ensuring qos in your voip development
DESCRIPTION
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 PresentationTRANSCRIPT
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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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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
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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.
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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
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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
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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
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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
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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)
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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.
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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
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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.
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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.
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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
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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.
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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
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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.
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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
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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.
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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
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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.
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Problem isolation procedure
Central QoS server
RTCP
Packet monitoring
SNMP
ALARM
Console
VoIP Network
1
2 3
4 56
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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.
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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.
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Q & A
Thank you!
Any questions?