Wowza Ultra-Low Latency Streaming

Ultra-Low Latency
Streaming for OTTtv
CONFIDENTIAL© 2017 Wowza Media Systems, LLC. All rights reserved. Confidential & Proprietary.
Content DeliveryWorld Summit

WOWZA MAKES VIDEO UBIQUITOUS
© 2017 Wowza Media Systems, LLC. All rights reserved. Confidential & Proprietary.
20,000+ CUSTOMERS IN 170+ COUNTRIES FROM THE WORLD’S MOST
SOPHISTICATED MEDIA, TELCO, AND TECH COMPANIES TO SMALL BUSINESSES & DEVELOPERS
SERVICE PROVIDERS
ENTERPRISE &
GOVERNMENT
EDUCATION
Video comprised 64% of
global internet traffic in 2014
and will reach 80% by 2019
94% of young executives say
video breaks down language
barriers in the global market
Viewers of sports on
mobile devices` doubled
from 2011 to 2015
66% of university
respondents said streaming
is a top strategic initiative
~40% OF CDNS
WORLDWIDE RUN
WOWZA
70+ PERCENT OF
GARTNER’S ENTERPRISE
VIDEO QUADRANT
RUNS WOWZA
EVERY MINOR LEAGUE
BASEBALL GAME RUNS
ON WOWZA
85% OF THE TOP 25 U.S.
UNIVERSITIES, WITH 100%
OF IVY LEAGUE RUN
WOWZA
EVERYONE
Houses of
Worship
Transportation
Public Safety
Public Transit
Medical
Security
Armed Services
20,000 customers across all
industries build on Wowza for
video & audio
SPORTS

Powering Leading-Edge Broadcasting

THE SOFTWARE BEHIND THE STREAMING REVOLUTION
• END-TO-END VIDEO & AUDIO WORKFLOW
• DEEP DEVELOPER INTEGRATION VIA API, IDE, SDK
• ENORMOUS GLOBAL CUSTOMER BASE
• STRATEGIC TO ALL CLOUD PLATFORMS
450 Million
Live Viewing Hours per Month
20,000
Paying Customers
43 Million
Live Broadcasts per Year
(not including social media)
OVER 25%
OF WORLDWIDE LIVE STREAMING
WOWZA POWERS
345,000
Hours of Live
Transcoding per Day

WHAT IS LATENCY?
Latency is a time interval between the stimulation and response, or, from a
more general point of view, a time delay between the cause and the effect of
some physical change in the system being observed.
In terms of streaming, latency is the delay between the initial capture of the
video and the viewer.

TERMS
• Time to first frame - Time delay from when a person clicks the play button
and when video appears
• Broadcast Delay – The practice of intentionally delaying the broadcast of live
material to prevent profanity, bloopers, or violence
• Quality - Higher quality = higher resolution = more data to send
• Scale - How many inputs/participants, how far, how many viewers
• Bandwidth - How much traffic can the infrastructure ideally handle
• Throughput - How much traffic is the infrastructure really delivering
• Bitrate - How many bits of data are being processed over time

LATENCY ADDS UP
The video streaming industry solved for scale, but sacrificed latency. Now we need the best of both worlds.

Success Depends on a Usable Experience
An interactive and social gaming audience
Synchronized streams for a “live” experience
Betting with near-real time information
Requires imperceptible delay
where up to the second
information means winning
or losing.
Requires the coordination
between two delivery
systems so the audience
experience is seamless.
Requires near-real time
viewing so an audience can
actively participate or interact
with the gaming experience.

STREAMING LATENCY

WHY IS LATENCY A PROBLEM?
• Latency is not usually a problem – but people think it is.
• For many (or possibly most) live streams, latency does not matter.
• HTTP streaming intentionally introduces latency for improved reliability.
For some live streams, latency is critical.

WHAT FACTORS IMPACT LATENCY?
1. Quality
• Resolution
• Two-way
• Multi-thread
• Frame rate
• Smooth playback
• Chunk/Block Size
• Buffering
2. Scale
• Distance
• Participants
• Viewers
• Streams
• Complexity
• Variability
• Diverse endpoints
Low
Latency
Large
Scale
High
Quality
As either (or both) increase latency increases

HOW QUALITY INJECTS LATENCY?
• Higher quality = higher resolution = more data to send.
• Resolution increases the amount of data in any frame, block, chunk or time
slice. If we hold infrastructure constant as resolution goes up the window and
latency to deliver a segment increases.
• Trade-offs: Some mitigation techniques:
Increasing Throughput
Increasing throughput with
more bandwidth(network)
and bitrate(compute) = More
infrastructure, more
overhead, more cost
Reducing Buffer
Reducing the buffer requirement will
reduce latency but it also makes
network fluctuations more visible to
the viewer – e.g. a 1ms buffer
ideally means content can be
viewed as little as 1ms after capture
but a 0.1-second network
interruption will result in 100 buffer
segments being un-recoverable in a
live stream.
Reducing Framerate
Reducing the (frame rate) number of
frames per second reduces the
smoothness of the viewing
experience – e.g. 15 fps is generally
a good viewing experience for low
movement broadcasts like
presentation screen capture/sharing
or many church services, it will
produce a choppy experience for
viewing sports

HOW SCALE INJECTS LATENCY?
• Scale increases distances, complexity and variability.
• As distance, the number of streams, and viewers increase, network
imperfections, variability, and degradation increase and can amplify latency.
• Trade-offs: Some mitigation techniques:
Increasing Throughput
Increasing throughput with
more bandwidth(network)
and bitrate(compute) = More
infrastructure, more
complexity, more overhead,
more cost
Increasing Buffer
Increasing buffer size adds ability
for both encoders, transcoders and
players to deal with variability
smoothly but increasing buffer
inherently increases latency. By
definition, buffer size is the
minimum amount of data for which a
process can be initiated. If more
data must be accumulated, then
processing will take longer.
TCP vs UDP
Because it is a lighter weight protocol
without error checking, monitoring,
order of messaging and headers
roughly 1/3 the size of TCP, UDP is
inherently faster but there is no
guarantee data is received.

WHERE STREAMING LATENCY HAS BEEN
Windows Media
• MMS, RTSP, HTTP (using UDP or TCP)
• Encoder, Server, and Player buffer size
management
• “Low delay” audio codecs
• Fast Start, Advanced Fast Start, Fast
Recovery, Fast Reconnect, FEC
• 2-3 seconds (on a good network)
Real Time Streaming Protocol (RTSP)
• Developed and by RealNetworks
• Relies on RTP and RTCP
• Frames can be sent one at a time
in real time
• Can leverage UDP
• Potential latency could be as low
as ~125 ms with minimal buffering
Real Time Messaging Protocol (RTMP)
• Developed & open sourced by Adobe
• 1-3 seconds and sub-1 second
HTTP
• Apple HLS
• 30+ seconds on iOS
• Using 10 second chunk sizes
• MPEG-DASH
• 10-20 seconds (variable)

WHERE LATENCY IS GOING
WebRTC
• Designed for real-time audio, video
and data delivery over less-reliable
connections
• Leverages TCP or UDP
• Multiple protocols related to
RTSP/RTP
• 1 second or less (as low as 200 ms)
WebSocket
• Designed to provide a standardized,
two-way, reliable communications
channel between a browser and a
server
• Works with TCP
• Can be used with other streaming
protocols including RTMP, WebRTC,
Haivision SRT, Wowza WOWZ and
Aspera FASP
• As low as 200 ms

HTTP
• Below 4 seconds (using 1 second chunks) for DASH
• Optimizing DASH MPD attributes (availabilityStartTime and minBufferTime)
• Below 8 seconds (using 2 second chunks) for HLS (includes GOP
adjustments as well)
• CMAF + HTTP/1.1 (using HTTP Chunked Transfer Coding) enables video
transfer, decode, and display before the end of the chunk encoding
• Video encoding enhancements that do not impact decoding (H.264 GDR)

Quick UDP Internet Connections (QUIC)
• Uses UDP (with TCP fallback)
• Focuses on security and reliability
• Tries to be like TCP while reducing
connect and round trip times, packet
loss, congestion control, and more
using intelligent retransmissions and
storage and delivery of information
• Built with HTTP/2 in mind
Secure Reliable Transport (SRT)
• A video transport protocol that
enables the delivery of high-
quality and secure, low-latency
video across the public Internet
• Designed to deliver the best
quality live video over the worst
networks
• Accounts for packet loss, jitter,
and fluctuating bandwidth while
maximizing quality

Wowza Streaming Cloud Ultra Low Latency Service
• A cloud-based, highly scalable streaming service
• Using Wowza WOWZ + WebSocket in an origin-edge architecture
• Sub 2 seconds end-to-end (origin to player)
• Completely API-driven, no user-interface
Features include:
• Scaled Origin Network
• Scaled Edge network
• Playback components: SDK for browser and native apps on mobile
• Publishing and playback security
• Basic Analytics
• Recording

Mid tier EdgeOrigin
{ }Analytics
Live encoderIP camera
REST APIPull distribution Push distribution
Push distribution
WOWZA ULTRA LOW LATENCY SERVICE

REDUCING WORKFLOW COMPONENT LATENCY
Think about latency at every step
• Content creation (codec, bitrate, resolution)
• Streaming workflow and devices
• Buffer management (encoder, player)
• Network considerations
• How you optimize delivery (protocol, transport layer)
• How you reach your audience (size, location, and quality)
WebRTC and WebSocket can scale using traditional origin/mid-tier/edge
configurations (still requires “aware” clients)
CDNs are investigating options to enable low latency through their networks
• “Tuned” HTTP is a front runner

MEASURING LATENCY
• This is not easy (especially at scale)
• Latency is commonly measured in seconds (or milliseconds) though some
prefer frames
• A visual test is easy but requires accurate clocks (synchronized using
“reliable” NTP servers)
• For “traditional” streaming protocols, markers in the stream can be used

© 2017 Wowza Media Systems, LLC. All rights reserved. Confidential & Proprietary
Network Characteristic that KILL real time video
Ø Packet loss – packets being discarded by routers
Ø Jitter – packets arriving at different times than expected
Ø Delay – the time from sender to receiver
Ø Bandwidth – the fluctuating capacity between points
Overcoming Adverse Network Conditions

SRT Fundamentals
Source Destination
SRT Workflow
AWS
AWS

SRT Fundamentals
bps
t
Constant frame rate
Encoder Variable Bitrate
Unreliable “Dirty" Network
Packet Loss, Network Jitter,
Variable Delay, Bandwidth Fluctuation
t
Erratic bitrate and frame rate
Decoder’s Worst Nightmare!
SRT Fundamentals

bps
t
Constant frame rate
t
Reconstructed video frame rate
Much Happier Decoder J
Sender
Buffer
Receiver
Buffer
Encapsulation & Network Time Stamps
Feedback
SRT Fundamentals
SRT Fundamentals

bps
t
Constant frame rate
t
Reconstructed video frame rate
Much Happier Decoder Jü Video
ü Audio
ü Metadata
ü Packet acknowledgement – error recovery
ü Timing – low latency streaming
ü Network connection characteristics
ü Bi-directional
ü Firewall Friendly
SRT Fundamentals
SRT Fundamentals

SRT Fundamentals
SRT Demo

SRT in Wowza Streaming Engine
Ø Greater reliability of throughput and packet loss
Ø Improved bandwidth utilization with network bandwidth fluctuations & long-haul video transport
RESULTS - Germany to Australia, using a consumer DSL connection:
• 7.5x increase of throughput with SRT, compared to RTMP connections over TCP.
• RTMP: 800/10,000 kbits throughput
• SRT: and 6000/10,000 kbits throughput
Ø Independent security / encryption
Ø Minimized buffer 4x RTT (re-transmission time)
Ø Logic for packet loss recovery and jitter only requires 4 x the network latency to the server
Ø Wide adoption: free open source, known & trusted
SRT in Wowza Streaming Engine Wowza Streaming EngineSRT Fundamentals
Benefits of SRT | Results

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visit wowza.com/legal/trademarks for more information. Third-party trademarks are property of their respective owners;
their use does not imply endorsement of Wowza products or services by the trademark owner.
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Wowza Ultra-Low Latency Streaming

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