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Effective Deployment of IP-PBXs

Date: Wed. October 15th

Speaker:

Tom KeatingCTO/Executive Technology Editor, TMC Labs

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BIO

• B.S. Computer Engineering

• With TMC for past 10 years

• Executive Technology Editor & CTO

• Head of TMC Labs

• Background in technologies includes PDAs, telecom, call center & CRM technologies, CTI, wireless, VoIP

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IP-PBX Advantages• Attractive ROI on long distance calling

– Domestic calling can pay back the cost of the IP-PBX in less than a year

– International calling can often pay back in just weeks.• Less expensive Than Traditional PBXs

– Up to 40% less expensive than traditional PBX solutions– If you include advanced integrated features (web call-through,

remote voice capabilities, no CTI links needed) IP-PBXs are even more cost advantageous

– “One wire” to the desktop• One system to support multiple sites

– Unified corporate-wide 3-4 digit extension dialing– Centralized voice mail– Consolidate toll and inbound 800 calling

• Total Cost of Ownership (TCO) can be lowered– A single voice and data network (Administration – Training)– Centralized administration for multiple sites– Lower costs for (MACs) Moves, Adds, and Changes

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What are the Issues That Face IP-PBX Implementation?

• Quality of Service– Voice Quality– Class of Service

• Standards– Interoperability– H.323 vs. SIP

• Security– NAT firewall– VPN

• Availability/survivability• Cost/Budget• Existing/legacy Infrastructure

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VoIP Enabling your Legacy Solutions

Public/PrivateNETWORK

IP Phone

LegacyPhones

RemoteIP Phone

VoIP Gateway with Legacy Phones

VoIP Adapter with Legacy Phone

Corporate Office

Branch/Remote OfficeLegacy Key/PBX

w/ VoIP blade

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100% “All IP” Convergence

Public/PrivateNETWORK

Corporate Office

Road WarriorUSB or IP Phone

Branch/Remote OfficeIP Phones

IP-PBX

TeleworkerIP Phone

IP Phones

Router

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So now you are Ready to Deploy, now what?

• Do you plan to DIY?

• Have professional installers? (VAR, interconnect, network reseller, etc.)

• Combination of the two? (install it yourself and have professionals come if any problems crop up)

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QoS Mechanisms• ToS – Type of Service• 802.1p – Layer 2 with 8 class levels• IntServ – Integrated Services• DiffServ – Differentiated Services• RSVP – Resource Reservation• CoS – Class of Service• CAR – Committed Access Rate• CIR – Committed Information Rate

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IP Class of Service Type of Service (TOS) for CoS Marking

• Now known as Differentiated Services Code Point (DSCP)

• Usually set to 0, but may indicate particular Quality of Service needs from the network

• The DSCP defines one of a set of class of service

Options and Padding

Destination IP Address

Source IP Address

Protocol ChecksumTTL

OffsetFlags

TOSVer/IHL

Identifier

Total length

32 Bits

IP Data

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Choose a VoIP Codec…

Coding algorithmBandwidt

hSample

size

IP bandwidt

h

G.711 PCM 64kbps 0.125ms 80kbps

G.723.1 ACELP

5.6kbps

30ms

16.27kbps

6.4kbps 17.07kbps

G.726 ADPCM 32kbps 0.125ms 48kbps

G.728 LD-CELP 16kbps 0.625ms 32kbps

G.729(A) CS-ACELP 8kbps 10ms 24kbps

*Source: www.erlang.com

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Bandwidth Calculations• Assumptions:

– Each voice sample carries a IP/UDP/RTP header overhead of 320 bits. – 1 Voice Sample is defined as the number of packets required for 1 second of

voice– Assume 1 packet contains or equals 20ms of voice samples– Therefore 50 packets=1s of voice samples (or 1 Voice Sample)

• Q: What is Packet Frequency?• A: Defined as number of packets required to be sent for 1 second of voice sample.

• Q: What is the Packet Frequency for 20ms of voice samples per packet?• A: 50 (it’s 50 packets/Voice Sample) (see Tip 1)

• Q: What is the overhead in bits per one second of voice samples (assuming 20ms Voice Sample)?• A: 16000 (16kbps) overhead bits per 1 second of Voice Samples (See Tip 2)

– Or 16,000 (16kbps) overhead bits/Voice Sample_________________________________________________________________________________• Tip 1: The Packet Frequency is the inverse of the duration in seconds represented by the voice

samples. (1/.02 = 50)• Tip 2: If each voice sample has 320 bits of overhead, then :

– 50 packets/voice sample X 320bits of overhead bits/voice sample = 16,000 overhead bits/voice sample

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The Weakest Link

• VoIP is only as strong as its “weakest link” (bandwidth bottlenecks)

– Leased line, frame relay, T1/E1 to branch offices.

• However, you can’t focus on just the weak links, since even “strong” links (10/100BaseT networks) can have network issues

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Troubleshooting VoIP network issues• Select a “generalized” network analyzer (i.e. Ethereal,

Network Instruments)– Good for finding “top talkers” (bandwidth hogs), general

network issues • Select a “specialized” network analyzer that specializes in

VoIP analysis (e.g. Agilent, Empirix, Radcom, Qovia)– Good for pinpointing specific VoIP issues, measuring

voice quality (MoS-Mean Opinion Score, PSQM-Perceptual Speech Quality Measurement scores), and graphically depicting VoIP ladder diagrams

• Find VolP experts/consultants to troubleshoot your network– Psytechnics “swat team”

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Security & SLA Challenges• For branch office IP-PBX deployment use Voice VPNs

for added security (though adds some latency due to encryption)

• How do I hold my service provider accountable for site-to-site Voice over IP-VPN performance?

• How do I verify I am getting the IP Class of Service performance I am paying for?

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One Sample “SLA” Solution

• “Unknown” traffic classification to identify mis-configured router

• Collected for 8 Classes

• Support TOS/DSCP bit ranges

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Open, Flexible, Extensible• When choosing an IP-PBX you want to make sure it has an

open architecture• Ability to utilize different vendors (Not locked into a vendor) for

different telephony applications• Wireless (802.11) support • PDA Phone support• Useful applications (i.e. unified messaging, screenpop)• Call center applications (predictive dialer, skills-based

routing, QM/call recording, etc.)• Support for branch offices/remote workers• Web application integration

• IP Contact Centers – collaboration –• Instant Messaging - Video

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• Single-line or multi-line phone? Not everyone needs a

multi-line phone – Why waste the money?

• Operators can use an Attendant Console (whether

software or hardware-based)

• Softphone with USB phone

• Headset option

Choosing the Right IP-Phones for each Individual

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Multi-Line IP Phone Consideration:

• Are there enough line appearances to fulfill the user’s call processing responsibilities?

• Buy IP Phones with as many line appearances as possible to accommodate future expansion and ease of use.

• Will additional IP devices be connected to the IP phone (PC, IP Phone, etc.)?

• Buy compatible multi-line IP Phones with multiple Ethernet ports.

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Multi-line Phones:

TelStrat (16 lines) Swissvoice (2 lines)

Cisco (4 and 6 lines) Teledex (2 lines)Polycom (3 lines) Mitel

Single Phones:

Pingtel SnomCisco SwissvoiceTeledex MitelSymbol (*WiFi) IP Dialog

Sampling of IP-Phones…

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H.323 Or SIP?• Today it is still an H.323 world

– Established protocol – lots of deployed solutions – Lots of vendors equipment interoperate– Telephony like signaling – complex

• SIP is the “hot” new VoIP standard– Newer standard, Internet-based, easier/less complex– Simpler – Create “sessions”

• Like HTTP – Wealth of available development resources• Integration with web based applications• Access via any SIP enabled technology – phone, video, IM

• Make sure your IP-PBX is 100% compliant with relevant VoIP protocols so that 3 rd party phones will work with it.

• For the immediate future both will be around, but expect SIP to eventually dominate

• There are transcoders for interoperability from H.323 to SIP and vice-versa

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Network Characteristics Required for High Quality Voice

• LANs provide good network characteristics – even better voice than the PSTN– Switched Ethernet to the desktop – lots of bandwidth– Estimate bandwidth requirements

• LAN –G711 with 20 ms packetization• Assume worst case phone utilization - 1 of 4 phones

• WAN characteristics are more challenging– Latency: Typical <100 ms Maximum < 200ms– Jitter: Typical < one sample Maximum < 40ms– Packet Loss: Typical – < 1% Maximum <5%– 85% maximum utilization at peak traffic

Source:www.erlang.com

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Minimize Bandwidth Requirements On The WAN

• Minimize overhead, maximize bandwidth on your WAN links. But how?

• First, you have to understand and minimize bandwidth requirements– Go with G.729/G.723. Is the voice

quality acceptable?– Silence suppression

• 35% savings– Header compression

• 50% savings– RTP multiplexing

• 50% savings

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QoS Can Help Support Good Voice Quality

• Support of VLAN and QoS capabilities

– IP Phones support of VLAN tagging• Keep all the voice traffic on a layer 2 VLAN

dedicated to voice and signaling traffic – 802.1q• Prioritization – 802.1p

– VoIP devices support of TOS or Diff-Serv• Prioritize voice traffic through the routers and

switches

– Required at every point in the network

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Supporting QoS Can be Challenging

• QoS becomes more challenging the larger the network– Multiple methods of differentiating service classes

• Requires compliance to the rules of the administrative domain• QoS must understand all methods of differentiation

– QoS must be supported at every hop (internal & external) Or NO QoS

• Any hop where QoS is not supported becomes a potential congestion point

• QoS characteristics can be lost between domains

• QoS is most needed where it is hard to support– WANs have the greatest latency.– Many remote sites use the Internet (as opposed to dedicate

leased line)

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Security At The Edge – Getting Around NAT

• Tale of two data paths – signaling and voice– Call signaling provide the information to establish the call

and is translated by the NAT

– The signaling embeds the source and destination IP addressing for the media – unknown to the NAT

• Various solutions– VoIP aware firewall – upgrades to existing some firewalls

– VoIP to VoIP gateway – VoIP Call Relay

– Site to Site VPNs

• Adds bandwidth and latency due to the encryption

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Security For The Media – Who Could Be Listening In?

• There are steps that can improve the security of the media stream.– Protection behind the firewall

• Support with some VoIP NAT capability

– Put VoIP equipment on virtual LANs and enable the security features on the switches.

– VPN routes between sites to support VoIP• Latency? QoS?

– RTP encryption where possible

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High Availability

• Deliver power to the phones over the Ethernet– Find an IP-PBX that supports the 802.3af power-over-

Ethernet standard– Ethernet switches that deliver the power on back-up

power supplies• Remote office gateway may require one or more PSTN

failover lines in case the connection to the IP-PBX is lost.• Real-time PSTN back-up trunks on the IP-PBX for off-net

calling

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Key Steps To Follow When Deploying IP Telephony

• Planning and evaluating for VoIP deployment– Assessment of the existing infrastructure– Evaluation of VoIP requirements

• Upgrade the IP network as required• Lay out the network, perform network trending• Run a pilot – Identify any issues, monitor traffic with

a network analyzer• Perform Full implementation• Continue to perform monitoring with network

analysis tools, VoIP quality of service tools, and voice quality measurement tools

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Conclusion• With proper planning, an IP-PBX can successfully be

deployed with voice quality at least as good at the PSTN• IP-PBXs offers both cost and application benefits• Many IP-PBXs provide a total end-to-end “converged”

solution that covers every feature you can think of – UM, screenpop, predictive dialer, ACD, web call-through, chat, presence, and more.

• There are a variety of solutions that allow you to begin small and grow your IP telephony network

• Invest in VoIP monitoring utilities/analyzers• If you decide to DIY when deploying your IP-PBX, make

sure you have the technical know-how to pinpoint network issues that can and will crop up. Don’t leave your most valuable business tool to chance!

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Questions?