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VoIP Testing With TEMS Investigation
VoIP Testing With TEMS Investigation
The interarrival jitter is calculated continuously as each data packet is
received [...] using this difference for that packet and the previous
packet in order of arrival (not necessarily in sequence), according
to the formula
The quantity is what is output in the VoIP RFC 1889 Jitter information
element. The latter is updated once every second.
3.5.1.2 Jitter Buffer
A jitter buffer is used to mitigate the effects of packet jitter. The jitter buffer
holds the received voice packets briefly, reorders them if necessary, and
then plays them out at evenly spaced intervals to the decoder.
These elements are updated once every second.
Ascom (2012) Document:
NT11-12850 9(18)
VoIP Decoding Errors (%) Percentage of audio frames that could not be
decoded by the speech codec.
VoIP Jitter Buffer Lost
Packets (%)
Percentage of packets that were missing from the
audio reproduction because they were not
delivered from the jitter buffer to the decoder in
timely fashion.
Note that the packet need not have been lost on
the way to the receiving party; it may just have
been delayed too long, so that it was discarded by
the jitter buffer.
VoIP Jitter Buffer Playout
Delay Average (ms)
Average playout delay in ms: that is, the average
time the voice packets were held by the jitter
buffer.
VoIP Jitter Buffer Playout
Delay Maximum (ms)
Maximum playout delay in ms.
VoIP Jitter Buffer Playout
Delay Minimum (ms)
Minimum playout delay in ms.
VoIP Jitter Buffer Size
Increase (%)
Percentage of audio frames where the VoIP client
decided to increase the jitter buffer size (because
the jitter was found to be too high). This
procedure results in a period of silence in the
audio reproduction as the jitter buffer accumulates
packets without releasing any.
VoIP Jitter Buffer
Overruns (%)
Percentage of audio frames with overruns.
The VoIP client tries to keep the delays caused by
the jitter buffer reasonably low. When the buffer
becomes too long, the VoIP client will throw away
received packets to decrease the buffer size. This
is referred to as overruns, and it affects the audio
reproduction.
Usually occurs after underruns (see below).
VoIP Jitter Buffer
Underruns (%)
Percentage of audio frames where the jitter buffer
was empty and had no packets to deliver to the
speech decoder.
Ascom (2012) Document:
NT11-12850 10(18)
3.5.1.3 Audio Quality Related
Data category
VoIP FER Combined
Packet Loss (%)
Total percentage of packet loss that affects the
reproduction of the audio. Encompasses decoding
errors, underruns, overruns, and jitter buffer size
increases: compare the information elements in
section 3.5.1.2. Should in general correlate
closely to PESQ and POLQA.
VoIP Speech Codec Speech codec selected for the VoIP client in the
governing script (VoIP Dial and VoIP Answer
activities: see section 3.2, steps O and O).
Media Quality category (see also section 3.4):
PESQ Score Downlink PESQ (ITU P.862.1) voice quality score.
For VoIP measurements the speech sentences
are 5.5 s in length. This means that a MOS score
will be calculated every 11 s (since transmissions
are done in semi-duplex). Note that a
performance degradation that occurs while the
measurement is done at the other end will not
register in the PESQ score.
POLQA NB Score Downlink POLQA (ITU P.863.1) voice quality score for
narrowband.
POLQA SWB Score
Downlink
POLQA voice quality score for super-wideband.
In the real time presentation, the PESQ and POLQA scores appear the
moment they have been computed. When loading a logfile for analysis, on
the other hand, the PESQ and POLQA scores are moved backward in time
to the point when the corresponding speech sentence was received by the
VoIP client. That is, sentences are aligned in time with their quality scores.
This is not much of an issue for PESQ, which takes only a fraction of a
second to compute, but it can be for POLQA, whose computation may
require several seconds (the worse the degradation of the signal, the more
complex POLQA is to evaluate).
3.5.2 Other Information Elements of Interest
Application throughput IEs (Data category).
RAN throughput at various protocol levels (details being dependent on
the cellular technology used; the IEs are found in the relevant category,
such as LTE, WCDMA).
Ascom (2012) Document:
NT11-12850 11(18)
3.5.3 VoIP Events
These events underlie the KPIs in section 3.5.4:
MTSI Registration Failure One of the parties failed to register with the SIP
server.
MTSI Registration Time Time required for the terminal to register with the
SIP server. Also functions as a success event.
MTSI Session Completion
Failure
A VoIP session that was successfully set up failed
to complete. Similar to dropped call for CS voice.
MTSI Session Completion
Time
Duration of the VoIP session. Also functions as a
success event.
Note: This event does not have an associated
KPI, since the VoIP session duration is not a
relevant performance measure.
MTSI Session Setup
Failure
The terminal failed in setting up a VoIP session.
Similar to blocked call for CS voice.
MTSI Session Setup Time Time required to set up the VoIP session. Also
functions as a success event.
TEMS Investigation also generates the following VoIP events, which are
unrelated to KPI computation:
VoIP Start A VoIP session was started.
VoIP End A VoIP session ended normally.
VoIP Error A VoIP session was aborted because of an error.
3.5.4 VoIP KPIs (Key Performance Indicators)
TEMS Investigation provides data for computation of the following KPIs.
The actual computation is done in TEMS Discovery or TEMS Automatic.
MTSI Registration Failure
Ratio (%)
Denotes the probability that the terminal cannot
register towards IMS when requested.
MTSI Registration Time (s) Denotes the time elapsing from the IMS
registration request until the terminal is registered
to IMS.
MTSI Session Setup
Failure Ratio (%)
Denotes the probability that the terminal cannot
set up an MTSI session. An MTSI session setup is
initiated when the user presses the call button
and concludes when the user receives, within a
predetermined time, a notification that the callee
has answered.
Ascom (2012) Document:
NT11-12850 12(18)
MTSI Session Setup Time
(s)
Denotes the time elapsing from initiation of an
MTSI session until a notification is received that
the session has been set up.
MTSI Session Completion
Failure Ratio (%)
Denotes the probability that a successfully set up
MTSI call is ended by a cause other than
intentional termination by either party.
3.6 Presentation in TEMS Investigation Windows
Suitable presentation windows for VoIP data:
VoIP Quality status window containing the information elements
described in section 3.5.1
VoIP Quality Line Chart tracking VoIP PESQ/POLQA NB/POLQA
SWB scores and VoIP FER Combined Packet Loss, and indicating
MTSI events
VoIP AMR Codecs Usage status window
Data Reports message window
IP Protocol Reports message window.
3.7 Ascom Test Setup
The VoIP function in TEMS Investigation has been tested with TekSIP, a
SIP Registrar and SIP Proxy for Windows (www.teksip.com), as well as
with an Ericsson IMS server.
4 Troubleshooting
4.1 Problem: Script Activity Fails
Check that caller and callee are synchronized, that is, that the callee
reaches VoIP Answer before the caller begins VoIP Dial. See
section 3.2, step 3, and section 3.2.1.
In the Events window, look for MTSI failure events.
In the Data Reports message window, look into the VoIP Error
Message category.
In the IP Protocol Reports message window, study the SIP messages.
If any other ports than SIP port 5060 and RTP port 4000 are used on
the SIP server, the corresponding settings have to be changed in the
file <TEMS Investigation install dir>\Application\Configuration\
Investigation.Voip.config.
If SIP response code 422 (Session interval too small) is received, set
DisableTimers="false" in the same file.
Ascom (2012) Document:
NT11-12850 13(18)
4.2 Problem: Bad Audio Quality (PESQ/POLQA
Score Low)
Investigate throughput and BLER values at different levels. Example:
For LTE, this includes the application layer, PDSCH, RLC, PDCP, and
MAC. Remember to look at both uplink and downlink.
Check channel quality indicators and serving/neighbor signal strength.
Example: In an LTE network, study CQI, Serving Cell RSRP, and
Neighbor Cell RSRP.
Check for excessively frequent handovers.
In the IP Protocol Reports message window, look into the RTP
messages.
Ascom (2012) Document:
NT11-12850 14(18)
5 Limitations
You cannot have any other internet connections in parallel while running
VoIP measurements. That is, the PCs cannot be connected to any
further IP addresses, whether through other external devices, through
an Ethernet cable, or by other means. All network interfaces except
the testing devices, both fixed and wireless, must be disabled. It is
however possible to make CS voice calls with devices connected to the
PCs.
When running a script for the first time with CounterPath, a pop-up will
appear warning about firewall configuration. The pop-up must be
acknowledged; however, this will also cause the first VoIP session to fail,
and the Service Control script must be restarted manually.
With CounterPath, if a failure event occurs for some activity in the script,
the script will terminate and must be restarted manually.
Ascom (2012) Document:
NT11-12850 15(18)
6 Appendices
6.1 SIP Response Codes
6.1.1 Informational Responses
100 Trying
180 Ringing
181 Call is being forwarded
182 Queued
183 Session progress
6.1.2 Successful Responses
200 OK
202 Accepted
Indicates that the request has been understood but actually cannot
be processed
6.1.3 Redirection Responses
300 Multiple choices
301 Moved permanently
302 Moved temporarily
305 Use proxy
380 Alternative service
6.1.4 Client Failure Responses
400 Bad request
401 Unauthorized
Used only by registrars or user agents. Proxies should use proxy
authorization 407
402 Payment required
Reserved for future use
403 Forbidden
404 Not found
User not found
405 Method not allowed
406 Not acceptable
407 Proxy authentication required
Ascom (2012) Document:
NT11-12850 16(18)
408 Request timeout
Could not find the user in time
409 Conflict
410 Gone
The user existed once, but is no longer available here
412 Conditional request failed
413 Request entity too large
414 Request URI too long
415 Unsupported media type
416 Unsupported URI scheme
417 Unknown resource priority
420 Bad extension
Bad SIP protocol extension used, not understood by the server
421 Extension required
422 Session interval too small
423 Interval too brief
424 Bad location information
428 Use identity header
429 Provide referrer identity
433 Anonymity disallowed
436 Bad identity info
437 Unsupported certificate
438 Invalid identity header
480 Temporarily unavailable
481 Call/transaction does not exist
482 Loop detected
483 Too many hops
484 Address incomplete
485 Ambiguous
486 Busy here
487 Request terminated
488 Not acceptable here
489 Bad event
491 Request pending
493 Undecipherable
Could not decrypt S/MIME body part
494 Security agreement required
Ascom (2012) Document:
NT11-12850 17(18)
6.1.5 Server Failure Responses
500 Server internal error
501 Not implemented
The SIP request method is not implemented here
502 Bad gateway
503 Service unavailable
504 Server timeout
505 Version not supported
The server does not support this version of the SIP protocol
513 Message too large
580 Precondition failure
6.1.6 Global Failure Responses
600 Busy everywhere
603 Decline
604 Does not exist anywhere
606 Not acceptable
6.1.7 Extended Codes
701 The called party has hung up
702 VoIP socket error
703 Connection cancelled because of timeout
704 Connection interrupted because of a SIP error
705 SIP memory error
706 SIP transaction memory error
751 Busy tone: No codec match between the calling and called
party
810 General socket layer error
811 General socket layer error: Wrong socket number
812 General socket layer error: Socket is not connected
813 General socket layer error: Memory error
814 General socket layer error: Socket not available check IP
settings/connection problem/VoIP setting incorrect
815 General socket layer error: Illegal application on the socket
interface
922 No DNS server known
923 DNS name resolution failed
Ascom (2012) Document:
NT11-12850 18(18)
924 Insufficient resources for DNS name resolution
925 URL error
6.2 Abbreviations
AMR-NB Adaptive Multi Rate Narrowband
AMR-WB Adaptive Multi Rate Wideband
BLER Block Error Rate
CQI Channel Quality Indicator
FER Frame Erasure Rate
IMS IP Multimedia Subsystem
IP Internet Protocol
KPI Key Performance Indicator
LTE Long Term Evolution
MAC Medium Access Control
MOS Mean Opinion Score
MTSI Multimedia Telephony Service for IMS
PDSCH Physical Downlink Shared Channel
PESQ Perceptual Evaluation of Speech Quality
POLQA Perceptual Objective Listening Quality Assessment
PSTN Public Switched Telephone Network
RAN Radio Access Network
RSRP Reference Signal Received Power
RTP Real-time Transport Protocol
SIP Session Initiation Protocol
VoIP Voice over IP