server
Media
InterfaceHTTP
Contact Center
Media Server

30
Web-based
agents
• WebRTC reduce cost of
supporting remote agents
• Optional video integration
Web Apps
Agent
Media Control
WebRTC Signaling
HTTP
Location
WebRTC Browser
Application
Server
Media
Interface
SBC Media Server
SIP
RTPPSTN
Contact Center

Case Study – the Airline
31

Video-enabling a Contact Center
Web App
Server
Media
Control
HTTP/
REST
Media
Interface
SBC Media Server
RTP
Contact Center
SRTP
PSTN
PSTN Customers
Agents
ACD
Kiosk Customer
RTP
SIP
HTTP/
REST

Case Study – the Airline
33
Lessons Learned:
• Applications are viable today when you have
end-to-end control
• Leveraging existing infrastructure shortens
time-to-market
• Assume you will have poor connectivity
• Be careful with variable cloud latencies

MULTI-PARTY CONFERENCING

Challenge: Multi-party topology
35
Desktop SIP video client
Video stream
Android tablet with SIP client
WebRTC browser
iPad with SIP client
Android smartphone with SIP client

Peer-to-Peer Mesh with a Few Parties
36
• Works for a few
parties
• No server cost
• Lowest latency

Peer-to-Peer Mesh with Many Parties
37
• Doesn’t scale for
many parties
• Heavy
processing
burden on all
parties
• Massive
bandwidth

Centralized Routing
38
• Some server cost
• Lots of bandwidth
• Lots of client
processing
Video Router

Centralized Routing with SVC
39
• Some server cost
• Reduced bandwidth
• Reduced client
processing
• Requires specialized
clients
Video Router

Streamline with Media Server
40
Multipoint Control
Unit (MCU)
• Provides scalability
• Improves mobile
performance
• Dynamic back-end
control
• Least bandwidth &
client processing
MCU

Challenge: mixed capabilities
41
Desktop SIP video client
Android tablet with SIP client
WebRTC browser
iPad with SIP client
Android smartphone with SIP client
1920x1080
30 FPS
H.264
640x480
15 FPS
H.261
352X288
30 FPS
VP8
1280x720
25 FPS
VP8
1280x720
30 FPS
VP8
1280x720
30 FPS
H.264

0
240
480
720
960
1200
1440
1680
1920
2160
0 320 640 960 1280 1600 1920 2240 2560 2880 3200 3520 3840
Video Resolutions will Continue to Expand
720P
.9MP
1080P
2.1MP
VGA/.3MP
4K
8.3MP
WebRTC
Resolutions
today

SECURITY CONSIDERATIONS

Scope All Traffic APIs VoIP Traffic
Identifier Address:Port API Key Phone #, URI
Controls Bandwidth, Address,
Ports
API Calls VoIP Messages,
Bandwidth, Ports
Topology
Hiding
No N/A Yes
WebRTC Security - Looks More like Web than Telco
API
Management
Session Border
Control
Firewalls

Basic, Telco-Centric Model – Trust the SBC, Not the Web
Web
server
SBC
SIP/IMS
Network
SIP
Javascript
App
H2S
WebMG
SIP
RTP
Firewall

Mature, Web-Centric Model – Re-use Web Security
Web
server
SIP/IMS
Network
SIP
WebMG
Firewall
APIManager
API Calls
H2S
Server
Identity
Server
App
Server
App
Server
WSS
server
STUN/
TURN
Internal
REST APIs

POWERMEDIA XMS

PowerMedia XMS
Virtualized
Cloud-ready
Audio
Video
WebRTC
Robust
Developer APIs
Standardized
Production
Interfaces
real-time media processing platform

SIP Web Java Speech
Audio Video WebRTC CPA
PowerMedia™ XMS
Media Platform for Developers & IT
RESTful
Java (JSR.309)
VXML
MSML
NETANN
MRCPv2
Javascript
PowerfulMediaAPIs
PowerMedia XMS
Diverse Developer Audiences
Extensive Capabilities

PowerMedia™ XMS
Multi-party mixing InterworkingTranscoding
Stream processing Person-to-machineRecording
PowerMedia XMS – Many Use Cases
RESTful
Java (JSR.309)
VXML
MSML
NETANN
MRCPv2
Javascript
Powerful Media APIs

PowerMedia XMS Media Resource Function (MRF)
IMS compliant MRF interfaces
XMS controlled via SIP from AS
MSML or VXML embedded inside SIP signaling
directed to XMS
All core SIP traffic brokered by S/I-CSCF
Offloads media processing from AS
Multi-party conferencing
Announcements & Prompts
IVR
Ring back tones
Voicemail
Transcoding
Future ready - comprehensive video &
WebRTC support
Starting point for NFV – 100% virtualized
software
Hypervisors: VMWare, KVM, XEN, OVM
TAS
MGCF/
MGW
SBC/
P-CSCF
CSCF
PowerMedia XMS MRF
SIP
RTP
Mr
Mb Mb
ISC
Mg/MiMw

PowerMedia XMS WebRTC Media Gateway
Converts WebRTC media to standard
VoIP media
Encryption interworking
Media plan interworking
NAT traversal negotiation
Rapidly integrates with applications
Web apps - RESTful & Javascript API
Java apps – JSR.309
SIP – MSML, NetAnn
Scripting interface – VXML
Proven integration with many ecosystem
partners
Streamlined media plane
Less infrastructure -reduced CAPEX & OPEX
Offloads media functions from applications'
SIP/IMS Networks
H2S H2S
H2S H2S
Media Control
Interfaces
PowerMedia XMS
WebRTC Media Gateway
Developer & Partner Applications
with Signaling

z
PowerMedia XMS Software Transcoder
Virtualized software-based transcoding
Leverage cloud assets instead of expensive DSPs
Inline transcoder
Acts as a B2BUA
No signaling control required
Transcoding gateway
Use media server control interfaces
RESTful API
RFC4117*
Comprehensive & rapid codec support
Audio - G.711u/a, G.723, G.726, G.729a, G.729b,
AMR-NB, iLBC*, Wideband: G.722, AMR-WB**
(G.722.2), Opus
Video - VP8, H.264, MPEG 4, H.263, H.263+,
H.263++, VP9*
MGW/SB
C
SIP/IMS
Core
MGW/
SBC
Codec A Codec B
Inline Transcoder
PowerMedia
XMS
MGW/SB
C
SIP/IMS
Core
MGW/
SBC
Codec A Codec B
Transcoding Gateway
PowerMedia XMS
Signaling
Media
** - Using the AMR-WB resource in connection with one or more Dialogic products mentioned herein does not grant the
right to practice the AMR-WB standard. To seek a patent license agreement to practice the standard, contact the VoiceAge
Corporation at http://www.voiceage.com/licensing.php.

• Audio Codecs
• G.711u/a, G.723, G.726, G.729a, G.729b, AMR-NB, iLBC*
• Wideband: G.722, AMR-WB** (G.722.2), Opus
• Video Codecs and Processing
• VP8, H.264, MPEG 4, H.263, H.263+, H.263++
• HD720p, VGA, QVGA,4CIF, CIF, QCIF
• Frame rates: Up to 30 FPS
• Bit rates: Up to 2Mbps
• MSML (SIP based)
• RESTful API (http based)
• JSR309 (Java based)
• VXML (script based)
• NETANN (SIP)
State of the art rich media processing
• Streaming and Recording
• Advanced Conferencing
• Transcoding, Trans-rating, Trans-sizing
• Video text and image overlays, scrolling
• Tone Generation / Detection
• Call Progress Analysis
• Positive Voice Detection (PVD)
• Positive Answering Machine Detection (PAMD)
• SIP (RFC 3261)
• RTP, RTCP, RTCP-XR, RTCP-HR, SRTP
• RTSP (client support, streaming media)
• MRCPv2 (ASR/TTS server support)
• ICE Lite / STUN* / TURN*
• IPv4, IPv6* ** - Using the AMR-WB resource in connection with one or more Dialogic products mentioned herein does not grant the right to practice the AMR-WB standard. To seek a patent license
agreement to practice the standard, contact the VoiceAge Corporation at http://www.voiceage.com/licensing.php.*planned for 2014
SIP / WebRTC / RTP Specifications
Audio / Video Media Functions Audio / Video Media Functions
Media Control Protocols

WebRTC Media Challenges

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