ensuring qos in your voip development

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Ensuring QoS in Your VoIP Development Choon Shim CTO and Senior VP of Engineering [email protected] , http://www.qovia.com

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Ensuring QoS in Your VoIP Development. Choon Shim CTO and Senior VP of Engineering [email protected] , http://www.qovia.com. VoIP problems. Outage: - Infrastructure: switch, router, bridge, UPS, etc - VoIP element: call server, SIP server, GW, GK, MCU, handsets. - PowerPoint PPT Presentation

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Page 1: Ensuring QoS in Your VoIP Development

Ensuring QoS in Your VoIP Development

Choon Shim

CTO and Senior VP of Engineering

[email protected] , http://www.qovia.com

Page 2: Ensuring QoS in Your VoIP Development

VoIP problems

Outage:- Infrastructure: switch, router, bridge, UPS, etc- VoIP element: call server, SIP server, GW, GK, MCU, handsets.- Carrier: T1/E1, analog signal trunk lines.

Voice Quality:- Delay: network bandwidth, processing power- Echo: hybrid, acoustic- Jitter: jitter buffer calculation, variable delay- Packet loss: sender base, receiver base- Out of order: complex topology

Page 3: Ensuring QoS in Your VoIP Development

Roots of the problems

IP is not designed for carrying real-time media stream.

Management was not considered by System/Elements Vendors.

Too many moving parts. Too many protocol layers. Too many API layers. Multi vendor products.

Page 4: Ensuring QoS in Your VoIP Development

VoIP ready network

Fast network: low latency and jitter Clean network: few packet loss and retransmit QoS ready network: Voice packet has priority Fault tolerant network: redundancy and backup Manageable network: monitoring and

management

Page 5: Ensuring QoS in Your VoIP Development

Bandwidth

Required bandwidth per call (bps):BW = (V + I + L) * 8 * PWhere, V is size of voice sample,

I is IP/UDP/RTP overhead, L is data link overhead and P is packets generated per second.Example) (160 + 40 + 18) * 8 * 50 = 87.2 kbps

Required bandwidth total:Total BW = BW * NWhere, N is total number of simultaneous callsExample) 87.2 * 50 = 4.36 Mbps

=> Increase bandwidth

Page 6: Ensuring QoS in Your VoIP Development

To reduce bandwidth requirement

Bandwidth requirement by codec and duplex

cRTP reduces 2-5 bytes overhead VAD reduces up to 50% payload

Link Type

(Sample time)

G.711

(10 ms)

G.711

(20 ms)

G.729

(20 ms)

Half duplex 220.8 174.4 78.4

Full duplex 110.4 87.2 29.4

Page 7: Ensuring QoS in Your VoIP Development

Clean network

Reduce hop counts Reduce complexity of network topology Remove duplex mismatch Remove black hole and loop Avoid half duplex link Use common sense for cabling

Page 8: Ensuring QoS in Your VoIP Development

QoS ready network

Layer 3:- Type of Service (TOS)- RSVP signaling (RFC 2205)- DiffServ (RFC2474)- Multiprotocol Label Switching (MPLS)

Layer 2:- 802.1p and 802.1q- Ethernet Class of Service (COS)

Page 9: Ensuring QoS in Your VoIP Development

Fault tolerant network -Outage detection

Carrier failure: T1, E1, Analog- No incoming or outgoing calls.- Checking the module LED.- Checking the event log, management console.- Running a loop back test for T1/E1.- Checking with T1 tester.- Receiving an alarm from the call server.

Page 10: Ensuring QoS in Your VoIP Development

Fault tolerant network –Outage detection (cont)

Infrastructure failure:- No dial tone or bad voice quality- Checking NMS console- Checking SNMP Traps- Testing cables- Testing switches, routers, bridges, etc- Checking UPS power load, power level, connection

Page 11: Ensuring QoS in Your VoIP Development

Fault tolerant network – Outage detection (cont)

VoIP element failure:

- No dial tone.

- Checking SNMP trap.

- Checking NMS console.

- Checking with the vendor management console.

- Checking event log, trace, etc.

Page 12: Ensuring QoS in Your VoIP Development

Outage detection issues

Lack of alarm implementation. Too many consoles to monitor: NMS, vendor

supplied management, third party software, carrier OSS/EOSS.

Too many elements could go wrong. Carriers are not monitoring the CSU or CPE.

Page 13: Ensuring QoS in Your VoIP Development

Alarm – Event driven

Switch

Router

T1/E1

UPS

GK

GW

TE

Bridge

Analog

Environ

Server

Management

Server

SNMP Trap

Email/Pager

Carrier console

VoIPm Console

Page 14: Ensuring QoS in Your VoIP Development

Checking vital signs

Blind polling: send a ping to every elements every x minutes. It triggers extra network traffics. Total number of packets per hour N = e * x / 60, where e = number of elements, x is minutes.

Severity base polling: send a ping to critical elements more often. For example) GW: every 5 mins, GK: every 6 mins, Switch: every 10 mins, Terminal element: 30 mins, etc.

Dynamic polling: recalculates number of pings based on the previous faults, traffic or volume. Number of packets N = f(1)..f(e), where f is the function being used for calculating faults, traffic and volume.

Page 15: Ensuring QoS in Your VoIP Development

Manageable VoIP Network –Voice quality measurement

MOS (Mean Opinion Score): - Subjective measurement of VoIP. - Pre selected voice sample over different media, replayed to mixed group of men and woman, who rate them from 1 to 5.4 – 5: Toll Quality3 – 4: Communication quality< 3 : Synthetic quality

Page 16: Ensuring QoS in Your VoIP Development

Voice quality measurement

PSQM (Perceptual Speech Quality Measurement, ITU-T P.861): - Automated scoring process using an algorithm that enables computer-derived scores to correlate to MOS scores. - Designed for circuit-switched network and does not take into effect important parameters such as jitter and packet loss, which affect voice quality on a VOIP network adversely.

Page 17: Ensuring QoS in Your VoIP Development

Voice quality measurement

PAMS (Perceptual Analysis and Measurement System):- Designed an intrusive listening speech quality assessment tool where speech quality is computed by injecting a speech like signal at one end and analysing the degraded signal at other end of the network.

Parameter

SCORE

1 2 3 4 5

Latency (ms) <50 50-75 75-100 100-200

>200

Packet/Loss (%) 0 0-1 1-2 2-3 >3

Jitter (ms) <5 5-10 10-50 50-100 >100

Page 18: Ensuring QoS in Your VoIP Development

Voice quality measurement

PESQ: PESQ (ITU–T P.862):- The latest standard for assessing voice quality and is expected to eventually replace PSQM. - It builds on the PSQM and PAMS algorithms by adding additional processing steps to account for signal-level differences and the identification of errors associated with packet loss.

Page 19: Ensuring QoS in Your VoIP Development

Voice quality measurement

Delay guideline: ITU-T G.114

Acceptable

Acceptable under conditions

Unacceptable0 150 40

00 – 150 ms: Good quality and no echo

151 – 400 ms: Acceptable under certain conditions and echo canceling is needed

401+: Unacceptable for real-time voice traffic and planning and testing purposes only

Page 20: Ensuring QoS in Your VoIP Development

Quality problem detection

Interpret RTCP and RTCP XR. Packet monitoring by Layer2 switch taping or

port mirroring. Probing and active monitoring by injecting a

test packet. SNMP, RMON or sFlow gathering.

Page 21: Ensuring QoS in Your VoIP Development

Problem isolation procedure

Central QoS server

RTCP

Packet monitoring

SNMP

ALARM

Console

VoIP Network

1

2 3

4 56

Page 22: Ensuring QoS in Your VoIP Development

Central QoS management server

Discover VoIP components/elements dynamically. Create a topology and aggregate multiple call servers, GW, GK,

MCU, SIP Servers, etc. Collect performance/delay data from various sources. Calculate variable polling period and injects an active packet. Make a statistical model to use for assign QoS. Organize elements/QoS data in the relational DBMS. Detect voice quality problem and send an alarm to console. Inject an active test packet to isolate the problem as per

console.

Page 23: Ensuring QoS in Your VoIP Development

Console

Display overall call quality. Display topology and status display. Display drill down to detail elements with

MOS/PESQ. Display real time status and quality changes. Trigger the problem isolation procedure.

Page 24: Ensuring QoS in Your VoIP Development

Q & A

Thank you!

Any questions?