Quality and Performance
Evaluation of VoIP End-points
Wenyu Jiang
Henning Schulzrinne
Columbia University
NYMAN 2002
September 3, 2002
Motivations


The quality of VoIP depends on both
the network and the end-points
Extensive QoS literature on network
performance, e.g., IntServ, DiffServ


Focus is on limiting network loss & delay
Little is known about the behavior of
VoIP end-points
Performance Metrics for VoIP
End-points

Mouth-to-ear (M2E) delay


Clock skew



Whether it causes any voice glitches
Amount of clock drift
Silence suppression behavior



C.f. network delay
Whether the voice is clipped (depends much on
hangover time)
Robustness to non-speech input, e.g., music
Robustness to packet loss

Voice quality under packet loss
Measurement Approach



Capture both original and output audio
Use “adelay” program to measure M2E delay
Assume a LAN environment by default

Serve as a baseline of reference, or lower bound
stereo
signal
PC
line in
notebook
speaker
original
audio
(mouth)
coupler
coupler
IP phone
output
audio
In
Out
ethernet
(ear)
LAN
IP phone
In
Out
ethernet
VoIP End-points Tested

Hardware End-points



Software clients




Cisco, 3Com and Pingtel IP phones
Mediatrix 1-line SIP/PSTN Gateway
Microsoft Messenger, NetMeeting (Win2K, WinXP)
Net2Phone (NT, Win2K, Win98)
Sipc (Solaris, Ultra-10)
Operating parameters:

In most cases, codec is G.711 -law, packet
interval is 20ms
Example M2E Delay Plot

3Com to Cisco, shown with gaps > 1sec
Delay adjustments correlate with gaps,
despite 3Com phone has no silence
suppression 60 experiment
1-1
experiment 1-2
silence gaps
M2E delay (ms)

55
50
45
40
35
0
50
100
150 200
time (sec)
250
300
350
Visual Illustration of M2E
Delay Drop, Snapshot #1



3Com to Cisco
1-1 case
Left/upper
channel is
original audio
Highlighted
section shows
M2E delay
(59ms)
Snapshot #2

M2E delay
drops to
49ms, at
time of
4:16
Snapshot #3

Presence of
a gap during
the delay
change
Effect of RTP M-bits on Delay
Adjustments
Cisco phone sends M-bits, whereas Pingtel
phone does not

Presence of M-bits results in more adjustments
100
Cisco to 3Com 1-1
Pingtel to 3Com 2-1
new talkspurt (M-bit=1)
90
M2E delay (ms)

80
70
60
50
40
30
20
0
50
100
150 200
time (sec)
250
300
Sender Characteristics
Certain senders may introduce delay
spikes, despite operating on a LAN
300
Mediatrix to 3Com 3-1
Mediatrix to Cisco 1-1
Mediatrix to Pingtel 1-1
250
M2E delay (ms)

200
150
100
50
0
50
100
150 200
time (sec)
250
300
Average M2E Delays for IP
phones and sipc

Averaging the M2E delay allows more compact
presentation of end-point behaviors
Receiver (especially sipc) plays an important role in
M2E delay
250
M2E delay (ms)

200
150
100
50
0
Receiver
3Com
Cisco
Mediatrix
Pingtel
sipc
Cisco G.729
Average M2E Delays for PC
Software Clients

Messenger 2000 wins the day




Its delay as receiver (68ms) is even lower than Messenger
XP, on the same hardware
It also results in slightly lower delay as sender
NetMeeting is a lot worse (> 400ms)
Messenger’s delay performance is similar to or better
than a GSM mobile phone.
A
B
AB
BA
MgrXP (pc)
MgrXP (notebook)
109ms
120ms
Mgr2K (pc)
NM2K (pc)
96.8ms 68.5ms
NM2K (notebook)
Mobile (GSM) PSTN (local number)
401ms
421ms
115ms
109ms
Delay Behaviors for PC Clients,
contd.

Net2Phone’s delay is also high



~200-500ms
V1.5 reduces PC->PSTN delay
PC-to-PC calls have fairly high delays
A
B
AB
BA
N2P v1.1 NT P-2 (pc2)
PSTN
(local number)
292ms
372ms
201ms
373ms
N2P v1.5 W2K K7 (pc)
196ms
401ms
N2P v1.5 W2K K7 (pc)
N2P v1.5 W98 P-3 525ms
(notebook2)
350ms
N2P v1.5 NT P-2 (pc2)
Effect of Clock Skew: Cisco to
3Com, Experiment 1-1




Symptom of
playout buffer
underflow
Waveforms
are dropped
Occurred at
point of delay
adjustment
Bugs in
software?
Clock Drift Rates


Mostly symmetric between two devices
Sipc has unusually high drift rates, > 300
ppm (parts per million)
Drift Rates 3Com
(in ppm)
Cisco
Mediatrix Pingtel
sipc
3Com
55.4
43.3
41.2
-333
-11.8
-12.1
-381
Cisco
-55.2
-0.4
Mediatrix
-43.1
11.7
Pingtel
-40.9
12.7
sipc
343
403
-0.8
2.8
-380
376
Drift Rates for PC Clients

Drift Rates not always symmetric!


But appears to be consistent between Messenger
2K/XP and Net2Phone on the same PC
Existence of 2 clocking circuits in sound card?
A
B
AB
BA
MgrXP (pc)
172
87.7
Mgr2K (pc)
MgrXP
(notebook)
165
85.6
NM2K (pc)
NM2K (notebook) ?
-33?
Net2Phone NT (pc2)
PSTN
290
-287
Net2Phone 2K (pc)
166
82
Mobile (GSM)
0
0
Packet Loss Concealment

Common PLC methods





Silence substitution (worst)
Packet repetition, with optional fading
Extrapolation (one-sided)
Interpolation (two-sided), best quality
Use deterministic bursty loss pattern



3/100 means 3 consecutive losses out of every
100 packets
Easier to locate packet losses
Tested 1/100, 3/100, 1/20, 5/100, etc.
PLC Behaviors

Loss tolerance (at 20ms interval)




Level of audio distortion by packet loss



By measuring loss-induced gaps in output audio
3Com and Pingtel phones: 2 packet losses
Cisco phone: 3 packet losses
Inaudible at 1/100 for all 3 phones
Inaudible at 3/100 and 1/20 for Cisco phone, yet
audible to very audible for the other two.
Cisco phone is the most robust

Probably uses interpolation
Effect of PLC on Delay
No affirmative effect on M2E delay

E.g., sipc to Pingtel
80
0/100
3/100
1/20
mouth-to-ear delay (ms)

75
70
65
60
55
50
0
10
20
30
40
time (sec)
50
60
Silence Suppression

Why?




Saves bandwidth
May reduce processing power (e.g., in
conferencing mixer)
Facilitates per-talkspurt delay adjustment
Key parameters


Silence detection threshold
Hangover time, to delay silence suppression and
avoid end clipping of speech

Usually 200ms is long enough [Brady ’68]
Hangover Time



Measured by feeding ON-OFF
waveforms and monitor RTP packets
Cisco phone’s is the longest (2.3-2.36
sec), then Messenger (1.06-1.08 sec),
then NetMeeting (0.56-0.58 sec)
A long hangover time is not necessarily
bad, as it reduces voice clipping


Indeed, no unnatural gaps are found
Does waste a bit more bandwidth
Robustness of Silence
Detectors to Music

On-hold music is often used in
customer support centers




Need to ensure music is played without
any interruption due to silence suppression
Tested with a 2.5-min long soundtrack
Messenger starts to generate many
unwanted gaps at input level of -24dB
Cisco phone is more robust, but still
fails from input level of -41.4dB
M2E Delay under Jitter

Delay properties under the LAN environment
serves as a baseline of reference
When operating over the Internet:



Fixed portion of delay adds to M2E delay as a constant
Variable portion (jitter) has a more complex effect
Preliminary results




Used typical cable modem
delay traces
Tested sipc to Cisco
No audible distortion due
to late loss
Added delay is normal
180
170
160
150
140
130
120
110
100
90
High jitter (uplink)
Low jitter (downlink)
mouth-to-ear delay (ms)

0
20 40 60 80 100 120 140 160 180
time (sec)
Conclusions

Average M2E delay:



Low (mostly < 80ms) for hardware IP phones
Software clients: low (< 120ms) for Messenger, particularly
Messenger 2000 (68.5ms)
The application (receiver) is the most vital in determining delay


Clock drift rates



Are high for some software clients (sipc, Net2Phone)
In some cases non-symmetric!
Packet loss concealment quality


Poorly implemented end-points can easily undo good network QoS
Acceptable in all 3 IP phones tested, w. Cisco being most robust
Silence detectors in Cisco phone, Messenger, NetMeeting


Long hangover time, no audible unwanted gaps for speech input
May falsely predict music as silence
Future Work




Further tests with more end-points on
how jitter influences M2E delay
Measure the sensitivity (threshold) of
various silence detectors
Investigate the non-symmetric clock
drift phenomena
Additional experiments as more brands
of VoIP end-points become available