Transcript VOIP - iCONS

VOIP
Packet loss, packet labeling and
packet classification
An T. Le
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VoIP Properties
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Real-time stream requested
Typical Internet applications use TCP/IP,
whereas VoIP uses RTP/UDP/IP.
In VoIP, voice is sending over IP network in
IP packets.
Latency
Packet Loss
…There will be no “re-send please” in VoIP
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OSI 7 layers model
Data flow
AH
DATA
PH AH
DATA
SH PH AH
DATA
TH SH PH AH
DATA
Network
NH TH SH PH AH
DATA
Data link
DH NH TH SH PH AH
DATA
Application
Presentation
Session
Transport
Physical
FH DH NH
TH SH PH AH
DATA
OSI model - Data flow
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OSI
Internet Suite
Application
Application
Presentation
Session
Transport
Transport
Network
Internet
Data link
Host - to - Network
Physical
OSI vs Internet
3
IP packets
Bit:
0
4
Version
8
IHL
16
Type of Service
20 octets
Indentification
Time to live
19
32
Total lenght
Flag
Protocol
Fragment offset
64
Header checksum
96
Source address
128
Destination address
160
Options + Padding
IPv4 Header
Bit:
0
4
Version
12
16
24
Traffic class
Payload lenght
32
Flow label
Next Header
Hop limit
64
96
40 octets
128
Source address
160
192
224
256
Destination address
288
320
IPv6 Header
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VoIP packet size
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VoIP packet size is very important.
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The packet size relates to the delay times that needs for sending
and receiving packet
 The lost of packets is impacting to the quality of reconstructed
voice stream.
S = VoIP size = Playload + RTP header + UDP/IP header
S = s+12bytes+28bytes
S = s+40bytes = s + 320bits
Headers size is large, payload size (s) is expected small enough to
reduce delay time
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VoIP bandwidth request
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R=CODEC rate (bps)
n= number of packet in second
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n=R/s where s is a size of payload in second
If s=R then n=1 (badly delay)
H= Header size
BW= n(R/n+H) = R(1+H/s)
Examples:
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If R=64kbps, s=1.28kbps (n=50 or length of payload is
20ms) then R=(64k+40*8*50)=80kbps
If R=64kbps, s=0.64kbps (n=100 or length of payload is
10ms) then R=(64k+40*8*100)=96kbps
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VoIP – End to End Stream
A/D
Coding
(Compressing)
Packetize
IP
Network
D/A
UnPacketize
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DeCoding
(DeCompressing)
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VoIP – End to End Stream with
delay
A/D
Coding
(Compressing)
Packetize
IP
Network
D/A
UnPacketize
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DeCoding
(DeCompressing)
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VoIP’s QoS
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Latency (end to end)
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CODE and DECODE processing delays
Completed (Header + payload) transfer delays
Application delays
Propagation delays
…
Packet loss
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Lost in transmission
Lost in congestion (jitter)
…Lost a bit in header - lost a packet
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Packet loss Ratio and QoS
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Packet loss vs MOS (Mean of Opinion Score)
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(source: http://www.kineto.com )
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Paket loss Ratio and QoS:
Reconstruction and examples
Reconstruction:
1
2
1
3
4
5
6
2
4
5
1
2
4
1
2
4
2
7
8
9
Send
6
8
9
Received
5
6
8
9
Silent insertion
5
6
8
9
Repeat last packet
6
If packet is large, interleaving is may not used due to real time
is requested
Samples: http://www.voiptroubleshooter.com
Silence
Insertion
Replay last
packet
G.711
Appendix 1
5% loss rate
10% loss rate
20% loss rate
40% loss rate
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Packet loss analysis
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Lost by delay:
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As a real-time system, a long delay packet
(>500ms) is considered as loss.
Lost by bad receiving:
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Uncompleted header packet, consider as loss
Uncompleted payload packet, consider as loss if
using any LPC codec
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Loss caused by delay
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Congestion will cause delay then loss
Quality of transmission link:
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Capacity=BW.log2(1+SNR)
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SNR is not independently with BW
Improve SNR by repeater with amplifier
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Optimal Packetization in VoIP
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Optimal packet’s size to reduce end to end delay
Optimal packet’s size to minimum loss ratio (usually
that makes packet smaller)
Three main (but not independently) parameters
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Bandwidth budget
 Min = 8000+320/0.1 = 11.2kbps (with G.729)
 Avg = 64000+320/0.02 = 80kbps (with G.711)
Delay threshold (max = 240ms ?)
MOS threshold (min=3 ?)
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VoIP – Bandwidth request
BW (bps)
BW = R + H/l
BW: Minimum bandwidth request
R : CODEC bitrate
H: VoIP header (320 bits)
R
l (sec)
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R-factor vs MOS
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An IP phone monitoring application using the SNMP
The R-factor is described in the ITU-T G.107 recommendation which defines a
computing model known as an E-model. The R-factor is a well-tried tool for
transmission planning and for determining the combined impact of various
transmission parameters which influence the call quality. All appropriate transmission
parameters are put together to calculate the R-factor as follows:
R = RO - IS - ID - IE-EFF + A
where
 RO is the basic signal-to-noise ratio,
 IS
is a sum of all impairments occurring during speech transmission,
 ID
is a degradation factor representing all impairments caused by the
voice signal delay,
 IE-EFF includes packet loss,
 A
is an advantage factor (permitted range is from 0 to 20)
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Acceptable MOS and R Scores for
Narrowband CODECs
(source: voicetroubeshooter.com)
User Opinion
R Factor
MOS Score
Maximum obtainable for G.711
93
4.4
Very satisfied
90-100
4.3-5.0
Satisfied
80-90
4.0-4.3
Some users satisfied
70-80
3.6-4.0
Many users dissatisfied
60-70
3.1-3.6
Nearly all users dissatisfied
50-60
2.6-3.1
Not recommended
0 - 50
1.0-2.6
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Packet labeling
Packet classification
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Concern about Class of Service (CoS)
Fast detect Packet class
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Give a label to packet
Use Packet classification
Packet labeling can be done by HW or SW
Packet classification, usually, done by HW
(Programmable Logic Controller)
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Packet labeling
Packet classification (cont)
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Bit 8-15 in IP header:
TOS: Type of Service
00
01
02
Precedence
03
04
05
06
07
D
T
R
M
0
Precedence. 3 bits.
D. 1 bit. Minimize delay.
Value
Value
Description
Value
Description
Description
T. 1 bit. Maximize throughput.
0
Routine.
0
Normal delay.
0
Normal throughput.
1
Priority.
1
Low delay.
1
High throughput.
2
Immediate.
3
Flash.
4
Flash override.
5
CRITIC/ECP.
6
7
R. 1 bit. Maximize reliability.
M. 1 bit. Minimize monetary cost.
Value
Description
Value
Internetwork control.
0
Normal reliability.
0
Normal monetary cost.
Network control.
1
High reliability.
1
Minimize monetary cost.
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Description
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Packet labeling
Packet classification (cont)
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RTP header, the use of PT
(Payload Type) bits (9-15)
PT
Name
Type
Clock rate
(Hz)
Audio
channels
References
0
PCMU
Audio
8000
1
RFC 3551
3
GSM
Audio
8000
1
RFC 3551
7
LPC
Audio
8000
1
RFC 3551
8
PCMA
Audio
8000
1
RFC 3551
12
QCELP
Audio
8000
1
18
G729
Audio
8000
1
19
reserved
Audio
20- 24
27
29- 30
35-71
77-95
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Packet labeling
Packet classification (cont)
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Use of UDP header
Source Port
Destination Port
Length
Checksum
Data:::
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Packet labeling
Packet classification (cont)
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Use of Payload ?
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Conclusion
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Packet loss
 by processing
 by end to end delay
 by congestion
 by bad transceiver and link
Solutions for packet loss
 Use “good" CODEC
 Optimal packet (based on Bandwidth, CODEC, desire of delay
and MOS
 Need of Real time QoS monitor and adaptive variable packet size
protocol.
Packet labeling and classification
 Fast packet class detection
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Reference
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1- Voice over Internet protocol (VoIP)
Goode, B.;
Proceedings of the IEEE
Volume 90, Issue 9, Sept. 2002 Page(s):1495 - 1517
Performance comparison between VBR speech coders for adaptive VoIP applications
Beritelli, F.; Casale, S.; Ruggeri, G.;
Communications Letters, IEEE Volume 5, Issue 10, Oct. 2001 Page(s):423 - 425
Congestion Avoidance Using DYnamic COdec MAnagement: A solution for ISP
Alcuri, L.; Saitta, F.; Fasciana, M.L.;
Communications, 2005 Asia-Pacific Conference on 03-05 Oct. 2005 Page(s):886 - 890
On packet loss concealment artifacts and their implications for packet labeling in voice over IP
Praestholm, S.; Jensen, S.S.; Andersen, S.V.; Murthi, M.N.;
Multimedia and Expo, 2004. ICME '04. 2004 IEEE International Conference on Volume 3, 27-30 June
2004 Page(s):1667 - 1670 Vol.3
Assessment of effects of packet loss on speech quality in VoIP
Ding, L.; Goubran, R.A.;
Haptic, Audio and Visual Environments and Their Applications, 2003. HAVE 2003. Proceedings. The 2nd
IEEE Internatioal Workshop on 20-21 Sept. 2003 Page(s):49 - 54
Voice-quality monitoring and control for VoIP
Manousos, M.; Apostolacos, S.; Grammatikakis, I.; Mexis, D.; Kagklis, D.; Sykas, E.;
Internet Computing, IEEE, Volume 9, Issue 4, July-Aug. 2005 Page(s):35 - 42
Digital Object Identifier 10.1109/MIC.2005.92
…….
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Others listed in this presentation
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THANK YOU
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