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Effective Deployment and
Migration Strategies of IP PBX
Alfredo Rizzo
Adapt
www.teamadapt.com
[email protected]
773.634.2044
What Was Holding VOIP Back?
• Proprietary
Protocols
• Lack of Features
• Reliability
Perceptions
• Legacy
Integration
Models
• Cost of End
Points
• Declining
Traditional Costs
(LD)
• Lack of
Applications
Convergence
• Remote
Survivability
Session Outline
• Define and Understand "Quality of Service"
– What are the Issues Affecting QoS
• Network Exposure and Security
– What is the impact of NATs and Firewalls on a VoIP
Implementation
• Issues when Integrating with Existing / Legacy
Infrastructure
• Preparing Your Network
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Cabling
Network Core
Power
Remote Survivability
Define and Understand “Quality
of Service” (“QoS”)
• Define “Quality”
• What are the Issues Affecting Quality?
– Delay (Latency)
– Jitter
– Bandwidth
• Define QoS
–
–
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Define Packet Shaping
Where and How to Implement QoS
LAN vs. WAN
Monitoring and Reporting
Define and Understand “Quality”
What is Quality? Quality is a
characteristic that can only be
measured in words, not numbers. A
phone call can be “good”, “noisy”,
“jittery” or “unintelligible”.
A way of measuring Quality
• A group of users make calls and rate them
“Excellent”, “Fair”, “Poor”, etc. The quality
of the calls will be the average of all their
scores, or the Mean Opinion Score (MOS).
• The European Telecommunications
Standards Institute (ETSI) developed an
accepted way of measuring voice quality
called the “E-Model”, which is based on the
MOS.
Delay can Affect Quality
• Delay (Latency) is defined as:
– the amount of time it takes for sound
from a talker’s mouth to arrive at the
listener’s ear.
• The maximum amount of delay that is
acceptable for a one-way
transmission is described by the
International Telecommunications
Union in Document G.114
G.114
ITU Recommendation
(in ms)
Private Network
Recommendation
(in ms)
Description
0 – 150
0 – 200
Acceptable for most
applications
150 – 400
200 – 250
Acceptable provided that
the administrators are
aware.
400+
250+
Unacceptable
G.114
Manage Your Delay Budget
• Serialization Delay - the speed at which the
router processes each packet. This adds
precious milliseconds to the delay budget.
Older, slower routers are not
recommended for voice applications.
• Packetization Delay - the amount of time it
takes for the telephony device (IP Phone,
Router, IP PBX) to packetize the audio
sample.
• Propagation Delay – the amount of time it
takes for packets to travel down the
medium.
Jitter
– Variation in delay
– Caused by network congestion
– Causes jitter buffer overruns can occur
Bandwidth
• How much is enough for IP Telephony?
– Depends on:
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•
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Number of simultaneous sessions
Codec(s) used
Will Voice Activity Detection (VAD) be used?
Transport Protocol (cRTP, etc.)
Control Protocol (RTCP)
Data Link Protocol (Ethernet, Serial, ATM, Frame)
– Very different considerations for LAN vs. WAN
Calculating Required Bandwidth
Quality of Service (QoS)
• Quality Of Service (QoS) refers to the
mechanisms in the network that make the
actual determination of which packets have
priority.
• QoS policies give priority to traffic based on
their relative importance to the business.
• However, this only prioritizes traffic; it does not
guarantee a level of bandwidth. Without
guaranteed bandwidth, high priority applications
will still experience performance degradation.
Traffic Shaping
• Traffic shaping can be used to
actually guarantee bandwidth for
certain types of traffic and limit
available bandwidth for others. Traffic
shaping can provide an effective way
to prevent congestion, minimizing the
impact of rogue traffic on missioncritical applications.
LAN Settings
• Where to I “tag” my packets?
– The VoIP endpoint can tag the packet, and the
switch can trust its tagging
– It is easiest to tag at the switch ports, if those
are used exclusively for VoIP devices
– This avoids router packet inspection – all they
must do is maintain the tags and enforce them
(or a separate packet shaper can)
• LAN-only traffic can use G.711, no VAD
– Less packetization delay
– Less expensive hardware
WAN Settings – Manage your
Scarcest Resources Most Efficiently
WAN Settings
• Can your router do traffic shaping or
do you need an external device?
• If using frame relay, you can use
separate PVCs for voice and data,
and thus guarantee your required
voice bandwidth
• Protocol selection and compression
algorithms are very important
Monitoring and Reporting
• Many packages available
• Allows you to do “what if” scenarios
• Allows you to report on QoS
performance and adherence to
requirements
• Allows you to plan for future growth
What Can Affect QoS
• Bad design/planning, resulting in:
– Inadequate network equipment to enforce QoS
and shape traffic
– Insufficient bandwidth
– Incorrect assumptions regarding bandwidthaffecting factors
– Insufficient management/reporting tools – you
must inspect what you expect
• Lack of end-to-end adherence
– Within your network
– Within others’ (carriers, etc.) networks
Network Exposure and Security
• What is the impact of NATs and
Firewalls on a VoIP Implementation?
– Significant
– Security issue – many VoIP protocols
use clear text messaging
– When extending VoIP service to remote
and home offices, you must consider
VPNs, which provide encrypted
tunneling
Firewalls and NAT
• Let’s consider the audience – carriers
and enterprises that will provide both the
VoIP and IP services to users. You
control the firewalls and NATs.
What’s the Problem with NAT?
• VoIP protocols for session control (SIP,
H.323, MGCP, MEGACO) are
Application Layer protocols
• But IP operates at the Network Layer
(Layer 3) and NAT devices change that
address.
– Now VoIP message (in UDP) comes back to
the sender’s public address, and is
discarded.
What’s the problem with
Firewalls?
• Firewalls control all TCP and UDP port
availability through policies.
• Typically only certain ports (static) are
allowed from certain source addresses
to certain destination addresses
• But VoIP sessions use a dynamically
generated port address just for that
session. No two sessions will use the
same port address at the same endpoint
(i.e., IP PBX).
What Can We Do?
• The IETF has come up with two ways
of getting around these problem:
• MIDCOM (RFCs 3303, 3304)
• STUN (RFC 3489)
• uPNP – created by an industry
consortium, primarily with the goal of
solving this puzzle in home networks
that use a NAT device for outside
communications. OS-dependent.
Middlebox Communications
(MIDCOM)
• MIDCOM is an IETF protocol that allows an
intermediate piece of hardware to mediate
the SIP (or H.323, FTP, etc.) session,
thereby getting around the problems of
NAT.
• Architecture http://www.ietf.org/rfc/rfc3303.txt
VoIP, NATs, and Firewalls
STUN
• Simple Traversal of UDP Networks
• A temporary solution until MIDCOM
reaches widespread adoption
• A service that can run on a server, or on a
piece of dedicated hardware
• Its only job it to translate the UDP packets
so the audio stream can make it to its
intended destination.
• Does not work with inbound-initiated
sessions
STUN
• Simple Protocol
• Works with Existing NAT
• Main Features
– Allows Client to Discover Presence of NAT
– Works in Multi-NAT Environments
– Allows Client to Discover Type of NAT
•
•
•
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Symmetric
Full Cone
Restricted Cone
Port Restricted Cone
– Allows Discovery of Binding Lifetimes
– Allows Clients to Discover if They are in the Same
Address Realm
– Stateless Servers
STUN – Binding Acquisition
• Client sends STUN Request
to Server
– STUN Server can be
ANYWHERE on Public Interne
• STUN Server Response
• Client knows Public IP for
that Socket
• Client Sends INVITE Using
that IP to Receive Media
• Call Flow Proceeds Normally
– No Special Proxy Functions
• Media Flows End-To-End
More Help is on the Way
• RFC 3581 - Making SIP “NAT
Friendly”
– “This extension defines a new parameter
for the Via header field, called "rport",
that allows a client to request that the
server send the response back to the
source IP address and port from which
the request originated.”
– Addresses SIP only, not RTP or other
session control protocols
Application Layer Gateways
(ALG)
• Firewall / NAT devices that give
special treatment to VoIP streams.
• Can perform RTP Relay
Issues when Integrating with
Existing/Legacy Infrastructure
• Support for analog devices
• Tie lining to legacy PBX – need a gateway?
• Coordinating extension and dial plans (no
news here)
• Messaging
– who does it? Will need cover paths and pilot
numbers into TUI.
– If both do it, will you replicate?
• AMIS – Audio Messaging Interchange Specification
• VPIM – Voice Profile for Internet Mail
New Issues
• Emergency Service (911/E911)
– Do you need to provide 911 service for
telecommuters and remote offices? What
happens if they dial 911 from their IP Phone?
– When the number follows the user, should 911
info? Who keeps that information?
– Some states require businesses with PBX
equipment to pass 911 information to the PSAP
based on the user’s location
Preparing Your Network
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Cabling
Network Core
Power
Remote Survivability
Cabling
• Cabling options:
– Separate CAT5 jacks for each IP phone/device.
• More wiring
• Less expensive phones
• Less switch configuration
– Same CAT5 jack for phone and PC
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Less wiring
More expensive phones
More switch configuration (inter-switch trunking)
If you reboot your phone, your PC looses its network
connection
Network Core
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Switches must support QoS
Consider switch redundancy options
Voice should go on separate VLANs
Configuring phones –
– DHCP for IP address assignment
– Mostly TFTP for configuration file
download
– new RFC for SIP information through
DHCP
Power
• Typically, you must maintain power to
phones for several hours in the event of an
outage
– 911 calling
– Business continuity, at least to a subset of
phones
• Possible solutions
– PoE – Power over Ethernet – IEEE 802.3af
• Powered Switches
• In-line Powered Patch Panels
– FXS Media Gateways in the closet (with UPS)
– UPSs on all phones
Remote Survivability
• Phones must be able to “get out” in
the event that a WAN link connecting
them to their IP PBX goes down
• Can be vendor-specific or standardsbased
• 911 – ANI must reflect correct
address to PSAP
Questions / Comments