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Documente Profesional
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PC-based Clients
Technical Paper
© Ascom (2013)
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Contents
1 Introduction ................................................................ 1
4 Troubleshooting....................................................... 11
4.1 Problem: Script Activity Fails ................................................11
4.2 Problem: Poor Audio Quality (PESQ/POLQA Score
Low) .........................................................................................12
5 Limitations ................................................................ 12
6 Appendices............................................................... 13
6.1 SIP Response Codes ..............................................................13
6.1.1 Informational Responses ..........................................................13
6.1.2 Successful Responses..............................................................13
6.1.3 Redirection Responses .............................................................13
6.1.4 Client Failure Responses ..........................................................13
6.1.5 Server Failure Responses ........................................................15
6.1.6 Global Failure Responses.........................................................15
6.1.7 Extended Codes .......................................................................15
6.2 Abbreviations ..........................................................................16
Caller Callee
side side
VoIP
server
VoIP
client Mobile phones VoIP
client
1 Two TEMS Investigation licenses are thus also required, as well as a special
license option for VoIP.
Screenshots of VoIP PC Dial snippet (left) and VoIP PC Answer snippet (right) as
displayed in the TEMS Investigation Service Control Designer. The numbering refers
to the step-by-step description that follows below.
First, both parties need to have an active data session. This is done in
TEMS Investigation through the Network Connect activity.
SIP Registration
Before a VoIP call can begin, both caller and callee must register with the
SIP server to be used for VoIP. The SIP Register activity is used for this
purpose. Here you indicate the IP address or host name of the server to
use. If no special domain needs to be chosen, enter the server address
under Domain, and leave Proxy empty. If on the other hand you need to
specify a domain within the server, enter the server address in the Proxy
field and the domain in the Domain field. You also specify the user and
password the client should use when registering.
Answer
The callee must be ready to answer before the caller can initiate a call. To
this end the callee executes the script activity Answer with Call Type set to
“VoIP PC”. In this activity you select the audio codec and encoding rate the
callee should use. The callee will communicate these settings to the caller,
so that the parties agree on the same codec and rate.
To ensure that the callee has reached Answer before the caller dials the
call, you should insert a wait period in the caller’s script. See section 3.2.1.
This detail has been left out of the above diagram to keep things
straightforward.
Dial
The caller initiates the call by running the activity Dial, again with Call Type
set to “VoIP PC”. In this activity you indicate the codec the caller should
use, which must be the same as the callee’s designated codec (Answer
activity, see step ). You also specify the codec rate and the phone
number to call.
Once the call has been connected, voice quality can be measured on both
sides using one out of several algorithms supported (see section 3.3 for
details on this matter). This is done with the Voice Quality activity, where
Audio Source is set to “VoIP (PC)”. The call durations should preferably
differ between caller and callee, so that it can be controlled which side
hangs up the call. Compare step .
It is possible to store audio files containing the received audio. All audio
sentences having a MOS score lower than or equal to the MOS limit will be
Hangup
One of the parties (the one with the duration of the Voice Quality activity
set lower) hangs up the call. This is to ensure that the party hanging up has
the time to do so before the other party unregisters; otherwise the hangup
will fail. This is done through the Hang Up activity (Call Type = “VoIP PC”).
In the above diagram, the caller performs the hangup.
SIP Unregister
Both sides unregister from the SIP server. This is done using the SIP
Unregister activity.
Network Disconnect
Use of while loops with VoIP scripting: caller (left) and callee (right).
3.4 Output
3.4.1.1 Jitter
Packet jitter or delay variation as defined in IETF RFC 1889, section 6.3.1:
|𝐷(𝑖 − 1, 𝑖)| − 𝐽
𝐽←𝐽+
16
The quantity 𝐽 is what is output in the “VoIP RFC 1889 Jitter” information
element. The latter is updated once every second.
VoIP Decoding Errors (%) Percentage of audio frames that could not be
decoded by the speech codec.
VoIP Jitter Buffer Lost Percentage of packets that were missing from the
Packets (%) 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 Average playout delay in ms: that is, the average
Delay Average (ms) time the voice packets were held by the jitter
buffer.
VoIP Jitter Buffer Size Percentage of audio frames where the VoIP client
Increase (%) 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 Percentage of audio frames where the jitter buffer
Underruns (%) was empty and had no packets to deliver to the
speech decoder.
“Data” category
VoIP FER Combined Total percentage of packet loss that affects the
Packet Loss (%) reproduction of the audio. Encompasses decoding
errors, underruns, overruns, and jitter buffer size
increases: compare the information elements in
section 3.4.1.2. Should in general correlate
closely to PESQ and POLQA.
VoIP Speech Codec Speech codec selected for the VoIP client in the
governing script (Dial and Answer activities: see
section 3.2, steps and ).
POLQA NB Score Downlink POLQA (ITU P.863.1) voice quality score for
narrowband.
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
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 A VoIP session that was successfully set up failed
Failure to complete. Similar to “dropped 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:
MTSI Registration Failure Denotes the probability that the terminal cannot
Ratio (%) register towards IMS when requested.
MTSI Session Setup Denotes the probability that the terminal cannot
Failure Ratio (%) 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.
MTSI Session Setup Time Denotes the time elapsing from initiation of an
(s) MTSI session until a notification is received that
the session has been set up.
4 Troubleshooting
4.1 Problem: Script Activity Fails
Check that caller and callee are synchronized, that is, that the callee
reaches Answer before the caller begins 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.
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.
100 Trying
180 Ringing
181 Call is being forwarded
182 Queued
183 Session progress
200 OK
202 Accepted
Indicates that the request has been understood but actually cannot
be processed
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