Quality and Performance Evaluation of VoIP End-points

1
Quality and Performance Evaluation of VoIP
End-points

• Wenyu Jiang
• Henning Schulzrinne
• Columbia University
• NYMAN 2002
• September 3, 2002

2
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

3
Performance Metrics for VoIP End-points

• Mouth-to-ear (M2E) delay
• C.f. network delay
• Clock skew
• Whether it causes any voice glitches
• Amount of clock drift
• Silence suppression behavior
• 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

4
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

5
VoIP End-points Tested

• Hardware End-points
• Cisco, 3Com and Pingtel IP phones
• Mediatrix 1-line SIP/PSTN Gateway
• Software clients
• Microsoft Messenger, NetMeeting (Win2K, WinXP)
• Net2Phone (NT, Win2K, Win98)
• Sipc/RAT (Solaris, Ultra-10)
• Robust Audio Tool (RAT) from UCL as media client
• Operating parameters
• In most cases, codec is G.711 ?-law, packet
  interval is 20ms

6
Example M2E Delay Plot

• 3Com to Cisco, shown with gaps gt 1sec
• Delay adjustments correlate with gaps, despite
  3Com phone has no silence suppression

7
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)

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Snapshot 2

• M2E delay drops to 49ms, at time of 416

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Snapshot 3

• Presence of a gap during the delay change

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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

11
Sender Characteristics

• Certain senders may introduce delay spikes,
  despite operating on a LAN

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Average M2E Delays for IP phones and sipc

• Averaging the M2E delay allows more compact
  presentation of end-point behaviors
• Receiver (especially RAT) plays an important role
  in M2E delay

13
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 (gt 400ms)
• Messengers 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) MgrXP (notebook) 96.8ms 68.5ms
NM2K (pc) NM2K (notebook) 401ms 421ms
Mobile (GSM) PSTN (local number) 115ms 109ms
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Delay Behaviors for PC Clients, contd.

• Net2Phones delay is also high
• 200-500ms
• V1.5 reduces PC-gtPSTN 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
N2P v1.5 NT P-2 (pc2) PSTN (local number) 201ms 373ms
N2P v1.5 W2K K7 (pc) PSTN (local number) 196ms 401ms
N2P v1.5 W2K K7 (pc) N2P v1.5 W98 P-3 (notebook2) 525ms 350ms
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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?

16
Clock Skew Rates

• Mostly symmetric between two devices
• RAT (Ultra-10) has unusually high drift rates, gt
  300 ppm (parts per million)
• High clock skews confirmed in many (but not all)
  PCs and workstations

Drift Rates (in ppm) 3Com Cisco Mediatrix Pingtel RAT
3Com -8.3 55.4 43.3 41.2 -333
Cisco -55.2 -0.4 -11.8 -12.1 -381
Mediatrix -43.1 11.7 1.3 -0.8
Pingtel -40.9 12.7 2.8 -3.5 -380
RAT 343 403 376 12.3
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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) MgrXP (notebook) 172 87.7
Mgr2K (pc) MgrXP (notebook) 165 85.6
NM2K (pc) NM2K (notebook) ? -33?
Net2Phone NT (pc2) PSTN 290 -287
Net2Phone 2K (pc) PSTN 166 82
Mobile (GSM) PSTN 0 0
18
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.

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PLC Behaviors

• Loss tolerance (at 20ms interval)
• By measuring loss-induced gaps in output audio
• 3Com and Pingtel phones 2 packet losses
• Cisco phone 3 packet losses
• Level of audio distortion by packet loss
• 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

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Effect of PLC on Delay

• No affirmative effect on M2E delay
• E.g., sipc to Pingtel

21
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

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Hangover Time

• Measured by feeding ON-OFF waveforms and monitor
  RTP packets
• Cisco phones 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

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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

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Acoustic Echo Cancellation

• Important for hands-free/conferencing (business)
  applications
• Primary metric Echo Return Loss (ERL)
• Measured by LAN-sniffing RTP packets
• Most IP phones support AEC
• ERL depends slightly on input level and
  speaker-phone volume
• Usually gt 40 dB (good AEC performance)

IP Phone 3Com Cisco ipDialog Pingtel Snom-100
ERL (dB) 40-45 53-? 49-54 33-42 ?-5 (no AEC)
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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

• Initial test
• Used typical cable modem delay traces
• Tested RAT to Cisco
• No audible distortion due to late loss
• Added delay is normal

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M2E Delay under Jitter, contd.

• Cisco phone generally within expectation
• Can follow network delay change timely
• Takes longer (10-20sec) to adapt to decreasing
  delay
• Does not overshoot playout delay
• More end-points to be examined

Artificial Trace
Real Trace with Spikes
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Conclusions

• Average M2E Adelay
• Low (mostly lt 80ms) for hardware IP phones
• Software clients lowest for Messenger 2000
  (68.5ms)
• Application (receiver) most vital in determining
  delay
• Poor implementation easily undoes good network
  QoS
• Clock skew high on SW clients (RAT, Net2Phone)
• Packet loss concealment quality
• Acceptable in all 3 IP phones tested, w. Cisco
  more robust
• Silence detector behavior
• Long hangover time, works well for speech input
• But may falsely predict music as silence
• Acoustic Echo Cancellation good on most IP
  phones
• Playout delay behavior good based on initial
  tests

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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