2017/eur/pdf/BRKCOL 2616 PDF

QoS Strategies and
Smart Media Techniques for

Collaboration Deployments
Stefano Giorcelli - Technical Leader
Glen Lavers - Technical Marketing Engineer
BRKCOL-2616
Abstract
In this session we will discuss the network infrastructure and application requirements to ensure a high-quality experience for
interactive voice and video collaboration sessions. Quality of Service (QoS) tools in the network are a key component to providing a
solid and consistent collaboration experience, but they are only one of the aspects needed for an end-to-end solution. This 2-hour
session will cover topics such as "smart" media techniques like media resilience and rate adaptation, showing how they help mitigate
the impact of network impairments such as packet loss and delay; media identification and classification (trust boundaries, DSCP
marking) queuing and scheduling (CBWFQ, WRED) provisioning and resource control (planning, bandwidth allocation) looking at h ow
these functions are applied to all types of Collaboration endpoints (hardware, software, mobile) in different parts of the ne twork (LAN,
WLAN and WAN) and provide configuration examples to illustrate the concepts. This session will provide design best practices for
common deployment scenarios and introduce updated collaboration components such as Expressway Edge QoS and Cisco Meeting
Server QoS integration.
This 2-hour session is divided in two main sections:
In the Collaboration Media section we will discuss video and voice traffic characteristics and requirements and introduce smart
media techniques such as media resilience and rate adaptation, showing how they help mitigate the impact of network impairments
such as packet loss and delay.
In the QoS Architecture and Design section we will describe the three main QoS functions in the context of Collaboration:
media identification and classification (trust boundaries, DSCP marking)
queuing and scheduling (CBWFQ, WRED)
provisioning and resource control (planning, bandwidth allocation)
We will look at how these functions are applied to all types of Collaboration endpoints (hardware, software, mobile) in different parts of
the network (LAN, WLAN and WAN) and provide configuration examples to illustrate the concepts. We will also take into account the
smart media techniques introduced in the previous section to introduce new queuing and provisioning models that simplify operations
and maximize resource usage. We will also provide design best practices for common deployment scenarios.
BRKCOL-2616 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public 3
Agenda
Introduction
Collaboration Media
QoS Architecture
Design and Deployment
Introduction
Evolution of Collaboration Landscape
On-premise
UC Services Call
Fixed, hardware endpoints Control
Managed networks
Central
Site
Managed
WAN Internet
MPLS
VPN
Remote Sites
HW Endpoints
On-premise Spark
UC Services Call
Managed networks
Central
Site Cloud Services
Managed
WAN Internet
MPLS
VPN
Remote Sites
HW Endpoints
On-premise Spark
UC Services Call
Managed networks
Central
Site Cloud Services
Managed
WAN Internet
MPLS DMVPN
VPN
Remote Sites
HW Endpoints
On-premise Spark
UC Services Call
Managed networks
Central
Site Cloud Services
Managed
WAN Internet
MPLS DMVPN
VPN
Remote Sites
HW Endpoints
On-premise Spark
UC Services Call
Managed networks
Central
Site Cloud Services
Mobile, software endpoints

Managed
Unmanaged networks WAN Internet
MPLS DMVPN
VPN
Remote Sites
HW Endpoints
Software
Clients
Managed vs. Unmanaged Networks
Where do your media packets go?
On-premise Call Control

UC Services
Central
Site
B2B
B2C
Managed
WAN Internet
MPLS DMVPN
VPN
Remote Sites Home/Mobile Users

On-premise Call Control

UC Services
Central
Site
B2B
B2C
Managed
WAN Internet
MPLS DMVPN
VPN

On-premise Call Control Spark

UC Services
Central
Site
Cloud Services
B2B
B2C
Managed
WAN Internet
MPLS DMVPN
VPN

On-premise Call Control Spark

How do you preserve
UC Services
user experience when
Central
Site
media traverses the
Cloud Services
Internet?
B2B
QoS-
capable
B2C
Managed
WAN Internet
MPLS DMVPN
VPN

Evolution of Collaboration Media Streams
Temporal
layers
Adaptive
video bitrate
Multi-device Simulcast
sessions multistreaming
Multipoint
Bridge Multipoint
Bridge
Collaboration
data
Active cascading
Deployment Challenges
Voice has been deployed to very large scale onto integrated IP networks
Video so far has seen smaller-sized deployments:
Overlay networks or islands
Smaller numbers of endpoints (e.g., room systems)
Deploying pervasive video is challenging
Multi-media, mobile endpoints
Varying bandwidth needs
Hard to plan/provision
Cloud services, B2B/B2C communications and remote/mobile access make it
even harder to control the user experience
Antiquated QoS Approaches
Classification and Scheduling Considerations
Same DSCP for audio and video
streams of a video call
Audio of
voice call EF Policer PQ During congestion, audio and video
streams are equally impacted
Audio of
Telepresence CS4
Video of
CBWFQ Different DSCPs for audio
WAN Link
Telepresence CS4 streams in video calls vs. voice
Audio of
AF41
calls
Desktop video
CBWFQ Media stream identification difficult
Video of
Desktop video AF41 for multi-media mobile clients
other queues Different queues for immersive/
room system video and desktop
video
Complex provisioning, sub-optimal
bandwidth usage
Our Strategy
Commonly Deployed New Design &
Smart Media Techniques
LTRF1
QoS Tools Deployment Guidelines
LTRF1
P1 P3
P2 P4
P5
P1
P5 EF Audio
P2 P4
... ... ... ... Queue
Encoder Decoder
? EF
AF42
WAN Link
OOS (P4) ACK LTRF1
AF42 Video
Encoder Decoder
Queue
AF41
LTRF 0111010001 R1 FEC

AF41
1000011001
Repair-P R1 0001100
1001000100
0011001011
1011110
FEC
1110010101
... ... R2 1011010010

1010010
R2
Leverage media resilience and

Consolidate mechanisms to
rate adaptation to enable
Use media resilience to
pervasive video deployments
reduce impact of packet loss identify Collaboration media
through:
Apply rate adaptation to Evolve classification and simplified provisioning
reduce network congestion scheduling recommendations
optimized bandwidth
utilization
Load Threshold
Receiver
Sender
Load Threshold
Receiver
Sender
1
Endpoints identify their media
streams to the network
Load Threshold
2
Network classifies traffic
Receiver
Sender and applies differential
treatment
1
Endpoints identify their media
Load Threshold
2
Receiver
treatment 3
Endpoints provide ongoing
1 media feedback (delay, loss, ...)
Endpoints identify their media Receiver Feedback
4
Network indicates congestion
by queuing and/or discarding
media packets
Load Threshold
2
Receiver
treatment 3
5
4 Media resilience techniques
Network indicates congestion preserve user experience
by queuing and/or discarding during congestion
media packets
Load Threshold
2
Receiver
treatment 3
5
4 Media resilience techniques
Network indicates congestion preserve user experience
by queuing and/or discarding during congestion
media packets
6
Endpoints dynamically
adapt media streams
Load Threshold
2
Receiver
treatment 3
Agenda of This Session
Technology Trends and Current Issues
Introduction Strategy Overview
Video Traffic
Collaboration Media Smart Media Techniques
(Media Resilience and Rate Adaptation)
Identification and Classification

QoS Architecture Queuing and Scheduling
Provisioning and Resource Control
Mobile and Remote Access

Design and B2B, B2C and Hybrid/Cloud Services
Deployment Example Enterprise Deployment
Collaboration Media
Collaboration Media Agenda
Video Traffic (i.e., a network administrators concerns with interactive media/video)
Audio vs Video
Bandwidth, Packet Loss, Delay

Media Resilience (i.e., how video can be easier on the network)
Rate Adaptation
Video Traffic: Requirements and Profiles
AUDIO Bandwidth
Bandwidth:
Constant bitrate (smooth)
Small footprint
Narrow operational range (1:6)
Time
G.729 (24 kbps) Loss-sensitive
AAC-LD (160 kbps)
Delay-sensitive
Operational bandwidth
VIDEO Bandwidth Bandwidth:

Variable bitrate (bursty)
Medium/large footprint
Wide operational range (1:40)
Time
Loss-sensitive
240p15 (150 kbps) 1080p60 (6 Mbps)
Delay-sensitive
Operational bandwidth
2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Video Traffic
Elasticity of Voice and Video
user
experience
best
better voice video
good
acceptable
unusable
bitrate (kbps)
elasticity
1 I-Frame
Video Traffic Intra-coded picture
Video Encoding Basics Entire picture encoded
1 as a static image
No reference to other
frames
2 P-Frame
Predicted picture
Based on a previously
2 encoded frame ( 1 )
Only the differences
from that frame are
encoded
3 P-Frame
Predicted picture
3 Reference for
prediction can be
another P-Frame ( 2 )
Video Traffic
Audio vs. Video Packet Distribution
Audio Packets Video Packets
P-Frame I-Frame P-Frame
1400 1400
1000 1000
Bytes
600 600
Audio
Samples
200 200
20 ms Time 33 ms
Video Traffic
Bandwidth Usage: High-definition Video Call
3500
I-Frames
3000
2500
Bandwidth (kbps)
2000
1500
1000
500
0
Time (s)
HD video call, 720p30 @ 1920 kbps (1792 kbps video + 128 kbps audio)
Video bandwidth shown (including L3 overhead)
Video Traffic
Impact of Packet Loss on a Video Stream Frozen video
I1 Artifacts I1
Video
Pulsing
P1 P3 P1
P2 P4 P5 P2 P4 P5
... ... ... ...
Encoder Decoder
?
P5 I1 I1 I1 P4 P3 P2 P1
Out of Sync (OOS)
Loss of a P-frame triggers request for a new I-frame

Encoding and transmitting large a I-frame takes time
If any of the I-frame packets get lost, the process needs to restart
I-frame creates burst that risks exacerbating network congestion (more packet loss!)
Flickering/pulsing of video when new I-frame arrives

Video freeze or artifacts when multiple packets are lost
Video Traffic
Delay in Home/Consumer Access Scenarios: Bufferbloat
Delay seen by 1 audio stream

sharing a queue with 1 TCP flow
100% link utilization
Queue
Queuing delay: ~100-500ms ! Never
Emptied
http://en.wikipedia.org/wiki/Bufferbloat
Delay seen by 1 audio stream

sharing a queue with 10 TCP flows
100% link utilization Queue
Never
Queuing delay: ~400-500ms ! Emptied
Collaboration Media Agenda
Video Traffic (i.e., a network administrators concerns with interactive media/video)
Audio vs Video
Bandwidth, Packet Loss, Delay

Media Resilience (i.e., how video can be easier on the network)
Rate Adaptation
Goals and Solutions
Goals Mechanisms
Make network congestion Media Resilience
less likely to occur Encoder Pacing
LTRF with Repair

Recover more efficiently
from packet loss FEC
Optimize use of available

Rate Adaptation
network resources
Media Resilience
Encoder Pacing
1400
P-Frame I-Frame P-Frame P-Frame Each frame must be
packetized onto the wire
1000
in 33 ms
Bytes
600 Endpoint packet

scheduler disperses
200 packets as evenly as
possible
33 ms Time
1400
P-Frame I-Frame P-Frame Large I-frames may need
to be spread over 2 or
1000
3 frame intervals
Bytes
Encoder may then skip 1-2
600
frames to stay within bitrate
budget
200
33 ms Time 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
Media Resilience
Long Term Reference Frame (LTRF) with Repair
Long-Term Reference Frame
LTRF1 (not actually sent on the wire) LTRF1
P5
Repair P-Frame P5
P1 P3 Built from last synced LTRF P1
P2 P4 P2 P4
... ... ... ...
Encoder Decoder
?
P5 P4 P3 P2 P1
OOS
ACK (P4)
LTRF1
Keep encoder and decoder in sync with active feedback messages

Encoder instructs decoder to store raw frames at specific sync points as Long-
Term Reference Frames (part of H.264 standard)
Decoder uses back channel (i.e. RTCP) to acknowledge LTRFs
When a frame is lost, encoder creates a Repair P-frame based on the last
synchronised LTRF instead of generating a new I-frame
Media Resilience
Forward Error Correction (FEC)
Encoder Decoder
LTRF Binary XOR R1 FEC

011101000
110000110
Repair-P R1 010001100 100100010
000110010
FEC Binary
111011110
XOR
... ... R2
111001010
110110100
101010010
R2
Allows decoder to recover from limited amount of packet loss without losing
synchronization
Can be applied at different levels (x FEC packets every N data packets) to protect
important frames in lossy environments
Correction code can be basic (binary XOR) or more advanced (Reed-Solomon)
Trade-off is bandwidth increasebest suited for non-bursty loss
Rate Adaptation
Key Idea
Video Packet
Bitrate Loss
SLOW
DOWN
Sender t1 t2 t1 t2 Receiver
RR 1
2
3
RTCP
Receiver observes delay and packet loss over periods of time and signals back
using RTCP Receiver Reports (RR)
Reports cause the sender to adjust bitrate so as to adapt to network conditions
(downspeeding, upspeeding)
Two approaches possible:
Sender-initiated adjustment based on RTCP Receiver Reports
Receiver-initiated adjustment via call signaling (H.323 flow control, TMBRR,
SIP Re-invite) or explicit request in RTCP message
Key Takeaways
Burstiness of traffic and mobility of the endpoints make deterministic
provisioning for interactive video difficult for network administrators
Media resilience mechanisms help mitigate impact of video traffic on the

network and impact of network impairments on video
Dynamic rate adaptation creates an opportunity for more flexible provisioning

models for interactive video in Enterprise networks
Media resilience and rate adaptation also help preserve user experience when
video traffic traverses the Internet or non-QoS-enabled networks
QoS Architecture:
IDENTIFICATION & CLASSIFICATION

QUEUING & SCHEDULING
PROVISIONING & RESOURCE
CONTROL
QoS Architecture
Network
Building an End-to-End Strategy Management
EF
WAN
AF41
BE
Provision Monitor
Identify Classify Schedule Resource Troubleshoot
Control Optimize
QoS Trust and Map identified Low Latency Bandwidth Prime Collaboration
Collaboration Traffic traffic to its correct Queuing (LLQ) Provisioning Monitoring Assurance
Identification QoS marking Priority Queuing and Diagnostics
Admission Control
(PQ)
QoS marking: Monitors Voice, Video,
Class Based
COS, DSCP Inventory and Fault
Weighted Fair
Queue (CBWFQ) Management
Weighted Random BRKCOL-2017

Simplify Collaboration
Early Detection
Deployments with Prime
(WRED) Collaboration
Wednesday, Feb 22, 4:30
p.m. - 6:00 p.m.
Identification and Classification Agenda
QoS Identification and Identify Classify Schedule Provision Monitor
Classification
DSCP Class DSCP ToS Prec.
Classification: DSCP Classes none
CS1
0
8
0
1
AF11 10 1
EF: Expedited Forwarding (PQ) AF12 12 1
Used for voice media AF13 14 1
CS2 16 2
AF: Assured Forwarding (CWBFQ) AF21 18 2
AF22 20 2
Used for video media AF23 22 2
CS3 24 3
CS: Class Selector AF31 26 3
Used for signaling AF32 28 3
AF33 30 3
CS4 32 4
AF41 34 4
AF42 36 4
AF43 38 4
CS5 40 5
EF 46 5
CS6 48 6
CS7 56 7
Campus QoS Design Considerations
Trust Boundaries
Trust Boundary
Access-Edge Switches WAN-Edge Router
Conditional
Trust Classification,
Examples: IP Phone + PC
Marking and
Queuing
Trusted IP WAN
Examples: Collab Server, MCU, TS

Trust Boundary
Untrusted
Examples: PC, Mac, Handheld
QoS Trust (Conditional vs Trusted vs Untrusted)
Drawbacks
Requires multiple access policies to implement different types of trust
(conditional vs full vs untrusted)
Trust Configuration Differs from Platform to Platform
Some Platforms trust by default others do not
ACL access policies for untrusted devices are required on every access port
Solution
SIMPLIFY: A single access policy for ALL endpoints
Trust and Enforcement
Devices and Places in the Network Multiple Access Policies
NOT IDEAL
Conditionally-Trusted Endpoints
Access Distribution Core WAN Edge
Immersive Endpoints
CDP Support required

IOS SIP Gateway
IP Phones, DX, TC, TX, CTS Mac/PC
Trusted Endpoints / Devices
SRST Gateway
WiFi AP
PC/GPO MAC Handheld
Unified BE
Untrusted Endpoints / Devices

Trusted
Trusted Devices
Untrusted
Conditionally
MAC Handheld
PC Trusted
Instant Messaging TelePresence
Unified CM
Unity
And Presence Servers IOS Router
Expressway
Connection Remarking
Trust and Enforcement Single Access Policy
Devices and Places in the Network IDEAL
SX, MX and IX Series Endpoints Access Distribution Core WAN Edge
Immersive and Room

System endpoints IOS SIP Gateway
SRST Gateway
IP Phones Jabber and DX Series

Endpoints
Unified BE
IP Phones
Trusted
Jabber Trusted Devices
Untrusted
Smart Desktop Conditionally
Trusted
Instant Messaging TelePresence
Unified CM And Presence Servers IOS Router
Mac/PC Unity Expressway
Trust and Enforcement 3 Single Access Policy
Deploy an Access
Devices and Places
1
in the Network Layer Policy
IDEAL
Configure endpoints 4
Configure Endpoint
SX, MX and IX Series Endpoints with DSCP and
QoS and Port Ranges
Access Distribution DeployCore
a WAN Edge WAN Edge
TCP/UDP Port Ingress Remarking Policy
Immersive and Room

System endpoints IOS SIP Gateway
SRST Gateway
IP Phones Jabber and DX Series

Endpoints
Unified BE
5
Deploy a WAN Edge
IP Phones Egress Queuing Policy
2 Trusted
Jabber Configure Trusted Devices
Application / Server Untrusted
Smart Desktop QoS Conditionally
Trusted
Instant Messaging Cisco Meeting
Unified CM And Presence Server IOS Router
Mac/PC Unity Expressway
Mapping Identifiable Media and Signaling Streams
1. Configure Endpoints
2. Configure Application / Servers
3. Deploy an Access Layer Policy: Endpoint identification and classification
4. Deploy a WAN Edge Ingress Remarking Policy (Catch All)
5. Deploy a WAN Edge Egress Queuing Policy
Unified CM QoS Classification
Audio Portion of Video/TelePresence calls (*Only supported on specific phone
software releases) Default Values
Unified CM System QoS Values and CAC Pool Associations
Service Parameter Name Media Stream Type DSCP Value PHB Value CAC Pool
DSCP for Audio Calls Audio Only 46 EF Voice
*DSCP forVideo
Audio Portion of
Calls
Audio of Video 34 AF41 Video
DSCP for Video Calls Video of Video 34 AF41 Video
*DSCPTelePresence
for Audio Portion of
Calls
Audio of TP 32 CS4 Video
DSCP for TelePresence Calls Video of TP 32 CS4 Video
Unified CM QoS Classification
Audio Portion of Video/TelePresence calls (*Only supported on specific phone
software releases). Recommended Values for CUCM 11.x deployment
Unified CM System QoS Values and CAC Pool Associations
Service Parameter Name Media Stream Type DSCP Value PHB Value CAC Pool
DSCP for Audio Calls Audio Only 46 EF Voice
*DSCP forVideo
Audio Portion of
Calls
Audio of Video 46 EF Voice
DSCP for Video Calls Video of Video 34 AF41 Video
*DSCPTelePresence
for Audio Portion of
Calls
Audio of TP 46 EF Voice
DSCP for TelePresence Calls Video of TP 34 AF41 Video
Custom QoS settings for SIP Devices New
(CUCM 11.0)
Clusterwide Parameters (System QoS)
DSCP for Audio Calls
DSCP for Video Calls
DSCP for Audio Portion of Video Calls TelePresence
DSCP for TelePresence Calls Endpoints
DSCP for Audio Portion of TelePresence Calls
Media Port Ranges (Common Port Range for Audio and Video)
Media Port Ranges (Separate Port Range for Audio and Video*) TelePresence Applicable DSCP settings:

* Only Supported on Jabber Endpoints DSCP for TelePresence Calls
DSCP for Audio Portion of TelePresence Calls
QoS Registration
Service SIP Profile Device
Parameters Config File
UC Video
Unified Endpoints
CM
UC Video Applicable DSCP settings:

DSCP for Video Calls
DSCP for Audio Portion of Video Calls Port ranges
Media Port Ranges (Separate Port Range for Audio and Video*)
New in 11.0
TelePresence Endpoints in Unified CM
TelePresence endpoints are identified as immersive video endpoints
Fixed setting (Not Configurable)
Check Devices for Capability:
Cisco Unified Reporting Tool > Immersive Video Support for

TelePresence Devices
New
(CUCM 11.0)
Custom QoS Settings For SIP Devices
SIP Profile Settings
New Separation of Audio and Video UDP Port Ranges
New DSCP parameters
Benefits
Granular configuration of SIP Endpoint groupings for both Trusted (DSCP
honored) and Untrusted (UDP Port Remarking) Endpoints.
New
Custom QoS settings for SIP Devices (CUCM 11.0)
SIP Profile 2
SIP Profile 1
TelePresence Desktop Jabber

Endpoints Video Clients
Endpoints
Prioritized Video: AF41 Opportunistic Video: AF42

New
Custom QoS settings for SIP Devices (CUCM 11.0)
SIP Profile 2
SIP Profile 1
TelePresence Desktop Jabber

Endpoints Video Clients
Endpoints
Prioritized Video: AF41 Opportunistic Video: AF42

Custom QoS Settings For SIP Devices
New
(CUCM 11.0) SIP Profile (Defaults Modified for Example)
Desktop and Room System use of UDP Port Ranges
Media and Signaling Port Range Common
1 2 3
Registration
Audio and
Config File Video
Unified 17000-17999
CM SEP[MAC_addr].cnf.xml
SIP Signaling
Media Port Range UDP
17000-17999 TCP 5060
SIP Signaling Port TCP
5060
DSCP Settings
1. Client registration, download configuration file

2. Entire port range used by both audio and video streams (no audio
and video differentiation possible with port range alone)
Jabbers use of UDP Port Ranges
Media and Signaling Port Range Common
Upper Half
1 2 for Video 3
Registration 4000-4999
Lower Half
Config File for Audio
Unified Jabber 3000-3999
CM SEP[MAC_addr].cnf.xml splits media
range in SIP
Media Port Range UDP Jabber Clients
half Signaling
3000-4999
TCP 5060
5060
DSCP Settings

2. Split media port range in half, upper half for video and lower half for
audio
New
Jabbers use of UDP Port Ranges (CUCM 11.0)
Media and Signaling Port Range - Separated
Video
1 2 Range 3
Registration 5000-5999
Audio
Config File Range
Unified Jabber 3000-3999
CM SEP[MAC_addr].cnf.xml uses the
two port SIP
Media Port Range UDP Jabber Clients
ranges for Signaling
3000-3999
media TCP 5060
Video Port Range UDP
5000-5999
5060
DSCP Settings

2. As of version 11.0, CUCM can provide Jabber with two separate port
ranges for audio and video streams
Identification and Classification 4 SINGLE
ACL Remark
UDP Ports DSCP
3xxx EF
5xxx Jabber classification:
AF42
Audio streams w/ UDP Port
5060 / Range 3xxx (voice-only
CS3
5061 Untrusted and video calls) marked EF
Jabber Video streams w/ UDP Port
Range 5xxx marked AF42
Match SWITCH IP Phone, Smart Desktop,

EF Room System Endpoints:
UDP Ports AF41 Audio streams w/ UDP Port
IP Phone / Smart Desktop DSCP
Range 17xxx and marked
CS3 EF (voice-only and video
EF 17xxx calls) mark EF
Video streams w/ UDP Port
AF41 17xxx
Room System Endpoints 5060 / remark AF41
CS3
5061
Spark Clients
Cisco Spark Media Flows
Without Hybrid Media Node
Cisco
Collaboration
Cloud
Internet media
Edge
Internet
Small Branch
MPLS WAN
WAN
Spark Room Edge
Large Branch
Endpoints Spark Clients
Campus
Cisco Spark Media Flows
With Hybrid Media Node
Cisco
Collaboration
Hybrid
Media
Cloud
Node
ECP
Internet
MPLS WAN
Spark Room
Endpoints Spark Clients
Branches
Port Usage Today Spark
Enterprise
Internet
Network
TCP TCP
443
Cisco
TCP/UDP Collaboration
5004 Cloud
Spark Room
Endpoints
TCP TCP
443
TCP/UDP
5004
Spark Clients
Port Usage Today Spark + Hybrid Media Node
Enterprise
Internet
Network
TCP TCP
443
Cisco
TCP/UDP Collaboration
5004 TCP TCP Cloud
Spark Room 443
Endpoints 33436-
33598
33436- TCP/UDP
TCP TCP 33598 5004
443
Hybrid 33436-
Media Node 33598
TCP/UDP
5004
Spark Clients
Unity Connection QoS
System settings Advanced Telephony
Sets QoS for audio, video and signaling (SCCP/SIP) separately
Default = Audio (46 / EF), Video (34 / AF41), Signaling (24 / CS3)
NOTE: When doing clustering over the WAN for Unity Connection nodes use the CLI command
utils cuc networking dscp on, sets intra-cluster traffic with a DSCP of 18 / AF21.
TelePresence Server QoS (Virtual and HW)
Network QoS
Sets QoS for audio and video separately
Default = Audio (46 / EF) and Video (34 / AF41)
Default Recommended
TelePresence VCS and Expressway Cisco Expressway Core
System Quality of Service (C) and Edge (E)
DSCP Signaling value 24 (Default) CS3

DSCP Audio value 46 (Default) EF
DSCP Video value 34 (Default) AF41
DSCP XMPP value 24 (Default) CS3
New in
X8.9
Cisco Meeting Server
Cisco Meeting Server
CMS DSCP is configured through command line (CLI).
Changes require a server restart!
Defaults for all values are DSCP 0
Commands to Set QoS:

us-cms1> dscp 4 signaling 24
us-cms1> dscp 4 voice 46
us-cms1> dscp 4 multimedia 34
us-cms1> dscp 4 oa&m 34
WAN Ingress QoS
Marking Policy Ingress Policy 1
! This section applies the policy-map to the Interface

Ingress
Policy
1 Router(config-if)# service-policy input INGRESS-MARKING
! Attaches service policy to interface
2
! This section configures the policy-map to set DSCP

for Trusted and Untrusted Voice, Video and SIP
Signaling on ingress
! This section configures the classes
class-map match-any VOICE policy-map INGRESS-MARKING
match access-group QOS_VOICE class VOICE
class-map match-any PRIORITIZED-VIDEO set dscp ef
match access-group QOS_PRIORITIZED_VIDEO class PRIORITIZED-VIDEO
class-map match-any JABBER-VIDEO set dscp af41
match access-group QOS_JABBER_VIDEO class JABBER-VIDEO
class-map match-any SIGNALING-SIP set dscp af42
match access-group QOS_SIGNALING class SIGNALING-SIP
3
set dscp cs3
class class-default
! This section configures the ACLs
ip access-list extended QOS_VOICE Ingress Policy 1
permit udp any range 17000 17999 any dscp ef
WAN
permit udp any range 3000 3999 any
ip access-list extended QOS_PRIORITIZED_VIDEO 5
permit udp any range 17000 17999 any dscp af41
ip access-list extended QOS_JABBER_VIDEO Egress
Policy
6
permit udp any range 4000 4999 any
ip access-list extended QOS_SIGNALING
permit tcp any any range 5060 5061 dscp cs3
! This section configures the policy-map to set DSCP
permit tcp any range 5060 5061 any dscp cs3
4 for Trusted and Untrusted Voice, Video and SIP
Signaling on ingress
! This section configures the classes
class-map match-any VOICE policy-map INGRESS-MARKING
match access-group QOS_VOICE class VOICE
class-map match-any PRIORITIZED-VIDEO set dscp ef
match access-group QOS_PRIORITIZED_VIDEO class PRIORITIZED-VIDEO
class-map match-any JABBER-VIDEO set dscp af41
match access-group QOS_JABBER_VIDEO class JABBER-VIDEO
class-map match-any SIGNALING-SIP set dscp af42
match access-group QOS_SIGNALING 3 class SIGNALING-SIP
set dscp cs3
class class-default
Egress Classification and Egress Policy 6
Queuing
WAN
! This section applies the policy-map to the Interface
6.1 Router (config-if)# service-policy output EGRESS-QUEUING
! Attaches service policy to interface
! This section configures the bandwidth for all

collab traffic
policy-map EGRESS-QUEUING
6.2
class VOICE ! This section applies the policy-map
priority percent 10 class-map match-all VOICE
! Provisions 10% LLQ to VOICE class match dscp ef
class VIDEO class-map match-any VIDEO
bandwidth percent 30 match dscp af41
! Provisions 30% CBWFQ to VIDEO class
match dscp af42
class SIGNALING
class-map match-all SIGNALING
bandwidth percent 2 6.3
match dscp cs3
! Provisions 2% CBWFQ to SIGNALING class
QoS Architecture:

CONTROL
QoS Architecture
Network
Building an End-to-End Strategy Management
EF
WAN
AF41
BE
Provision Monitor
Control Optimize
Trusted devices Map identified Low Latency

traffic to its correct Queuing (LLQ)
Untrusted devices:
QoS marking Priority Queuing
Map UDP/TCP (PQ)
Port Ranges QoS marking:
Class Based
COS, DSCP
Weighted Fair
Queue (CBWFQ)
Weighted Random
Early Detection
(WRED)
Queuing and Scheduling Agenda
WAN Queuing Considerations Identify Classify Schedule Provision Monitor
IOS Queuing (WFQ, CBWFQ, LLQ)
Congestion Avoidance (WRED)
Dual video queue
EF
Audio of IP Phone
EF PQ
BW Assigned to LLQ Classes

Audio of Desktop Video
TelePresence CS4
AF41
CBWFQ
Video of Desktop Video
TelePresence CS4
AF41
Audio of Desktop Video AF41

CBWFQ
Video of Desktop Video AF41
other queues
Single video queue
EF
Audio of IP Phone
EF PQ

Audio of Desktop Video
TelePresence AF41
AF42
Video of Desktop Video

TelePresence AF41
AF42 Video
CBWFQ
Audio of Desktop Video AF42
Video of Desktop Video AF42
other queues
WAN Queuing Considerations
Summary
AF41 WRED thresholds
(i.e., drop AF41 last) Map audio streams of voice and
video calls (EF) to a priority queue
EF
Audio of
IP Phone
EF Map video streams of video calls
PQ
Audio of Video EF (AF41 and AF42) to a class-based

Audio of Jabber EF queue with WRED:
AF41: higher drop thresholds
(e.g., 50-100% of queue depth)
Video of Video AF41 Video
CBWFQ
AF42: lower drop thresholds
(e.g., 15-35% of queue depth)
other queues
Video of Jabber AF42
During congestion, AF42 traffic is
dropped first:
AF42 WRED thresholds Packet loss triggers rate adaptation
(i.e., drop AF42 first)
Media resilience limits the impact
IOS Queuing
Low-Latency Queuing (LLQ)
IOS Interface Buffers
1 Mbps
VoIP policy-map LLQ
Policer
class VOIP
LLQ
priority 1000

Packets
In Packets
Out
CBWFQ
Scheduler
Tx-Ring
FQ CBWFQ
Pre-Sorters
See Network Infrastructure chapter of SRND 11.0
random-detect dscp <dscpvalue> <min-threshold> <max-
Congestion Avoidance threshold> [mark-probability-denominator]
Weighted Random Early Detection (WRED)

Tail Front
of of
Queue Video CBWFQ Queue
Fair- Direction
Queue
Pre- of
Sorter
Packet
Flow
AF43 Minimum WRED Threshold:

Begin randomly dropping AF43 Packets
AF42 Minimum WRED Threshold: policy-map BULK-WRED

Begin randomly dropping AF42 Packets class DesktopVideo
bandwidth percent 10
random-detect dscp-based
random-detect dscp 34 100 128
Maximum WRED Thresholds for AF41, AF42 and AF43 are set to the tail of the queue in this example
random-detect dscp <dscpvalue> <min-threshold> <max-
Congestion Avoidance threshold> [mark-probability-denominator]
Weighted Random Early Detection (WRED) Recommended Opportunistic Video

Tail Front
of of
Queue Video CBWFQ Queue
Fair- Direction
Queue
Pre- of
Sorter
Packet
Flow

30
AF42 Maximum WRED Threshold: policy-map BULK-WRED
Begin dropping ALL AF42 Packets class DesktopVideo
80 bandwidth percent 10
random-detect dscp 34 120 256 50
256 120
Maximum WRED Thresholds for AF41 are set to the tail of the queue
Congestion Avoidance
WRED Configuration Example: 10 Mbps WAN Interface
100%
Drop Probability
policy-map WAN-EDGE
class VIDEO
~ ~ bandwidth percent 40
~
mark-probability-denominator
5%
WRED AF42
mark 2%
probability AF41
(100/denom)
30 80 120 256
WRED
min threshold Queue Occupancy (packets)
WRED
See Network Infrastructure chapter of SRND 11.0
max threshold
Congestion Avoidance
How to choose WRED Parameters? Good Starting Points
WAN Link Speed 622 Mbps 155 Mbps 34-44 Mbps
10 Mbps 5 Mbps
WRED Values (OC12) (OC3) (E3/DS3)
min-threshold 240 180 120 120 60
AF41 max-threshold 512 384 256 256 128
mark-probability-
50 50 50 50 50
denominator
min-threshold 40 30 30 30 15
AF42 max-threshold 180 135 80 80 40
mark-probability-
20 20 20 20 20
denominator
Video queue bandwidth % 43 53 55 55 30
WAN Queuing Considerations
Summary
Question:
Why implement a single CBWFQ with WRED instead of multiple separate
CBWFQs?
Answer:
Because unused bandwidth from one class of video can be fully used by another
class of video if both classes are in the same CBFWQ.
For example, Jabber video can use the unused bandwidth from room system
video if they are in the same CBWFQ. If room system video were in a separate
CBWFQ, then unused bandwidth from that queue would be equally shared across
ALL other queues!
WAN Queuing and Scheduling
Key Takeaways
Simplified Ingress Remarking Policy
Egress Queuing Policy
Single video queue model is recommended
Use WRED to tune the CBWFQ giving AF42 a higher drop precedence over
AF41
This forces AF42 (Jabber) traffic to reduce its rate during congestion before
AF41 is dropped
Alternative: Use AF41 for ALL video, same WRED values!!!
QoS Architecture:

CONTROL
QoS Architecture
Building an End-to-End Strategy Network
Management
EF
WAN
AF41
BE
Provision Monitor
Control Optimize
Trusted devices Map identified Low Latency Bandwidth

traffic to its correct Queuing (LLQ) Provisioning
Untrusted devices:
QoS marking Priority Queuing
Map UDP/TCP Admission Control
(PQ)
Port Ranges QoS marking:
Class Based
COS, DSCP
Weighted Fair
Queue (CBWFQ)
Weighted Random
Early Detection
(WRED)
Provisioning and Resource Control Agenda
Identify Classify Schedule Provision Monitor
Bandwidth Provisioning
Admission Control: Enhanced Location CAC
A word about Regions
CUCM Locations work in conjunction with Regions to define the characteristics of a
call flow
Regions define the type of compression or bit rate (8 kbps or G.729, 64 kbps or
G.722/G.711, and so forth) that is used between devices
Location links define the amount of available bandwidth for the path between
devices.
You assign each device in the system to both a region (by means of a device pool)
and a location (by means of a device pool or by direct configuration on the device
itself).
Building region matrix to manage max video bit rate (video resolution) for groups of
devices.
Group devices into max video bit rate categories
The smaller the number of groups the easier to calculate bandwidth requirements
Consider the default region settings to simplify the matrices!
A word about Regions
CUCM Locations work in conjunction with Regions to define the characteristics of a
call flow
Regions define the type of compression or bit rate (8 kbps or G.729, 64 kbps or
G.722/G.711, and so forth) that is used between devices
Location links define the amount of available bandwidth for the path between
devices.
You assign each device in the system to both a region (by means of a device pool)
and a location (by means of a device pool or by direct configuration on the device
itself).
Building region matrix to manage max video bit rate (video resolution) for groups of
devices.
Group devices into max video bit rate categories
The smaller the number of groups the easier to calculate bandwidth requirements
Consider the default region settings to simplify the matrices!
Region Matrices for Max Video Bit Rate
Example: Matrix with 3 groups
Re-use regions configured for audio-only IP phones
Audio codec configuration is shared, so if video calls need to use different audio codecs you
need to configure new regions
Per-site regions may not be needed if a single audio codec is used for both intra-
region and inter-region calls
Consider the default region settings to simplify a more complex matrix
Endpoint Room System +

Jabber Immersive + MCU
Groupings Smart Desktop
Jabber 1500 1500 1500
Room System + Smart

1500 2500 2500
Desktop
Immersive + MCU 1500 2500 20000
Provisioning and Resource Control Agenda
Identify Classify Schedule Provision Monitor
Bandwidth Provisioning
Admission Control: Enhanced Locations CAC
Control Plane
NO
E-LCAC
YES
?
Call Admission Control
QoS
Scheduling + Queuing
IP WAN
Data Plane
LCAC Limitations E-LCAC Solutions
Limited WAN Topology Support: Network Modeling:
Hub and Spoke WAN topology support Convert UCM locations to a model capable
Large gap between RSVP and Locations of supporting real network topologies
CAC
Multi Cluster Support: Inter-Cluster (Inter-cluster) CAC:

Multiple Clusters that managed endpoints in Implement a bandwidth-accounting
same branch sites could only inefficiently scheme that works between multiple
subdivide inter-branch bandwidth to avoid Unified CM clusters and dynamically
quality degradation (Ships in the night CAC)
learns the topology from one another
TelePresence: Immersive Bandwidth Allocations:

Did not Support CAC (overlay design) Implement an immersive BW pool in locations
TelePresence and UC or 3rd party video on a CAC
single cluster Provide better CAC interop support between
Limited CAC support for TelePresence video TelePresence Video and Desktop Video
interoperability (P2P calls without an MCU)
Network Modeling - Concepts A
Location Deduct BW!
Administrator builds a Network Model using locations and
links
A Location represents a LAN. It could contain endpoints
or simply serve as a transit location between links for WAN B
C
network modeling
Links interconnect locations and are used to define Deduct BW!
bandwidth available between locations. Links logically Link
represent the WAN link
Weights are used on links to provide a cost to the
effective path. Weights are pertinent only when there is
more than 1 path between any 2 locations D E
Deduct BW!
UCM calculates shortest paths (least cost) from all
locations to all locations and builds the effective paths
The Effective paths are the paths with the least
cumulative weight F
UCM tracks bandwidth across any link that the network
model indicates from originating Location to terminating
location. Effective Path
Network Modeling Locations and Links
Location A Location B
BETWEEN
Link A < > B BW Allocated

BW Allocated
Audio Unlimited BW Allocated Audio Unlimited
Video 100MB Audio 1500k Video Unlimited
Immersive 250MB Video 3000k Immersive Unlimited TelePresence
Immersive 5000k
IN/OUT Links Provide Bandwidth IN/OUT

Accounting Between
Locations
And Interconnect Locations
WITHIN WITHIN
Locations Provide Bandwidth

TelePresence and UC Video
Accounting WITHIN the Linking Locations Endpoints Can Reside in the
Location as well as IN or OUT
Same Location*
of the Location
EUG PDX SEA BLD 2017 Cisco and/or its affiliates. All rights reserved. Cisco Public
New
New Enhanced Locations CAC Feature (CUCM 11.0)
Call Admission Control Enhancement for Audio Portion of Video Calls:
When enabled CAC for video calls deducts the audio portion of the video call
from the voice pool and the video portion of the video call from the video pool (or
immersive pool if its enabled).
Custom QoS settings for SIP Devices ensure that the audio of video and
immersive calls are set to the correct DSCP (EF) to align with the CAC
bandwidth deduction.
Audio and Video Admission Control
Considerations
No Admission Control
Over-provision queues
Rely on Video Rate Adaptation and Media Resiliency Capabilities
Audio is much easier to over-provision in pervasive video deployments
QoS is critical and rate adaption is highly beneficial for both managed/unmanaged
networks
Benefits: Simplicity
Admission Control
Strict provisioning (Mapping CAC to Queuing)
Mobility? Device Mobility feature (Adds OPEX)
Benefits:
Manage lower bandwidth links, use Automated Alternate Routing (AAR) for PSTN redirect
Ensure quality audio during the busy hour by avoiding oversubscription and packet loss
Safe when over-provisioning is not an option
Bandwidth Provisioning and Admission Control
Key Takeaways
CAC audio use the new feature to deduct ALL audio from the voice pool
Simplify the region matrices when possible
Settle on a single audio codec for use within and between sites. This will greatly
reduce the number of required regions.
Determine the need to CAC video and the advantages of not CACing video
Design and Deployment
Design and Deployment Agenda
B2B, C2B and Cloud Interactions
Example Enterprise Deployment
Media Path Summary Media Traversal
C calls A on-premise
QoS-enabled Expressway solution provides firewall
Enterprise Network B
traversal for media
Inside firewall DMZ Outside firewall Expressway C de-multiplexes media

and forwards toward A
(Intranet)
Media Relay
Collaboration
Internet
Services C C calls B off-premise
UCM Expressway Expressway Media is relayed via Expressway C

C E
Optimized Media (future ICE support)
D B calls D off-premise
SIGNALING
Both B and D are ICE-enabled
A MEDIA
STUN binding success
Media flows are optimized between
endpoints
QoS Best Practices
All media coming from remote endpoints is relayed via Expressway C
Expressway C can mark traffic with a Multiple DSCPs

System Quality of Service
DSCP Signaling value 24 (Default) CS3
DSCP Audio value 46 (Default) EF
DSCP Video value 34 (Default) AF41
DSCP XMPP value 24 (Default) CS3
Recommendations:
Leave DSCP Defaults and Configure DSCP trust
on access switch port connected to Expressway C
(optional) If using CAC, configure Device Mobility
using Expressway Cs IP addresses and assign endpoints to
an Internet location
WebEx Considerations: CMR Hybrid and CMR Cloud
Voice/video traffic is relayed

CUCM
WebEx into the Enterprise network
CMR Hybrid
via VCS/Expressway C/E
DSCP marking options:
On-premise
Conferencing
(TS) Expwy-C Expwy-E
AF42 (36) for all video traffic
Internet from VCS/Expressway C
Company A AF41 (34) for all video traffic
from VCS/Expressway C
CUCM
WebEx For CMR cloud, AF41
CMR Cloud
recommended for
Expwy-C Expwy-E environments where CAC is
configured
Company B Internet
WebEx Considerations: VoIP and Video in WebEx Meetings Client
Enterprise Network
WebEx Video in WebEx Meetings

Client IP WAN client is rate-adaptive
Internet
Branch office HQ
To provide better-than-
Protocol Port Number WebEx Access Type best-effort QoS when
TCP 80 WebEx Client Access traversing Enterprise
TCP 443 WebEx Client Access - Secure Traffic (SSL Sites) WAN:
TCP/UDP 1270 WebEx Client Access (Non SSL Sites)
classify traffic with an ACL
based on UDP ports 9000
TCP/UDP 53 WebEx Domain Name System (DNS)
and 9001
TCP/UDP 5101 WebEx MMP
remark it with DSCP AF42
TCP 8554 WebEx Audio Streaming Client Access
UDP 7500-7501 WebEx Audio Streaming
UDP 9000-9001 WebEx VoIP/Video

CUCM
Multipoint
Video (TS) Central
500 users:
Site
500 Jabber clients
300 IP phones Expressway B2B
30 video endpoints
(room + desktop) Cloud
1 immersive telepresence Services
2 Telepresence Servers
Mobile
Internet Users
MPLS VPN
50 users: 15 users: 5 users:

50 Jabber clients 15 Jabber clients 5 Jabber clients
30 IP phones 10 IP phones 3 IP phones
6 video endpoints 2 video endpoints 1 video endpoint
(room + desktop) (room)
Large Branch Small Branch Micro Branch
Identification and Classification SINGLE
ACL Remark
UDP Ports DSCP
Jabber classification:
3xxx EF Audio streams w/ UDP Port
Range 3xxx (voice-only
5xxx AF42 and video calls) marked EF
5060 / Video streams w/ UDP Port
CS3
5061 Untrusted Range 5xxx marked AF42
Jabber
Match SWITCH IP Phone, Smart Desktop,

EF TelePresence Endpoints:
IP Phone / Smart Desktop AF41 Audio streams w/ UDP Port
DSCP UDP Ports
Range 17xxx and marked
CS3 EF (voice-only and video
EF 17xxx calls) mark EF
Video streams w/ UDP Port
AF41 17xxx
TelePresence 5060 / remark AF41
Endpoints
CS3
5061
Scheduling AF41 WRED thresholds
(i.e., drop AF41 last)
Audio of
IP Phones EF Audio/EF is mapped to the priority
queue
EF PQ Includes audio streams from video
Audio of endpoints and Jabber clients (voice-
Video EF
only and video calls)

Audio of EF
Jabber Video streams of video calls (AF41)
and video streams of Jabber calls
Video of (AF42) are mapped to the same
Video AF41 Video CBWFQ
CBWFQ
WRED is configured on the video
Video of queue:
AF42
Jabber min-max thresholds for AF42:
~10% - ~30% of queue limit
queues
min-max thresholds for AF41:

other
AF42 WRED thresholds

(i.e., drop AF42 first) ~45% - 100% of queue limit
Provisioning and Resource Control
Admission Control
Priority queue is over-provisioned
EF Voice or protected by ELCAC for audio
Queue from all users at the site
AF41
Video queue is over-provisioned
for video room and desktop
Video endpoints:
Queue WAN Usage ratios are applied to desktop
AF42 Link video endpoints
Jabber video streams can use any
bandwidth unused by video room
systems
other During congestion, Jabber video
voice-only call
queues streams are subject to WRED drops

and dynamically reduce video bitrate
Audio is not impacted by drops
Provisioning Central Site
Voice queue (PQ): 10 Mbps (L3 bandwidth)
500 users: 125 calls @ G.711/G.722
500 Jabber clients Voice
300 IP phones 10 Mbps
30 video endpoints
Unified CM Location link bandwidth for the
(room + desktop) voice pool:
1 immersive telepresence
2 Telepresence Servers Video WAN
125 * 80kbps = 10,000 kbps
55 Mbps
Link
Video queue: 55 Mbps (L3 bandwidth)
voice 100
10% Mbps Immersive Endpoint: 2 Mbps * 1 call = 2 Mbps
Video endpoints: 1.2 Mbps * 30 calls * 0.2 = 7.2
Default
31 Mbps
Mbps
video
55% TelePresence Servers: 1.5 Mbps * 40 calls * 0.5
= 30 Mbps
55 - (2 + 7.2 + 30) = 15.8 Mbps for Jabber video
11 Jabber Video calls @ 720p or 18 @ 576p
or 50 @ 288p (plus any leftover bandwidth)
Provisioning Large Branch
Voice queue (PQ): 3.4 Mbps (L3
50 users: bandwidth)
Voice
50 Jabber clients
30 IP phones
3.4 Mbps 42 calls @ G.711/G.722
6 video endpoints
(room + desktop)
Unified Location link bandwidth for the
Video WAN
18.7 Mbps
Link
voice pool:
42 * 80 Kbps = 3.360 Mbps
voice 34
10%
Mbps
Video queue: 18.7 Mbps (L3
video
Default
10.5 Mbps
bandwidth)
55%
Video endpoints: 1.2 Mbps * 6 calls = 7.2 Mbps
18.7 - 7.2 = 11.5 Mbps for Jabber video
6 Jabber Video calls @ 720p or 10 @ 576p
or 36 @ 288p (plus any leftover bandwidth)
Provisioning Small Branch
Voice queue (PQ): 2 Mbps (L3
15 users: Voice
2 Mbps
bandwidth)
15 Jabber clients
10 IP phones
25 calls @ G.711/G.722
2 video endpoints
(room) Video WAN Unified Location link bandwidth for the
4 Mbps Link
voice pool:
10
Mbps 25 * 80 Kbps = 2 Mbps
voice
20%
video
Default
3.6 Mbps
Video queue: 18.7 Mbps (L3
40%
bandwidth)
Video endpoints: 1.2 Mbps * 2 calls = 2.4 Mbps
4 - 2.4 = 1.6 Mbps for Jabber video
1 Jabber video calls @ 720p or 2 @ 576p or
5 @ 288p (plus any leftover bandwidth)
Provisioning Micro Branch
5 users: Broadband Internet connectivity +

Voice
5 Jabber clients 1.25 Mbps DMVPN to central site
3 IP phones
1 video endpoint WAN
Video
1.5 Mbps Link Configure interface of VPN router to
5
match broadband uplink speed
Mbps
Internet Default Enable QoS on VPN router to
2 Mbps
DMVPN
prevent bufferbloat* from TCP flows
Tunnel Broadband
Modem/Router
Asymmetric download/upload
QoS-enabled
Router
voice
25%
broadband: consider limiting
video
30% transmit bitrate on video endpoint
*: see reference slide #31 in this session
CAC for Video Considerations
In branch sites with lower-speed WAN links over-provisioning the video queue is
not feasible
E-LCAC will be applied also to video streams in these sites
Use site-specific region configuration to limit maximum bandwidth used by video
endpoints and Jabber clients
Device mobility needed if Jabber users roam across sites
Provisioning Large Branch with Constrained WAN Link (CAC for video)
Voice queue (PQ): 2 Mbps (L3 bandwidth)
50 users:
Voice 25 calls @ G.711/G.722
50 Jabber clients 2 Mbps
30 IP phones
4 video endpoints
Unified Location link bandwidth for the
(room + desktop)
Video WAN voice pool:
4 Mbps Link
25 * 80 Kbps = 2 Mbps
10
voice
Mbps Video queue: 4 Mbps (L3 bandwidth)
20%
Default Possible breakdown:
video 3.6 Mbps
40% 1 call @ 720p (1,220 kbps) + 3 calls @ 576p (810
kbps) = 3,650 kbps
Or 2 calls @ 576p (768 kbps) + 5 calls @ 288p
(320 kbps) = 3136 kbps
CUCM Location link bandwidth for video
calls: 3.2 Mbps (L3 bandwidth)
Leaves room for L2 overhead and burstiness
Conclusion
Combine QoS tools, media resilience and dynamic adaptation to build a self-
regulating system that makes optimal use of available network resources
Use CAC when and where needed
When managing bandwidth with Media Resilience and Rate Adaptation techniques is
not an option (i.e. extreme contention on WAN bandwidth)
Future:
Leverage SDN approach to simplify QoS deployment and enable dynamic QoS
policies for media from untrusted devices
Further enhancements of media resilience and rate adaptation techniques
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QoS Strategies and

Smart Media Techniques for

This 2-hour session is divided in two main sections:

Mobile, software endpoints

On-premise Call Control

Remote Sites Home/Mobile Users

On-premise Call Control

Remote Sites Home/Mobile Users

On-premise Call Control Spark

Remote Sites Home/Mobile Users

On-premise Call Control Spark

Remote Sites Home/Mobile Users

LTRF 0111010001 R1 FEC

... ... R2 1011010010

Leverage media resilience and

Identification and Classification

Mobile and Remote Access

Smart Media Techniques

VIDEO Bandwidth Bandwidth:

better voice video

Out of Sync (OOS)

Loss of a P-frame triggers request for a new I-frame

Flickering/pulsing of video when new I-frame arrives

Delay seen by 1 audio stream

Delay seen by 1 audio stream

Smart Media Techniques

LTRF with Repair

Optimize use of available

600 Endpoint packet

Keep encoder and decoder in sync with active feedback messages

LTRF Binary XOR R1 FEC

Media resilience mechanisms help mitigate impact of video traffic on the

Dynamic rate adaptation creates an opportunity for more flexible provisioning

IDENTIFICATION & CLASSIFICATION

Weighted Random BRKCOL-2017

Examples: Collab Server, MCU, TS

CDP Support required

Trusted Endpoints / Devices

Untrusted Endpoints / Devices

Immersive and Room

IP Phones Jabber and DX Series

Immersive and Room

IP Phones Jabber and DX Series

DSCP for Audio Calls Audio Only 46 EF Voice

DSCP for Video Calls Video of Video 34 AF41 Video

DSCP for TelePresence Calls Video of TP 32 CS4 Video

DSCP for Audio Calls Audio Only 46 EF Voice

DSCP for Video Calls Video of Video 34 AF41 Video

DSCP for TelePresence Calls Video of TP 34 AF41 Video

DSCP for Audio Calls

DSCP for Audio Calls

Fixed setting (Not Configurable)

Check Devices for Capability:

Cisco Unified Reporting Tool > Immersive Video Support for

TelePresence Desktop Jabber

Prioritized Video: AF41 Opportunistic Video: AF42

TelePresence Desktop Jabber

Prioritized Video: AF41 Opportunistic Video: AF42

1. Client registration, download configuration file

1. Client registration, download configuration file

1. Client registration, download configuration file

Match SWITCH IP Phone, Smart Desktop,

DSCP Signaling value 24 (Default) CS3