Prepared by: Date: Document:

Ascom Network Testing 28 June 2011 NT11-12850 ver 1.0

Ascom (2011)
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VoIP Testing with TEMS Investigation
Technical Paper






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Contents
1 Introduction ................................................................ 1
2 How VoIP Works: A Brief Technology
Overview ..................................................................... 1
3 Testing VoIP with TEMS Investigation ..................... 2
3.1 Physical Configuration .............................................................2
3.2 Scripting ....................................................................................3
3.2.1 Tips on Scripting .........................................................................5
3.3 Choice of VoIP Client................................................................7
3.4 Output ........................................................................................7
3.4.1 VoIP-specific Information Elements ............................................7
3.4.2 Other Information Elements of Interest .......................................9
3.4.3 VoIP Events ................................................................................9
3.4.4 VoIP KPIs (Key Performance Indicators) ..................................10
3.5 Presentation in TEMS Investigation Windows ......................10
3.6 Ascom Test Setup ..................................................................11
4 Troubleshooting....................................................... 11
4.1 Problem: Script Activity Fails ................................................11
4.2 Problem: Bad Audio Quality (VoIP PESQ Score Low)..........11
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





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1 Introduction
VoIP, Voice over IP, is a technology for delivering voice communications
over IP networks such as the Internet. Being a voice-over-data service,
VoIP has characteristics from both realms. On one hand it is very delay-
sensitive (like circuit-switched voice); on the other it is subject to all of the
various challenges associated with packet-switched services, such as
keeping down packet loss and jitter. This makes VoIP a very complicated
service to optimize.
The present paper describes VoIP in general terms and tells how to test
and measure VoIP performance using TEMS Investigation.
2 How VoIP Works: A Brief Technology
Overview
VoIP (Voice over IP) is an umbrella term for technologies that enable
delivery of voice calls and multimedia sessions over IP networks, such as
the Internet, rather than the public switched telephone network (PSTN).
The voice signal is digitized and encoded using audio codecs, just as in
circuit-switched cellular telephony, and then divided into IP packets for
transmission over the packet-switched network. On the receiving side
similar steps are applied in the reverse order to reproduce the original voice
stream: reception and decoding of IP packets followed by digital-to-analog
conversion.
The range of audio codecs used differs between VoIP implementations;
some implementations rely on narrowband and compressed speech, while
others support high fidelity stereo codecs.
VoIP systems employ session control protocols to control the setup and
teardown of calls. Examples of such protocols are:
H.323
Media Gateway Control Protocol (MGCP)
Session Initiation Protocol (SIP)
Real-time Transport Protocol (RTP)
Session Description Protocol (SDP)
Of these, SIP and RTP have gained particularly widespread use, and these
protocols also figure in the present document.





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3 Testing VoIP with TEMS Investigation
3.1 Physical Configuration
VoIP testing is conducted using two instances of TEMS Investigation,
installed on two different PCs, each of which have a mobile phone
connected.
1
This setup is necessary to enable end-to-end speech quality
measurement for VoIP.
The calling device (caller) is connected to one PC and the called device
(callee) to the other. Audio is sent in semi-duplex fashion between the
parties, that is, in both directions but only in one direction at a time.
TEMS Investigation has a built-in VoIP client; the VoIP clients thus reside in
the PCs and not in the mobile devices.
It should be noted from the outset that no further devices running data
services can be connected to the PCs during VoIP testing. See also
chapter 0.

Schematic diagram of physical configuration for VoIP testing.



1
Two TEMS Investigation licenses are thus also required, as well as a special
license option for VoIP.
VoIP
server
Caller
side
Callee
side
PCs running TEMS Investigation
Mobile phones VoIP
client
VoIP
client




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3.2 Scripting
Two scripts are needed in TEMS Investigation, one for the caller and one
for the callee. Predefined snippets, VoIP Call Dial and VoIP Call Answer,
are supplied with TEMS Investigation for this purpose.
It is worth underlining that the timing between caller and callee is essential.
The callee must be registered with the SIP server and finish its
preparations for answering before the caller dials the call. See step O
below.
VoIP Call Dial snippet VoIP Call Answer snippet

Screenshots of VoIP Call Dial snippet (left) and VoIP Call Answer snippet (right) as
displayed in the TEMS Investigation Service Control Designer. The numbering refers
to the step-by-step description that follows below.

O
O
O
O
O
O
O
O
O
O
O
O
O





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OO
Network Connect
First, both parties need to have an active data session. This is done in
TEMS Investigation through the Network Connect activity.
OO
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.
OO
VoIP Answer
The callee must be ready to answer before the caller can initiate a call. To
this end the callee executes the script activity VoIP Answer. 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 VoIP Answer before the caller dials
the call, you should insert a wait period in the callers script. See
section 3.2.1. This detail has been left out of the above diagram to keep
things straightforward.
OO
VoIP Dial
The caller initiates the call by running the activity VoIP Dial. In this activity
you indicate the codec the caller should use, which must be the same as
the callees designated codec (VoIP Answer activity, see step O). You also
specify the codec rate and the phone number to call.
OO
VoIP Voice Quality Measurement
Once the call has been connected, voice quality can be measured on both
sides in terms of PESQ. This is done with the VoIP Voice Quality activity.
The call durations should preferably differ between caller and callee, so that
it can be controlled which side hangs up the call. Compare step O.
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
stored on the PC.
2
If three consecutive PESQ reports with indication of
silence are received, the VoIP call is terminated.
If for some reason you do not wish to measure voice quality, simply use a
Wait activity instead of VoIP Voice Quality. However, be aware that in this


2
Storage location: C:\Users\<user>\Documents\TEMS Product Files\TEMS
Investigation 13.0\PESQ.





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case you will not obtain any other VoIP quality measurements either, such
as FER and jitter buffer metrics (see the list in section 3.4.1), nor will MTSI
Session Completion Failure events (dropped calls) be generated.
OO
VoIP Hangup
One of the parties (the one with the duration of the VoIP 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 VoIP Hang Up activity. In the
above diagram, the caller performs the hangup.
OO
SIP Unregister
Both sides unregister from the SIP server. This is done using the SIP
Unregister activity.
OO
Network Disconnect
Finally the data session is terminated as each party performs a Network
Disconnect. If the snippet is executed in a loop, this activity is necessary to
force a disconnect from the network after each VoIP call (desirable for the
purpose of KPI calculation).
3.2.1 Tips on Scripting
The above description covered the key steps in conducting a VoIP call. In
practice, the scripts should be made slightly more sophisticated.
Suggested setup:
On both sides, run all activities within a while loop. Add an extra Wait
activity as the last item in each loop, with the callers wait period longer
than the callees, to make sure the callee is ready and waiting for the
call when the caller dials. Suggested wait durations are 30 s for the
caller and 10 s for the callee. Compare step O in section 3.2 above.
To ensure that the timing becomes right for the first VoIP call, start the
script on the callee side first, then the caller script.
If the parties are not in sync, that is, if the callee is not registered when the
caller places the call, then the caller will generate an MTSI Session Setup
Failure event (see section 3.4.3).





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Caller Callee

Use of while loops with VoIP scripting: caller (left) and callee (right).






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3.3 Choice of VoIP Client
TEMS Investigation offers CounterPath and PJSIP as VoIP clients.
The rationale for providing more than one client is as follows:
Different customers use different clients. The situation is comparable in
principle to the (much larger) range of devices available for the voice
service.
Clients vary in their level of compliance with the VoIP standard and also
in terms of general quality and performance. It should also be kept in
mind that the VoIP standard is less specific than voice telephony
standards such as 3GPP.
In the TEMS Investigation VoIP testing implementation, PJSIP is strongly
recommended for PESQ measurement. During tests with the CounterPath
client, unexplained dips in the PESQ score have been observed.
The technical description of VoIP and VoIP testing given in this document is
valid regardless of the choice of client.
3.4 Output
3.4.1 VoIP-specific Information Elements
3.4.1.1 Jitter
VoIP RFC 1889 Jitter (ms)
Packet jitter or delay variation as defined in IETF RFC 1889, section 6.3.1:
An estimate of the statistical variance of the RTP data packet interarrival
time [...] The interarrival jitter [ is defined to be the mean deviation
(smoothed absolute value) of the difference in packet spacing at the
receiver compared to the sender for a pair of packets. As shown in the
equation below, this is equivalent to the difference in the relative transit
time for the two packets [...].
If S

is the RTP timestamp from packet i, and R

is the time of arrival in

RTP timestamp units for packet i, then for two packets i and ], may be
expressed as
(i, ]) = (R
]
R

) (S
]
S

) = (R
]
S
]
) (R

)
The interarrival jitter is calculated continuously as each data packet i is
received [...] using this difference for that packet and the previous
packet i 1 in order of arrival (not necessarily in sequence), according
to the formula
[ [ +
|(i 1, i)| [
16






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The quantity [ is what is output in the VoIP RFC 1889 Jitter information
element. The latter is updated once every second.
3.4.1.2 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.
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.





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3.4.1.3 Audio Quality Related
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. Should correlate closely to PESQ.
VoIP PESQ Score PESQ (ITU P.862.1) is a method for modeling
speech quality as perceived by human listeners.
Its output is a MOS score between 1 (worst) and 5
(best). The PESQ score depends on speech
codec and codec rate, on the radio environment,
and more.
For VoIP measurements the speech sentences
are 8 s in length. This means that a MOS score
will be calculated every 16 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.
3.4.2 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).
3.4.3 VoIP Events
These events underlie the KPIs in section 0:
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.





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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.
3.4.4 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.
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.
3.5 Presentation in TEMS Investigation Windows
Suitable presentation windows for VoIP data:
VoIP Quality status window containing the information elements
described in section 3.4.1
VoIP Quality Line Chart tracking VoIP PESQ Score 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.





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3.6 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 0.
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.
4.2 Problem: Bad Audio Quality (VoIP PESQ 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, Neighbor
Cell RSRP.
Check for excessively frequent handovers.
In the IP Protocol Reports message window, look into the RTP
messages.






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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.






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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





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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





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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





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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
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