7/31/2019 SIP VoIP Settings User Guide v1 1 En
1/28
SIP VoIP Settings User Guide v1.1
F O R U M N O K I A
Version 1.1; September 25, 2008
VoIP
7/31/2019 SIP VoIP Settings User Guide v1 1 En
2/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 2
Copyright 2007, 2008 Nokia Corporation. All rights reserved.
Nokia and Forum Nokia are trademarks or registered trademarks of Nokia Corporation. Other product and
company names mentioned herein may be trademarks or trade names of their respective owners.
Disclaimer
The information in this document is provided as is, with no warranties whatsoever, including any warranty ofmerchantability, fitness for any particular purpose, or any warranty otherwise arising out of any proposal,
specification, or sample. This document is provided for informational purposes only.
Nokia Corporation disclaims all liability, including liability for infringement of any proprietary rights, relating to
implementation of information presented in this document. Nokia Corporation does not warrant or represent
that such use will not infringe such rights.
Nokia Corporation retains the right to make changes to this document at any time, without notice.
License
A license is hereby granted to download and print a copy of this document for personal use only. No other license
to any other intellectual property rights is granted herein.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
3/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 3
Contents
1 Introduction................................................................................................................................................52 SIP VoIP Settings application.................................................................................................................. 63 VoIP services ............................................................................................................................................... 7
3.1 Profile settings .............. ............... ................ ............... ............... ............... ................ ................... .............93.2 Used SIP profiles ....................................................................................................................................133.3 Codecs .......................................................................................................................................................13
3.3.1 Adaptive Multi-Rate Narrow Band (AMR-NB) codec parameters.........................153.3.2 Pulse Code Modulation -law (PCMU) (G.711 -law) codec parameters...........163.3.3 Pulse Code Modulation A-law (PCMA) (G.711 A-law) codec parameters ...........173.3.4 iLBC codec parameters ......................................................................................................183.3.5 G.729 codec...........................................................................................................................183.3.6 Comfort Noise (CN) codec.................................................................................................19
4 NAT FW settings .......................................................................................................................................204.1 Domain parameters..............................................................................................................................214.2 IAP parameters.......................................................................................................................................23
5 Terms and abbreviations.......................................................................................................................256 References .................................................................................................................................................277 Evaluate this resource ............................................................................................................................28
7/31/2019 SIP VoIP Settings User Guide v1 1 En
4/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 4
Change history
October 10, 2007 Version 1.0 Initial document release
September 25, 2008 Version 1.1 Updates to all chapters for a new version of the application
7/31/2019 SIP VoIP Settings User Guide v1 1 En
5/28
7/31/2019 SIP VoIP Settings User Guide v1 1 En
6/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 6
2 SIP VoIP Settings applicationSIP VoIP Settings is a stand-alone application that enables voice over IP (VoIP) settings to be
configured on an S60 device. It can be used to create or modify VoIP profiles as well as to create
network address translation (NAT) firewall (FW) settings and modify the VoIP parameters, includingthe codec and NAT FW settings, which on S60 devices are typically invisible to users.
With the SIP VoIP Settings application, it is also possible to save all the VoIP profile settings to an Open
Mobile Alliance (OMA) Client Provisioning (CP) document and import OMA CP VoIP settings to an S60
device from an XML file.
Note: Editing is available for VoIP settings only in profiles created with the SIP VoIP Settings
application. The settings for a provisioned VoIP profile cannot be displayed or edited. Such a profile
can, however, be used as the basis of a new profile created using SIP VoIP Settings. The settings of
such a new profile can be displayed and edited.
The VoIP Settings application provides the following editing features: Create VoIP profile A VoIP profile can be created using a default or an existing VoIP profile or
using an imported XML-based CP document.
Delete VoIP profile VoIP profiles can be deleted. Save settings to file VoIP profile settings can be saved to an XML file. Edit VoIP profile settings Profile-specific VoIP settings can be viewed and modified. Select SIP profile The name of the Session Initiation Protocol (SIP) profile used in the VoIP
profile can be viewed and changed.
Create audio codec An audio codec can be created. Delete audio codec An audio codec can be deleted. Move audio codec The order (priority) of the codec can be updated. Edit audio-codec settings Audio-codec settings can be viewed and modified.The application provides the following features for editing NAT FW settings:
Create domain-specific settings Domain-specific settings can be created. Edit domain-specific settings Domain-specific settings can be viewed and modified. Delete domain-specific settings Domain-specific settings can be deleted. Create IAP-specific settings Internet Access Point (IAP) settings can be defined. Edit IAP-specific settings IAP-specific settings can be viewed and edited. Delete IAP-specific settings IAP-specific settings can be deleted.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
7/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 7
3 VoIP servicesSIP VoIP Settings provides for the creation and maintenance of a VoIP profile, which includes VoIP-
service-specific settings, such as:
VoIP service name. VoIP service parameters. Speech-codec settings. SIP profiles used.In addition, features for importing and exporting VoIP profile settings are provided.
The VoIP profile name is the same as the VoIP service name that will be seen by a user of an S60
device when selecting a VoIP service. If a VoIP service provider also configures the access networks
used so that they have, for example, a different billing or connectivity mode, two or more VoIP profiles
are needed. If the service provider does not set up the access networks, only one VoIP and one SIPprofile are needed. The VoIP profile settings are then linked to the access point selected when the user
is successfully registered to the service.
To access the VoIP services, in the SIP VoIP Settings screen, select the VoIP services item, and then
choose Options > Open, as shown in Figure 1. Now a list of any available VoIP services is displayed,
though initially there may be none to display, as shown in Figure 2.
Figure 1: Select the VoIP services
item, and then choose Options >
Open.
Figure 2: Initially, no VoIP services
may be available.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
8/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 8
To create a new VoIP profile, in the VoIP services screen, select Options > New > Use default profile or
(if profiles are present) Use existing profile, as shown in Figure 3. If using an existing VoIP profile,
select one to be copied (see Figure 4 and Figure 5).
Figure 3: Create a new VoIP profile. Figure 4: Select the VoIP profile. Figure 5: A VoIP services profile is
created.
You can also configure the VoIP profile settings by importing an XML-based CP document. To import
VoIP profile settings, in the VoIP services screen, select Options > Import, as shown in Figure 6. Then
select the memory and location of the file to be imported, as shown in Figure 7 and Figure 8.
Figure 6: Import VoIP profile
settings.
Figure 7: Select the memory. Figure 8: Select the location of the
file.
To open or delete a VoIP profile, in the VoIP services screen, choose a profile, and then select Options
> Open or Delete, as shown in Figure 9.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
9/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 9
To export all VoIP profile settings to an XML file, in the VoIP services screen, choose the profile to be
exported, and then select Options > Export, as shown in Figure 9. Next, choose the memory and
location where the file is to be saved, as shown in Figure 10 and Figure 11. The saved file can be used,
for example, for importing VoIP settings into another device.
Note: Passwords and WLAN security settings are not included in the exported XML document.
Figure 9: Select Export to save a
chosen VoIP profile setting to a file.
Figure 10: Select the memory
location in which to store the
exported file.
Figure 11: Select the folder in
which to save the exported file.
3.1 Profile settingsTo modify the VoIP profile settings, choose a profile in the VoIP services screen, and then select Profile
settings. Scroll to choose the parameter to be updated, and then select Options > Change (see Figure
12, Figure 13, and Figure 14).
Figure 12: Select VoIP profile
settings.
Figure 13: Select the profile
parameter to be modified.
Figure 14: Modify the profile
parameters value.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
10/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 10
The profile settings available are:
Provider Name:o Text, for which the maximum length is 32 characters.o
The provider of the VoIP profile settings as described in /R.1/ OMA-WAP-ProvCont-v1_1-20021112-C, which is displayed on the device UI as the sender of the settings and cannot
be edited.
Profile Name:o Text, for which the maximum length is 32 characters.o The profile name, based on the provisioning parameter NAME as described in /R.1/ OMA-WAP-
ProvCont-v1_1-20021112-C, which is displayed on the device UI as the service name.
Media QoS:o A number in the range of 063.o The Quality of Service (QoS) setting for VoIP media. The parameter value corresponds to the
DiffServ Code Point (DiffServ, DSCP bits) QoS values used in IP headers (Ipv4 TOS and Ipv6 TC)
or those defined by IETF Request for Comments (RFC) 2598, an Expedited Forwarding PHB [3].
The IETF and WMM [4] specifications conflict on the QoS values used for voice packets. From
S60 3rd Edition, Feature Pack 1 onward, the U-APSD power-save scheme of WMM is also
enabled with the IETF default value (46), if the feature is supported by the device and the
WLAN access point.
o Default value: 46. Start media port:
o An even number in the range of 102465534.o The lower limit of the real-time transport protocol (RTP) port range.o Default value: 49152.
End media port:o An even number in the range of 102465534.o The upper limit for the allocated RTP ports. The value must be at least 4 over the Start media
ports number to guarantee two simultaneous calls.
o Default value: 65534. DTMF inband:
o DTMF tones are sent as compressed audio; they are part of a VoIP calls audio stream. Notethat the DTMF tones may be degraded if a high-compression-rate codec (AMR-NB, G.729, oriLBC) is in use for a VoIP call.
o It is recommended that this value not be changed, because if enabled (see below) and ifsupported by the other peer in the VoIP call, the DTMF tones are sent as out-band.
o On: Enabled.o Off: Disabled.o Default value: On.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
11/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 11
DTMF outband:o DTMF tones are sent as RTP payload, as specified in IETF RFC 2833. If both in- and out-band
DTMF signaling methods are enabled (setting value 1), the DTMF out-band mode is used if the
peer supports it.
o Typically, both in- and out-band DTMF should be enabled; however, disabling the out-bandsignaling is required in some special cases.o On: DTMF digits out-band are generated, if requested by the remote side.o Off: DTMF digits out-band are not generated.o Default value: On.
Allow VoIP over WCDMA:o If this setting is enabled, the Internet telephone application shows the available WCDMA
access points.
o On: VoIP over WCDMA is allowed.o Off: VoIP over WCDMA is not allowed.o Default value: Off.
RTCP reporting:o This setting enables the RTCP reports defined in RFC 3550.o On: RTCP reporting is enabled.o Off: RTCP reporting is disabled.o Default value: Off.
UA Header: term. type:o On: The device type is appended to the UA header.o Off: The device type is not appended to the UA header.o Default value: On.
UA Header: MAC address:o Off: The MAC address is not appended to the UA header.o On: The MAC address is appended to the UA header.o Default value: Off.
UA Header: free string:o Text, for which the maximum length is 32 characters.o UA information string that is appended to the SIP UA header, for example, to separate two
configurations using different IAPs.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
12/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 12
Securecall preference:o This parameter enables media security (secure RTP) if SIP Transport Layer Security (TLS) has
been used for signaling. Secure-call preference is supported from Nokia S60 VoIP
Release 2.1 onward.
o Prefer secure: A secure call is preferred. If the other end does not support security, a fallbackto a nonsecure call takes place.o Prefer nonsecure: A nonsecure call is preferred.o Use secure only: Security is mandatory for mobile originated (MO) call establishment.o Default value: Prefer nonsecure.
Count of VoIP digits:o This parameter defines the meaningful count of caller-ID characters for caller identification. It
is supported from Nokia S60 VoIP Release 2.1 onward.
o 0: All caller-ID characters are meaningful.o 320: Number of meaningful caller-ID characters.o Default value: 0.
Ignoring domain part:o This parameter defines the rule for displaying the domain part of a uniform resource
identifier (URI) for incoming Internet calls in the user interface. It is supported from Nokia S60
VoIP Release 2.1 onward.
o On: The domain part is not displayed if E.164 numbers are used in the user part of the URI.o Numbers only: The domain part is never displayed.o Off: The domain part is displayed.o Default value: Off.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
13/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 13
3.2 Used SIP profilesTo view the name of the SIP profile used in the VoIP service profile or to select an alternative SIP
profile, choose the desired VoIP service from the VoIP services screen, and then select Used SIP
profiles, as shown in Figure 15, to display the SIP profiles associated with the service. The Used SIPprofiles screen, as shown in Figure 16, displays the IDs of the SIP profiles used by the selected VoIP
service. The VoIP service profile can refer to one or more SIP profiles. If there are several SIP profiles,
select from the list the profile to be used.
Figure 15: The Used SIP profiles
item in a VoIP service is displayed.
Figure 16: The associated SIP
profiles are displayed, and the
active profile can be selected.
3.3 CodecsThe VoIP service includes settings for one or more speech codecs. For codec order, see the documentNokia S60 VoIP Implementation Configuration Tutorial[2], which is available in the Documents section
of the Forum Nokia Web site.
To modify the speech-codec settings, in the VoIP service screen, choose and open a VoIP profile, then
choose and open the Codecs item, as shown in Figure 17. A list is then displayed of the codecs
associated with the VoIP profile, as shown in Figure 18. Now choose the codec to be modified and
select Options > Change to open the codecs setting details, as shown in Figure 19.
Figure 17: The Codecs item in a VoIP
profile is displayed.
Figure 18: The codecs available in a
VoIP profile are displayed.
Figure 19: Select Options > Open to
open the chosen codec settings.
http://www.forum.nokia.com/info/sw.nokia.com/id/71e0c73b-fb83-4297-ad5e-5e18c6fb2ebf/Nokia_S60_VoIP_Implementation_Configuration_Tutorial_v1_3_en.pdf.htmlhttp://www.forum.nokia.com/documentshttp://www.forum.nokia.com/documentshttp://www.forum.nokia.com/info/sw.nokia.com/id/71e0c73b-fb83-4297-ad5e-5e18c6fb2ebf/Nokia_S60_VoIP_Implementation_Configuration_Tutorial_v1_3_en.pdf.html7/31/2019 SIP VoIP Settings User Guide v1 1 En
14/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 14
A list of the codec parameters is now displayed, as shown in Figure 20. Choose a parameter to change,
and then select Options > Change, as shown in Figure 21, to open the parameter editor, as shown in
Figure 22.
Figure 20: The codec parameters
available are listed.
Figure 21: Choose the codec
parameter to be modified, andthen select Options > Change.
Figure 22: Modify the codecs
parameter value in the editor.
To change the order of the speech codecs, choose the codec to be reordered, and then select Options
> Move, as shown in Figure 23. Select a new position for the codec from the position list, as shown in
Figure 24, and the codecs position will be reordered, as shown in Figure 25.
Figure 23: Choose the codec to be
moved, and then select Options >
Move.
Figure 24: Select a new position for
the codec.
Figure 25: The position of the
codec is changed.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
15/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 15
To create a new speech codec, select Options > New, as shown in Figure 26. Select the codec to be
added, as shown in Figure 27, and the new codec is added as the last item in the codec list, as shown
in Figure 28.
Figure 26: To add a new speechcodec, first select New.
Figure 27: Select the codec to beadded.
Figure 28: A new codec is added.
To delete a codec, choose the codec to be deleted, and then select Options > Delete.Each codec has its own set of parameters and appropriate values for those parameters. These are
described in the following section.
3.3.1 Adaptive Multi-Rate Narrow Band (AMR-NB) codec parameters Ptime:
o The length of time in milliseconds represented by the media in a packet. The ptime may varybetween the codecs default ptime and maxptime so that the ptime is increased by themultiples of its allowed values. If other allowed values are not mentioned, the default value
and its multiples should be considered as the allowed value.o Default value: 20, indicating a 20-ms speech block in one RTP packet.
Maxptime:o Time in milliseconds; a value in the range of 20200.o The maximum amount of media that can be encapsulated in each packet, expressed as time
in milliseconds. The time shall be calculated as the sum of the time the media in the packet
represents. The time should be a multiple of the frame size. If this parameter is not present,
the sender may encapsulate any number of speech frames into one RTP packet. This attributeis meaningful probably only for audio data, but it may be used with other media types if it
makes sense. It is a media attribute, and it is not dependent on the character set. Note that
this attribute was introduced after RFC 2327, and implementations that have not been
updated will ignore this attribute.o Default value: 200.
Jitter buffer size:o A positive integer (milliseconds) in the range of 20200.o Default value: 200.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
16/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 16
Voice activ. detection:o Enabling of VoIP discontinuous transmission (DTX), so that RTP packets are not sent during
silent periods. AMR generates Silence Description (SID) packets during inactivity, but the
packet frequency is reduced.
o On: Enabled.o Off: Disabled.o Default value: Off.
Octet align:o On: Enabled. Octet-aligned framing used according to RFC 3267.o Off: Disabled. Bandwidth-efficient framing is employed.o Default value: Off.
Maxred:o
Definition of the maximum length of time in milliseconds that elapses between the firsttransmission of a frame and any redundant transmission that the sender will use. This
parameter allows the receiver to have a bounded delay when redundancy is used. The
allowed values are between 0 and 100. If the parameter is omitted, no limitation on the use
of redundancy is present. The value must be a multiple of the used frame time. The parameter
is supported from Nokia S60 VoIP Release 2.2 onward.
o 0: No redundancy will be used.o 20100: The maximum length of time in milliseconds that elapses between the first
transmission of a frame and any redundant transmission that the sender uses.
o Default value: 0. Mode change period:
o Definition of the frame blocks at which codec mode changes are allowed for the sender,which is also known as the frame-block period. The initial phase of the interval is arbitrary,
but changes must be separated by a period of N frame blocks; that is, a value of 2 allows the
sender to change the mode every second frame block. The value of N can be either 1 or 2. If
this parameter is not present, mode changes are allowed at any time during the session, the
equivalent of N=1. The parameter is supported from Nokia S60 VoIP Release 2.2 onward.
o 1: Mode changes are allowed at any time during the session.o 2: Mode changes are allowed every second frame block.
3.3.2 Pulse Code Modulation -law (PCMU) (G.711 -law) codec parameters Ptime:
o The length of time in milliseconds represented by the media in a packet. The ptime may varybetween the codecs default ptime and maxptime so that the ptime is increased by the
multiples of its allowed values. If other allowed values are not mentioned, the default value
and its multiples should be considered as the allowed value.o Default value: 20.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
17/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 17
Maxptime:o Time in milliseconds; a value in the range of 20200.o The maximum amount of media that can be encapsulated in each packet, expressed as time
in milliseconds. The time shall be calculated as the sum of the time the media in the packet
represents. The time should be a multiple of the frame size. If this parameter is not present,the sender may encapsulate any number of speech frames into one RTP packet. This attribute
is meaningful probably only for audio data, but it may be used with other media types if it
makes sense. It is a media attribute, and it is not dependent on the character set. Note that
this attribute was introduced after RFC 2327, and implementations that have not been
updated will ignore this attribute.o Default value: 200.
Jitter buffer size:o A positive integer (milliseconds) in the range of 20200.o Default value: 200.
Voice activ. detection:o VoIP DTX enabled so that RTP packets are not sent during silent periods. Comfort Noise (CN)
packets are also generated during inactivity if the CN codec is enabled, but the packet
frequency is reduced.
o On: Enabled.o Off: Disabled.o Default value: Off.
3.3.3 Pulse Code Modulation A-law (PCMA) (G.711 A-law) codec parameters Ptime:
o The length of time in milliseconds represented by the media in a packet. The ptime may varybetween the codecs default ptime and maxptime so that the ptime is increased by the
multiples of its allowed values. If other allowed values are not mentioned, the default value
and its multiples should be considered as the allowed value.o Default value: 20.
Maxptime:o Time in milliseconds; a value in the range of 20200.o The maximum amount of media that can be encapsulated in each packet, expressed as time
in milliseconds. The time shall be calculated as the sum of the time the media in the packet
represents. The time should be a multiple of the frame size. If this parameter is not present,the sender may encapsulate any number of speech frames into one RTP packet. This attribute
is meaningful probably only for audio data, but it may be used with other media types if it
makes sense. It is a media attribute, and it is not dependent on the character set. Note that
this attribute was introduced after RFC 2327, and implementations that have not been
updated will ignore this attribute.o Default value: 200.
Jitter buffer size:o A positive integer (milliseconds) in the range of 20200.o Default value: 200.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
18/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 18
Voice activ. detection:o VoIP DTX enabled so that RTP packets are not sent during silent periods; CN packets are also
generated during inactivity if the CN codec is enabled, but the packet frequency is reduced.
o On: Enabled.o Off: Disabled.o Default value: Off.
3.3.4 iLBC codec parameters Ptime:
o The length of time in milliseconds represented by the media in a packet. The ptime may varybetween the codecs default ptime and maxptime so that the ptime is increased by the
multiples of its allowed values. If other allowed values are not mentioned, the default value
and its multiples should be considered as the allowed value. The allowed values for this codec
are 20 and 30 and their multiples.o Default value: 30.
Maxptime:o Time in milliseconds; a value in the range of 20200.o The maximum amount of media that can be encapsulated in each packet, expressed as time
in milliseconds. The time shall be calculated as the sum of the time the media in the packet
represents. The time should be a multiple of the frame size. If this parameter is not present,
the sender may encapsulate any number of speech frames into one RTP packet. This attribute
is meaningful probably only for audio data, but it may be used with other media types if it
makes sense. It is a media attribute, and it is not dependent on the character set. Note that
this attribute was introduced after RFC 2327, and implementations that have not been
updated will ignore this attribute.o Default value: 180.
Jitter buffer size:o A positive integer (milliseconds) in the range of 20200.o Default value: 200.
Voice activ. detection:o On: Enabled.o Off: Disabled.o Default value: Off.
3.3.5 G.729 codec Ptime:
o The length of time in milliseconds represented by the media in a packet. The ptime may varybetween the codecs default ptime and maxptime so that the ptime is increased by the
multiples of its allowed values. If other allowed values are not mentioned, the default value
and its multiples should be considered as the allowed value. The allowed values for this codec
are 10 and its multiples.o Default value: 20.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
19/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 19
Maxptime:o Time in milliseconds; a value in the range of 10200.o The maximum amount of media that can be encapsulated in each packet, expressed as time
in milliseconds. The time shall be calculated as the sum of the time the media in the packet
represents. The time should be a multiple of the frame size. If this parameter is not present,
the sender may encapsulate any number of speech frames into one RTP packet. This attributeis meaningful probably only for audio data, but it may be used with other media types if it
makes sense. It is a media attribute, and it is not dependent on the character set. Note that
this attribute was introduced after RFC 2327, and implementations that have not been
updated will ignore this attribute.o Default value: 200.
Jitter buffer size:o A positive integer (milliseconds) in the range of 20200.o Default value: 200.
Voice activ. detection:o On: Enabled.o Off: Disabled.o Default value: Off.
AnnexB:o An enhancement enabled according to IETF RFC 3555 annex-b.o On: Yes.o Off: No.o Default value: Off.
3.3.6 Comfort Noise (CN) codecThis codec typically is included when PCMU or PCMA is enabled.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
20/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 20
4 NAT FW settingsThe Nokia S60 VoIP implementation has Simple Traversal of User Datagram Protocol (UDP) Through
Network Address Translators (NATs) (STUN) protocol support for NAT traversal and NAT binding-refresh
features. The traversal features of the NAT FW enable the VoIP function behind certain types of NATs.With the SIP VoIP Settings application, the user can also create the STUN settings.
NAT FW settings by default refer to SIP domain-specific settings. Refresh timers are overridden with
IAP-specific NAT FW settings, if IAP-specific values are defined.
To access the NAT FW domain or IAP-specific settings, from the main screen, choose the NAT FW
settings item, and then select Options > Open, as shown in Figure 29, to display the NAT FW settings
screen, as shown in Figure 30.
Figure 29: The main SIP VoIP
Settings screen includes the NAT
FW settings item.
Figure 30: Within the NAT FW
settings are options for domain
and IAP parameters.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
21/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 21
4.1 Domain parametersTo create domain parameters, choose and open Domain parameters, and then select Options >
Create parameters, as shown in Figure 31. The application will then request that a domain be
selected, as shown in Figure 32. The selected domain is then displayed in the NAT FW Settings screen,as shown in Figure 33.
Figure 31: Create parameters is
used to create new NAT FW
settings.
Figure 32: A domain is then
selected.
Figure 33: The selected domain is
displayed in the NAT FW screen.
To modify the domain-specific settings, choose the domain, and then select Options > Open, as shown
inFigure 34. Now select a parameter from the settings list and select Options > Change, as shown in
Figure 35, to open an editor for the setting, as shown in Figure 36.
Figure 34: Select Options > Open to
access the parameters for a
domain.
Figure 35: Choose a domain
parameter and select Options >
Change to modify it.
Figure 36: Use the parameter
editor to modify the domain
parameters value or setting.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
22/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 22
The following parameters are available for determining the NAT FW behavior for a domain:
STUN server name:o Definition of the STUN server address in the domain-specific NAT FW settings. This parameter
is optional. By default, the DNS SRV query tries to find the STUN server.o Use the value 0.0.0.0 to disable the STUN server, if, for example, a Session Border Controller
(SBC) is taking care of the NAT traversal.
o Default value: Not defined. STUN server port:
o A number in the range of 102465535.o Definition of the STUN server port in the domain-specific NAT FW settings. This parameter
is optional.
o Default value: 3478. TCP NAT bind. refresh:
o A number in the range of 09999.o Definition of the NAT-refresh interval for TCP in the domain-specific NAT FW settings. The
refresh interval is defined in seconds. If an IAP-specific value for this interval is defined, it
overrides this value. This parameter is optional.
o Default value: 1200. UDP NAT bind. refresh:
o A number in the range of 09999.o This parameter defines the NAT-refresh interval for UDP in the domain-specific NAT FW
settings. The refresh interval is defined in seconds. If an IAP-specific value for this interval is
defined, it overrides this value. This parameter is optional.
o Default value: 28. CRLF refresh:
o Definition of the use of CRLF-based NAT-binding refresh. It enables CRLF refresh to theoutbound proxy (or to the registrar if no proxy is defined) over any transport. This parameter
is optional, but enabling it is strongly recommended when there is known to be either a NAT
or firewall on the route or when the SIP proxy requires refresh to keep the persistent TCP/TLS
connection alive.
o On: Enabled.o Off: Disabled.o Default value: Off.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
23/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 23
4.2 IAP parametersNAT FW settings can also be defined for IAPs; however, by default, only IAPs that have parameters
defined are available for VoIP. All IAP types are supported.
To create IAP parameters, choose the IAP parameters item in the NAT FW settings screen, as shown in
Figure 37, and then select Options > Open. An empty list of settings will be displayed. To add an IAP,
select Options > Create parameters, as shown in Figure 38, and then select an IAP from the list
displayed, as shown in Figure 39.
Figure 37: Choose the IAP
parameters item.
Figure 38: To add an IAP, first
select Create parameters.
Figure 39: Select the IAP to be
added.
To modify the IAP-specific settings, first choose the IAP,then select Options > Open,as shown in
Figure 40. Next, choose a parameter from the list and select Options > Change, as shown in Figure 41.
The settings editor will then be displayed, as shown in Figure 42.
Figure 40: Open the IAP. Figure 41: Select the IAP parameter
to be modified, then select Change.
Figure 42: Modify the IAP
parameter value.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
24/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 24
The following parameters are available for determining the NAT FW behavior for an IAP:
TCP NAT bind. refresh:o A number in the range of 09999.o
This parameter defines the NAT-refresh interval for TCP in the IAP-specific NAT FW settings.The refresh interval is defined in seconds. The value overrides the domain-specific NAT refresh
TCP value, if it is defined. This parameter is optional.
o Default value: 1200. UDP NAT bind. refresh:
o A number in the range of 09999.o This parameter defines the NAT refresh interval for UDP in the IAP-specific NAT FW settings.
The refresh interval is defined in seconds. The value overrides the domain-specific NAT refresh
UDP value, if it is defined. This parameter is optional.
o Default value: 28. STUN retransmission:
o A number in the range of 09999.o This parameter defines the STUN-request-retransmit timer (time in milliseconds) in the IAP-
specific NAT FW settings. This parameter is optional.
o Default value: 250.
7/31/2019 SIP VoIP Settings User Guide v1 1 En
25/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 25
5 Terms and abbreviationsTerm or abbreviation Meaning
A-law Name of G.711 PCMU algorithm (European)
AMR Adaptive Multi-Rate
CN Comfort Noise
CP Client Provisioning
DSCP DiffServ Code Point
DTMF Dual-Tone Multi-Frequency
DTX Discontinuous Transmission
FW Firewall
IAP Internet Access Point
ID Identity
IEEE Institute of Electrical and Electronics Engineers
IETF Internet Engineering Task Force
iLBC Internet Low Bitrate Codec
Maxptime The maximum amount of media that can be encapsulated in a payload
packet
NAT Network address translation
NB Narrow Band
OMA Open Mobile Alliance
PCMA Pulse Code Modulation A-law
PCMU Pulse Code Modulation -law
Ptime Packetization interval
PHB Per-hop forwarding behavior
QoS Quality of Service
RFC Request for Comments
RTP Real-time transport protocol
SBC Session Border Controller
SID Silence Description
SIP Session Initiation Protocol
STUN Simple Traversal of User Datagram Protocol (UDP) Through Network
Address Translators (NATs); a protocol that allows applications to detect
that network address translation (NAT) is being used.
TC Traffic Class
TLS Transport Layer Security
http://www.ieee.org/http://www.ietf.org/http://www.openmobilealliance.org/http://www.openmobilealliance.org/http://www.ietf.org/http://www.ieee.org/7/31/2019 SIP VoIP Settings User Guide v1 1 En
26/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 26
Term or abbreviation Meaning
TOS Type of Service
U-APSD Unsolicited Automatic Power Save Delivery
URI uniform resource identifier
VAD Voice Activation Detection
VoIP Voice over IP
WLAN Wireless LAN, wireless local area network
WMM Wireless Multimedia
-law Name of G.711 PCMU algorithm (North American)
7/31/2019 SIP VoIP Settings User Guide v1 1 En
27/28
Forum.Nokia.com
SIP VoIP Settings User Guide v1.1 27
6 References[1] SIP VoIP Settings application[2] Nokia S60 VoIP Implementation Configuration Tutorial[3] IETF RFC 2598, an Expedited Forwarding PHB[4] IEEE Standard 802.11e, WiFi Multimedia (WMM) specification
http://www.forum.nokia.com/info/sw.nokia.com/id/d2d27e6c-bd52-4534-9aa6-19e606b80709/SIP_VoIP_Settings_v1_0_en.zip.htmlhttp://www.forum.nokia.com/info/sw.nokia.com/id/71e0c73b-fb83-4297-ad5e-5e18c6fb2ebf/Nokia_S60_VoIP_Implementation_Configuration_Tutorial_v1_3_en.pdf.htmlhttp://www.ietf.org/rfc/rfc3246.txthttp://www.ietf.org/rfc/rfc3246.txthttp://ieee802.org/11/http://www.wi-fi.org/knowledge_center/wmmhttp://www.wi-fi.org/knowledge_center/wmmhttp://www.wi-fi.org/knowledge_center/wmmhttp://ieee802.org/11/http://www.ietf.org/rfc/rfc3246.txthttp://www.forum.nokia.com/info/sw.nokia.com/id/71e0c73b-fb83-4297-ad5e-5e18c6fb2ebf/Nokia_S60_VoIP_Implementation_Configuration_Tutorial_v1_3_en.pdf.htmlhttp://www.forum.nokia.com/info/sw.nokia.com/id/d2d27e6c-bd52-4534-9aa6-19e606b80709/SIP_VoIP_Settings_v1_0_en.zip.html7/31/2019 SIP VoIP Settings User Guide v1 1 En
28/28
Forum.Nokia.comForum.Nokia.com
7 Evaluate this resourcePlease spare a moment to help us improve documentation quality and recognize the resources you
find most valuable, by rating this resource.
http://www.forum.nokia.com/main/1%2C%2C90%2C00.html?surveyId=c8824cfe-fff7-4fde-af62-da21a8938de2/SIP_VoIP_Settings_User_Guide_v1_1_en.pdfhttp://www.forum.nokia.com/main/1%2C%2C90%2C00.html?surveyId=c8824cfe-fff7-4fde-af62-da21a8938de2/SIP_VoIP_Settings_User_Guide_v1_1_en.pdfTop Related