Nokia S60 VoIP Release 3 0 Configuration Tutorial v1 0 En

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F O R U M N O K I A

Nokia S60 VoIP Release 3.0 Configuration

Tutorial

Version 1.0; October 1, 2008

VoIP


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Copyright 2008 Nokia Corporation. All rights reserved.

Nokia and Forum Nokia are trademarks or registered trademarks of Nokia Corporation. Java and all Java-basedmarks are trademarks or registered trademarks of Sun Microsystems, Inc. Other product and company namesmentioned 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 toimplementation of information presented in this document. Nokia Corporation does not warrant or representthat 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 licenseto any other intellectual property rights is granted herein.

Nokia S60 VoIP Release 3.0 Configuration Tutorial 2


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Contents

1 Introduction................................................................................................................................................ 52 Supplementary VoIP settings ................................................................................................................. 7

2.1 Terminal-specific VoIP settings ...........................................................................................................73 VoIP profile settings.................................................................................................................................. 8

3.1 Profile-specific VoIP settings................................................................................................................83.2 Speech codec settings..........................................................................................................................12

3.2.1 AMR NB codec .......................................................................................................................123.2.2 PCMU (G.711 -law) codec................................................................................................133.2.3 PCMA (G.711 A-law) codec ................................................................................................143.2.4 iLBC codec ..............................................................................................................................153.2.5 G.729 codec...........................................................................................................................163.2.6

Comfort Noise codec ..........................................................................................................16

3.3 Voice mailboxspecific VoIP settings..............................................................................................17

4 SIP profile settings ..................................................................................................................................184.1 Creating a SIP profile............................................................................................................................18

4.1.1 Proxy server ..........................................................................................................................194.1.2 Registrar server ...................................................................................................................20

5 SIP/SIMPLE presence settings ...............................................................................................................215.1 Presence settings ..................................................................................................................................215.2 XDM settings............................................................................................................................................22

6 Destination network, access point, and WLAN settings................................................................237 NAT/Firewall traversal settings............................................................................................................24

7.1 Domain-specific settings.....................................................................................................................247.1.1 STUN-specific settings .......................................................................................................25

7.2 IAP-specific settings..............................................................................................................................258 VoIP settings not visible from the standard UI................................................................................279 Usage ...................................................................................................................................................2810 Terms and abbreviations.......................................................................................................................2911 References .................................................................................................................................................3112 Evaluate this resource ............................................................................................................................32



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Change history

October 1, 2008 Version 1.0 Initial document release



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1 IntroductionThis tutorial describes the configuration of Nokia S60 Voice over IP (VoIP) Release 3.0.

Nokia S60 VoIP Release 3.0 can be configured with the following setting groups:

1. VoIP profile settingsa. Codec settings

2. Supplementary VoIP settings3. SIP profile settings4. SIP/SIMPLE presence settings5. Destination network settings6. Access point settings7. NAT/Firewall traversal settingsThe following figure presents the relations between the settings.

VoIP Settings

VoIP Profile

SettingsCodec SettingsSupplementary

VoIP Settings

SIP Profile

Settings

SIP/SIMPLEPresence

Settings

NAT/FirewallTraversal

Settings

DestinationNetwork Settings

Access Point

Settings

1..1

1..1

1..1

1..1

1..1 1..1

1..n

1..1

1..11..1

1..1

1..1

0..1

0..1

0..1

1..n

0..1 0..1

1..n

1..1

0..n

1..1

Figure 1: Relations between the VoIP settings

Table 1 explains the arrow numbers displayed in Figure 1.



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Item Meaning

1..1 One setting group

1..n One or any number of setting groups

0..1 No setting group or one setting group

0..n No setting group or any number of setting groups

Table 1: Explanation of the arrow numbers

The terminal user interface (UI) has only one group of VoIP terminal settings which affects all VoIPservices. For VoIP terminal settings, see Chapter 2.

The VoIP profile includes settings for one or multiple speech codecs. For codec order, see Section 3.1"Profile-specific VoIP settings". The VoIP profile may also have SIP/SIMPLE presence settings definedfor each SIP profile. Nokia S60 VoIP Release 3.0 supports only SIP protocol. For SIP/SIMPLE presencesettings, see Chapter 5.

Each VoIP profile can only refer to one SIP profile which, again, can only refer to one destination

network with multiple Internet Access Points (IAPs). But a SIP profile can also refer straight to oneaccess point.

The NAT/Firewall traversal settings can either refer to a SIP domain or access point, or both. For SIPprofile settings, see Chapter 4.

Many of these settings are not visible from the UI but are provisioned by the service provider. OMAClient Provisioning (CP) and OMA Device Management (DM) are technologies that enable terminalprovisioning by using the over-the-air (OTA) technology. For information on editing settings that arenot visible from the standard settings UI, see Chapter 8.

Note: For documentation on the supported OTA provisioning, see www.forum.nokia.com. When usingOMA Client Provisioning for Nokia S60 VoIP Release 3.0 configuration, the message encoder mustsupport APPREF and TO-APPREF parameters to make the linking between the setting groups work. Theparameters are defined in the Open Mobile Alliance document OMA-DM-2004-0211R01-CR_AC.docavailable at http://member.openmobilealliance.org/ftp/Public_documents/DM/2004/OMA-DM-2004-0211R01-CR_AC.zip.

http://www.forum.nokia.com/http://member.openmobilealliance.org/ftp/Public_documents/DM/2004/OMA-DM-2004-0211R01-CR_AC.ziphttp://member.openmobilealliance.org/ftp/Public_documents/DM/2004/OMA-DM-2004-0211R01-CR_AC.ziphttp://member.openmobilealliance.org/ftp/Public_documents/DM/2004/OMA-DM-2004-0211R01-CR_AC.ziphttp://member.openmobilealliance.org/ftp/Public_documents/DM/2004/OMA-DM-2004-0211R01-CR_AC.ziphttp://www.forum.nokia.com/


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2 Supplementary VoIP settingsSupplementary or terminal-specific VoIP settings affect all VoIP profiles and the generic behaviour ofthe S60 telephone applications. Modifying any terminal-controlled supplementary settings, such as callwaiting, and anonymous call blocking, are matters of personal user preference. But these settings canalso be pre-adjusted at the service provisioning; the latest group of settings provisioned overdrivesthe previous one.

For terminal-specific settings on the UI, select Menu > Settings > Telephone > Call settings.

For information on provisioning terminal-specific VoIP settings, see [1] and [2].

2.1 Terminal-specific VoIP settings Internet call waiting:

o 0: Disabled. The called party is not indicated about an incoming call. The calling partyreceives a busy tone.

o 1: Enabled. The called party is indicated about a waiting call. The calling party receives awaiting indication and/or the phone continues ringing.

o Default value: 0. Anonymous call block rule:

o The value determines the rule for the Anonymous call block feature.o 0: Anonymous calls are received.o 1: Anonymous calls are rejected.o Default value: 0.

VoIP calling line identification presentation/restriction (CLIP/CLIR):o 0: CLIR disabled and CLIP enabled. The caller ID is sent.o 1: CLIR enabled and CLIP disabled.o Default value: 0.

Do not disturb setting (DND):o 0: The Do not disturb feature is disabled.o 1: The Do not disturb feature is enabled. The calling party receives a busy tone, but the call

event is registered in the log.

o Default value: 0.



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3 VoIP profile settingsVoIP profile includes VoIP service-specific settings, such as:

VoIP service name VoIP service parameters Speech codec settings SIP profile usedThe VoIP profile name is the same as the VoIP service name shown on the terminal UI. The VoIP profilesettings are linked to the access point selected when the user is successfully registered to the service.

The VoIP services work in automatic mode, in other words, once the service is enabled, it tries toregister if a network is available. The WLAN network availability is detected when the WLAN scan istriggered (the scanning frequency can be set in the WLAN settings).

For information on provisioning VoIP profile settings, see [2].

3.1 Profile-specific VoIP settings ID of the VoIP settings:

o Settings ID. The ID value is also the priority value of the VoIP profile. 0 = first profile. Provider of the settings:

o Text, the maximum length is 32.o Provider of the VoIP profile settings as described in /R.1/ OMA-WAP-ProvCont-

v1_1-20021112-C.

o This text is displayed on the terminal UI as the sender of the settings. If the VoIP service has acustomer brand name, put it here.

Name of the settings:o Text, the maximum length is 32.o Based on the provisioning parameter NAME as described in /R.1/ OMA-WAP-ProvCont-

v1_1-20021112-C.

o This text is displayed on the terminal UI as the service name. VoIP codecs in preferred order

o The following default order of speech codecs is used when the settings have been createdmanually through the Advanced Settings of the SIP VoIP Settings application (see Chapter 8):

1. AMR NB2. PCMU (G.711 -law)3. PCMA (G.711 A-law)4. iLBC5. G.7296. Comfort Noise for PCMU, PCMA, and iLBC (not a separate codec, but handled as such)

o Different order can also be defined, or some of the codecs can be entirely disabled, but this isnot recommended due to possible problem scenarios. See also Section 3.2, Speech codecsettings.

o If no codecs have been provisioned, all default speech codecs are taken into use in the defaultorder mentioned above.



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o The Comfort Noise is not an actual codec by itself, but is handled as such. This codec enablesthe usage of generic comfort noise RTP payload format with iLBC, PCMU, and PCMA codecs. TheRTP payload format is defined in RFC 3389 [8].

o If the Comfort Noise codec is disabled from the VoIP profile, the generic comfort noise RTPpayload format is neither supported nor used.

SIP profile:o ID of the SIP profile used by this VoIP profile. The VoIP profile can only refer to one SIP profile.

Start media port number:o An even number in range of 102465534.o The lower limit of the RTP port range.o Default value: 49152.

End media port number:o An even number in range of 102465534.o The upper limit for the allocated RTP ports. The value shall be at least 4 over the Start media

port number to guarantee two simultaneous calls.o Default value: 65534.

Media QoS:o A number in range of 063.o Quality of Service for VoIP media. DiffServ Code Point (Diffserv, DSCP bits) QoS values used in IP

headers (IPv4 TOS and IPv6 TC). IETF RFC 2598, an Expedited Forwarding PHB. The IETF [3] andWMM [4] specifications conflict on the QoS values used for voice packets. From S60 3rd Edition,FP1 onwards, the U-APSD power save scheme of WMM is also enabled with the IETF defaultvalue (46), if the feature is supported by the terminal and the WLAN access point.

o Default value: 46

Whether to generate DTMF in-band signaling:o DTMF tones are sent as compressed audio; they are part of the actual VoIP call audio stream.

Note that the DTMF tones may be degraded if a high-compression rate codec (AMR-NB, G.729or iLBC) is in use for a VoIP call.

o It is not recommended to change this value because if enabled (see below) and if supportedby the other peer in the VoIP call, the DTMF tones are sent as out-band.

o 0: Disabled.o 1: Enabled.o Default value: 1.

Whether to generate DTMF out-band signaling:o DTMF tones are sent as RTP payload as specified in IETF RFC 2833 [9]. If both in- and out-bandDTMF signalling 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-bandsignalling is required in some special cases.

o 0: DTMF digits out-band are not generated.o 1: DTMF digits out-band are generated, if requested by the remote side.o Default value: 1.

Secure call preference:o This parameter enables media security (secure RTP) if SIP TLS has been used for signalling.

Supported from Nokia S60 VoIP Release 2.1 onwards.o 0: A non-secure call is preferred.



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o 1: A secure call is preferred. If the other end does not support security, a fallback to non-secure call takes place.

o 2: Security is mandatory for mobile originated (MO) call establishment.o Default value: 0.

RTCP reporting:o This setting enables the Real-Time Transport Control Protocol (RTCP) reports defined in

RFC 3550 [10].

o 0: RTCP reporting is disabled.o 1: RTCP reporting is enabled.o Default value: 0.

SIP VoIP User Agent header: terminal type:o 0: The terminal type is not appended to the UA header.o 1: The terminal type is appended to the UA header.o Default value: 1.

SIP VoIP User Agent header: WLAN MAC address display:o 0: The MAC address is not appended to the UA header.o 1: The MAC address is appended to the UA header.o Default value: 0.

SIP VoIP User Agent header: free string:o Text, the maximum length is 32.o User agent information string that is appended to the SIP UA header, for example, to separate

two different configurations using different IAPs.

o Samples of the User Agent header:Terminal type 1, WLAN MAC 0, no freetext (default)

User-Agent: Nokia RM-92 V 4.0632.0.37

Terminal type 1, MAC 1, no free text User-Agent: Nokia RM-92 V 4.0632.0.37 00-16-bc-7a-14-f6

Terminal type 0, MAC 1 and free text User-Agent: 00-16-bc-7a-14-f6 MyVoIPconnection

Terminal type 1, MAC 0 and free text(recommended)

User-Agent: Nokia RM-92 V 4.0632.0.37 MyVoIPconnection

Caller ID characters:o This parameter defines the meaningful count of caller ID characters for caller identification. It

calculates the number of less important digits in the numeric phone number received as a SIP

user name in the incoming call to find the caller name in the phonebook. For example,[email protected] is calling and 0400123456 is saved in the phonebook. If thecharacter count is 9 or less, the caller is mapped. Supported from Nokia S60 VoIP Release 2.1onwards.

o 0: All caller ID characters are meaningful.o 320: Number of meaningful caller ID characters.o Default value: 0.

URI domain ignore rule:o This parameter defines the rule for displaying the domain part of an address (URI) for

incoming Internet calls in the user interface. For example, it forces using the default SIPdomain for all incoming calls disregarding the end users intention and should therefore only

be used to conceal variation in the SIP domain. Option 1 is safer, because the SIP domain is

mailto:[email protected]:[email protected]


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harmonized only when there is a valid international phone number in the SIP user name.Supported from Nokia S60 VoIP Release 2.1 onwards.

o 0: The domain part is displayed.o 1: The domain part is not displayed if only E.164 numbers are used in the user part of the

URI.

o 2: The domain part is never displayed.o Default value: 0.

Auto-accept buddy request:o This parameter defines the rule for automatic acceptance of presence subscription requests.

Supported from Nokia S60 VoIP Release 3.0 onwards.

o 0: Buddy requests are not accepted automatically.o 1: Buddy requests are accepted automatically.o Default value: 0.

Add user phone:o This parameter defines the rule for adding the text string user=phone to all SIP URIs

containing only numbers as a user name. Supported from Nokia S60 VoIP Release 3.0onwards.

o 0: User=phone is not added to all numbers.o 1: User=phone is added to all numbers.o Default value: 0.

Service provider bookmark URI:o This parameter defines the URI for the service provider bookmark, for example,

http://www.example.com . Supported from Nokia S60 VoIP Release 3.0 onwards.

NAT/FW profile ID:o ID of the domain-specific NAT/FW settings. Supported from Nokia S60 VoIP Release 3.0

onwards.

Min session interval:o This parameter defines the minimum value for a session interval in delta seconds. Supported

from Nokia S60 VoIP Release 3.0 onwards.o Default value: 90.

Session expires:o This parameter defines the session interval for a SIP session in seconds; a value in range of

0-3600. Supported from Nokia S60 VoIP Release 3.0 onwards.o

Default value: 1800.

Presence settings ID:

o ID of the presence settings. Supported from Nokia S60 VoIP Release 3.0 onwards. Destination network ID:

o ID of a destination network. Links the provisioned VoIP settings to the existing destinationnetworks stored in the terminal. Supported from Nokia S60 VoIP Release 3.0 onwards.

Auto-enable:o This parameter defines the VoIP service enabling after provisioning. Supported from Nokia

S60 VoIP Release 3.0 onwards.

o 0: The VoIP service is enabled manually.o 1: The VoIP service is enabled automatically.



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o Default value: 03.2 Speech codec settings3.2.1 AMR NB codec

VoIP codec ID:o The VoIP codec ID is referred to by the VoIP settings. A generated, positive decimal number.

Media type name:o Value: audio

Media subtype name:o Value: AMR

Jitter buffer size:o A positive integer (milliseconds) in range of 20200.o

Default value: 200. Octet-align:

o 0: Disabled. Bandwidth efficient framing is employed.o 1: Enabled. Octet-aligned framing used according to RFC 4867 [12].o Default value: O.

Ptime:o The length of time in milliseconds represented by the media in a packet. The ptime may vary

between 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 valueand its multiples should be considered as the allowed value.

o Default value: 20, which means a 20 ms speech block in one RTP packet. Maxptime:

o Time in milliseconds; a value in range of 20200.o The maximum amount of media which can be encapsulated in each packet, expressed as time

in milliseconds. The time shall be calculated as the sum of the time the media present in thepacket represents. The time should be a multiple of the frame size. If this parameter is notpresent, the sender may encapsulate any number of speech frames into one RTP packet. Notethat this attribute was introduced after RFC 2327 [13], and non-updated implementations willignore this attribute.

o Default value: 200. Voice Activation Detection (VAD):

o Enabling VoIP Discontinuous Transmission (DTX), that is, RTP packets are not sent duringsilent periods; AMR generates Silence Description (SID) packets also during inactivity, but thepacket frequency is reduced.


Mode-set:o This parameter defines the requested AMR mode set. Restricts the active codec mode set to a

subset of all modes. A possible value is a space-separated list of modes from the set 07 inascending order.

o 0: AMR 4,75 kbit/sNokia S60 VoIP Release 3.0 Configuration Tutorial 12


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o 1: AMR 5,15 kbit/so 2: AMR 5,90 kbit/so 3: AMR 6,70 kbit/s (PDC-EFR)o 4: AMR 7,40 kbit/so 5: AMR 7,95 kbit/so 6: AMR 10,2 kbit/so 7: AMR 12,2 kbit/s (GSM-EFR)

Mode-change-period:o This parameter defines a number of frame-blocks, N (1 or 2), that is, the frame-block period,

at which codec mode changes are allowed for the sender. The initial phase of the interval isarbitrary, but changes must be separated by a period of N frame-blocks, that is, a value of twoallows the sender to change the mode every second frame-block. The value of N shall beeither 1 or 2. If this parameter is not present, mode changes are allowed at any time duringthe session, that is, N=1.

o 1: Mode changes are allowed at any time during the session.o 2: Mode changes are allowed every second frame-block.

Mode-change-neighbor:o 0: Change between any two modes in the active codec mode set is allowed.o 1: Mode changes SHALL only be made to the neighboring modes in the active codec mode

set. Neighboring modes are the ones closest in bit rate to the current mode, either the nexthigher or the next lower rate.

Maxred:o This parameter defines the maximum length of time in milliseconds that elapses between the

first transmission of a frame and any redundant transmission that the sender will use. Thisparameter 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 useof redundancy is present. The value must be a multiple of the used frame time. Supportedfrom Nokia S60 VoIP Release 2.2 onwards.

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 will use.

o Default value: not used.3.2.2 PCMU (G.711 -law) codec

VoIP codec ID:o The VoIP codec ID is referred to by the VoIP settings. A generated, positive integer.


Media subtype name:o Value: PCMU

Jitter buffer size:o A positive integer (milliseconds) in range of 20200.o Default value: 200

Ptime:



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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. The allowed values for this codec are 10 or its multiples.

o Default value: 20. Maxptime:




o Enabling VoIP DTX, that is, RTP packets are not sent during silent periods; Comfort Noisepackets are also generated during inactivity if enabled as CN codec, but the packet frequency

is reduced.


3.2.3 PCMA (G.711 A-law) codec VoIP codec ID:

o The VoIP codec ID is referred to by the VoIP settings. A generated, positive integer. Media type name:

o Value: audio Media subtype name:

o Value: PCMA Jitter buffer size:

o A positive integer (milliseconds) in range of 20200.o Default value: 200.


between the codecs default ptime and maxptime so that the ptime is increased by the

multiples of its allowed values. The allowed values for this codec are 10 or its multiples.







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o Enabling VoIP DTX, that is, RTP packets are not sent during silent periods; Comfort Noisepackets are also generated during inactivity if enabled as CN codec, but the packet frequencyis reduced.

o 0: Disabledo 1: Enabledo Default value: 0

3.2.4 iLBC codec VoIP codec ID:

o 0 or a positive integer. The VoIP codec ID is referred to by the VoIP settings. A generated,positive integer.


Media subtype name:o Value: iLBC

Jitter buffer size:o A positive integer (milliseconds) in range of 20200.o Default value: 200.


between 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 valueand its multiples should be considered as the allowed value. The allowed values for this codecare 20 and 30 or their multiples.






Mode-set:o The iLBC codec has support for two basic frame lengths (modes): 20 ms at 15.2 kbit/s and 30

ms at 13.33 kbit/s. When the codec operates at block lengths of 20 ms, it produces 304 bitsper block. Similarly, for block lengths of 30 ms, it produces 400 bits per block. The mode valueis configured the same way as the mode-set in AMR configuration.

o Default value: 30. If the mode-set=20 is configured, the terminal will offer the iLBC with themode=20 (15.2 kbit/s), but will fall back to using the mode=30 (13.33 kbit/s) if the remote

end does not include the mode=20 in its answer.



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3.2.5 G.729 codec VoIP codec ID:

o The VoIP codec ID is referred to by the VoIP settings. A generated, positive integer. Media type name:

o Default value: audio Media subtype name:

o Default value: G729 Jitter buffer size:

o A positive integer (milliseconds) in range of 20200.o Default value: 200.


between 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 valueand its multiples should be considered as the allowed value. The allowed values for this codecare 10 or its multiples.






Annex b:o A number. Enable enhancement according to IETF RFC 3555 [14] annex-b .o 1: Yeso

0: No

o Default value: 03.2.6 Comfort Noise codec

This codec is typically included if PCMU, PCMA or iLBC is enabled.

VoIP codec ID:o The VoIP codec ID is referred to by the VoIP settings. A generated, positive decimal number.


Media subtype name:



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o Value: CN3.3 Voice mailboxspecific VoIP settings

Used SIP profile:o SIP settings ID. A positive integer. Supported from Nokia S60 VoIP Release 3.0 onwards.

IP voice mailbox URI:o A SIP or TEL URI (IETF RFC 3261 [11] or 3966 [15]) defining the Message Waiting Indication

(MWI) address of the users IP voice mailbox. For example,sip:[email protected] . Supported from Nokia S60 VoIP Release 3.0 onwards.

IP voice mailbox listening URI:o A SIP or TEL URI (IETF RFC 3261 [11] or 3966 [15]) defining the listening address of the users IP

voice mailbox. For example, sip:[email protected]. Supported from Nokia S60 VoIPRelease 3.0 onwards.

Used IAP ID:o ID of an Internet Access Point (IAP). If this parameter is not present, the IAP/SNAP used by the

SIP settings is used. Supported from Nokia S60 VoIP Release 3.0 onwards.

Re-subscribe interval:o This parameter defines the interval for sending a re-SUBSCRIBE message in seconds.

Supported from Nokia S60 VoIP Release 3.0 onwards.



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4 SIP profile settingsSession Initiation Protocol (SIP) profile settings include items such as:

Public user name (SIP AOR). SIP registrar server address and authentication credentials. Proxy server address and authentication credentials.Multiple S60 client applications can use a shared SIP profile simultaneously, but it is recommended toconfigure VoIP to use unique SIP profiles, to avoid conflict situations. The SIP profile is configured toWhen needed mode, but if the VoIP service has been enabled, registration to the service is alwaysautomatic whenever under the WLAN coverage.

The SIP profiles provisioned with the VoIP service are always locked and therefore cannot be edited.

For information on provisioning SIP profile settings, see References [5] and [2].

4.1 Creating a SIP profileTo create a SIP profile for VoIP use, the following SIP settings are needed:

Profile name:o Give a name to describe the settings, identify this particular set of parameters. Using the same

name as for the VoIP profile settings is recommended.

Service profile:o A selection between IETF and 3GPP SIP dialects.o Default value: Internet (use 3GPP with IMS).

Destination network ID:o ID of a destination network. Links the provisioned SIP settings to the existing destination

networks stored in the terminal. A destination network can contain multiple access points. Ifpresent, this parameter overrides the default access point setting. Supported from Nokia S60VoIP Release 3.0 onwards.

Default access point:o WLAN access point should be used, select the one created for VoIP use in provisioning. If a

destination network is defined, this parameter is read only.

Signaling QoS:o A number in range of 063.o Quality of Service for SIP signalling. DiffServ Code Point (Diffserv, DSCP bits) QoS values used in

IP headers (Ipv4 TOS and Ipv6 TC). For more information, see IETF RFC 2597 [16], 3260 [17],and 4594 [18].

o Default value: 40. Public user name:

o A SIP URI for the user including the hostname part, that is, the SIP domain name provided bythe service provider. For example, sip:[email protected] or

sip:[email protected] .

o To enable registering a SIPS URI, use the prefix sips: Note that always when registering aSIPS URI, a persistent TLS connection will be created towards the proxy (or registrar if noproxy is used).



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Use compression:o Signalling compression can be used with a cellular radio to reduce the data generated by the

SIP signalling, requires an outbound proxy support, may cause error situations in poor WLANcoverage, and is thus not recommended with WLAN.

o Default value: No. Registration mode:

o When needed: Registration manually.o Always on: Registration when the terminal is started and a configured WLAN access point is

available.

Security negotiation:o Requires a sec-agree support from the SIP server side. Used mainly with IMS in 3GPP mode.o Default value: Inactive.

Proxy server:o Needed if an outbound proxy is used.

Registrar server:o Always needed.

4.1.1 Proxy server Proxy address:

o The address of the SIP outbound proxy, leave empty if the outbound proxy is not used.o If the proxy server is defined as FQDN, resolving the related IP address will be done with the

procedures specified within RFC 3263 [19]. A short summary is as follows:

- If the FQDN is supplied without port and transport parameters, the terminal willtry to resolve it using the DNS NAPTR, SRV, and finally the A and AAAA queries.

- If the FQDN is supplied without the port but with the transport parameter, theterminal will try to resolve it using the DNS SRV and finally the A and AAAAqueries.

- If the FQDN is supplied with the port parameter, the terminal will try to resolve itusing only the A and AAAA queries.

o Apply value 0.0.0.0 to enable the discovery of SIP proxy using the DHCP Option 120.o This setting creates a pre-route set according to RFC 3261 [11]; the Route header is inserted in

the initial request.

o The proxy has to insert Record-Route headers to keep itself in the route set on later requestsinside a SIP session.

o To enable SIP over TLS towards the proxy, use the prefix sips:. User name:

o Needed for proxy authentication (not necessarily the same as the public user name). Theauthentication user name may simply be the user name part of the public SIP URI, but it mayalso contain the SIP domain or the SIP scheme as a prefix, that is:

john.doe

[email protected]

sip:[email protected]

o Depends on the proxy vendor and configuration. This value is only needed if the proxy andregistrar require different credentials.

mailto:[email protected]:[email protected]


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Password:o Needed if proxy authentication is used (provided by the service provider).o This value is only needed if the proxy and registrar require different credentials.

Loose routing:o Yes for RFC 3261-compliant SIP proxy.o No for SIP 1.0 specification-compatible strict routing mode.

Transport protocol:o This setting affects all initial requests. A possible transport parameter on the next hop in the

Record-Route or Contact overdrives this setting.

o UDP UDP transport forced to be used. Set if the proxy does not support TCP.o TCP TCP transport forced to be used. Set if a persistent TCP is used for NAT traversal.o Auto transport selected according to RFC 3261 [11], that is, UDP is used for 1300 bytes,

and TCP for >1300 bytes long initial requests. The transport is TLS (RFC 3263 [19]) for SIPS.

Port:o TCP and/or UDP port the SIP proxy is listening to, the default value of 5060 is typically used.

4.1.2 Registrar server Registrar address:

o FQDN of the registrar server or the SIP domain, that is, the hostname part of the users SIP URI.o To enable SIP over TLS towards the registrar, use the prefix sips:.

User name:o Needed for user authentication. Often, but not necessarily always the same as the user name

of the public SIP URI. This parameter is often called a private user ID. Normally, the same value

is also used for the proxy authentication, see Section 4.1.1, "Proxy server". Password:

o Needed for registrar authentication. Typically the same value is also used for the proxyauthentication.

Transport protocol:o UDP UDP transport forced to be used. Set if the registrar does not support TCP or if UDP is

always to be used in case an outbound proxy is not defined.

o TCP TCP transport used for the REGISTER request sent to the registrar and forces to use TCPtransport for all initial requests in case an outbound proxy is not defined.

o Auto transport selected according to RFC 3261 [11], that is, UDP is used for 1300 bytes long initial requests. The transport may also be TLS (RFC 3263 [19]).

o This setting should only be used if no proxy has been defined. In that case it affects all initialSIP requests. Otherwise, it only adds a transport parameter to the Request-URI of the requestsent by the proxy to the registrar. The transport between the VoIP client and the proxyremains unaffected.

Port:o TCP and/or UDP port the SIP registrar is listening to, the default value of 5060 is typically used.



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5 SIP/SIMPLE presence settingsPresence is the ability and willingness of a user to communicate across a various set of services anddevices. The ability to communicate is generated automatically by the device capabilities but thewillingness can be defined by the user. Applications such as instant messaging (IM) use presenceinformation, that is, they show the contacts presence status in the contact list to inform the userwhether the recipient is available to respond to the communication request. Presence information islocated on the presence server, from where the user can access it with different devices.

In communication applications such as VoIP, IM, and presence, it is necessary for the network serversto access per-user information (for example, buddy lists and presence authorization lists) in theprocess of serving a request. This information resides within the network but is managed by the endusers themselves. The XML Configuration Access Protocol (XCAP) defines a common mechanism thatmakes the user-specific service-related information accessible to the service enablers. The XCAPspecifies how this information is defined in well-structured XML documents, and the common protocolfor access and manipulation of such documents.

This document describes the presence settings that are used for IETF SIP SIMPLE presence version 1.0

and the XDM settings that are needed to access the XDM server. The user can receive the presence andXDM settings, for example, by OMA Client Provisioning, but the settings can also be created manually.

To create presence and XDM settings manually, select Menu > Settings > Connection > Presence/XDMsettings.

5.1 Presence settings Server name:

o The name of the presence server. SIP profile:

o The SIP profile used for the presence settings. The available profiles are presented on a list indefault order.

XDM settings:o The XDM settings ID used for the presence settings. The available XDM settings are presented

on a list in default order.

Object size:o The maximum size of the MIME object data in bytes. A value in range of 0999999999.o Default value: 5120.

Publish interval:o The minimum time interval in seconds between two consecutive publications. A value in

range of 0999999.

o Default value: 3600. Maximum subscriptions:

o The maximum number of presence subscriptions. A value in range of 09999. If this value isleft empty, the number of subscriptions is unlimited.

Max. contacts in list:o The maximum number of contacts in the subscription list. A value in range of 09999. If this

value is left empty, the number of contacts is unlimited.

Domain syntax:o

The syntax of the service URI, for example, @domain.



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5.2 XDM settings XDM set name:

o The name of the XDM setting set. Access point:

o Select the access point for the XDM settings. XCAP root URI:

o The URI of the XDM server. HTTP username:

o Needed for the XDM settings authentication. HTTP password:

o Needed for the XDM service authentication. Optional.



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6 Destination network, access point, and WLAN settingsInstead of explicit internet access point (IAP)association, an application can ask for aconnection to a certain destination network, for example, Internet or Intranet. Each destination can

contain numerous access technologies that can be used to reach the destination or service domain inquestion. When establishing a connection to a destination, the device automatically selects the bestavailable access point and roams between the access points defined inside that particular destination.The user can create new destinations and manage the existing ones, but destinations can also beprovisioned to the device.

WLAN scanning is activated when the VoIP service is set to 'auto-enable' state.

Note that hidden WLAN networks are not scanned if the terminal is in Offline mode.

To create access points manually, select Menu > Settings > Connection > Network destinations.



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7 NAT/Firewall traversal settingsNokia S60 VoIP Release 3.0 has Simple Traversal of UDP through NAT (STUN) support for NAT traversaland NAT binding refresh features.

The NAT/Firewall settings are not visible from the standard UI (see Chapter 8).

If the NAT/Firewall settings are configured without the STUN server address, the terminal tries to findone with a DNS SRV query using the Public User Identity domain of the used SIP profile.

Additional dummy packet (CRLF) refresh for signaling may be enabled by provisioning. The CRLFrefresh is also enabled automatically if the STUN server is situated in a different address from the SIPproxy/registrar. The CRLF refresh is also enabled if there is no STUN server configured, and the usersown IP address is in a private address space, since the terminal is then presumably behind a NAT.

NAT/Firewall settings by default refer to SIP domain-specific settings. Refresh timers are overriddenwith IAP-specific NAT/FW settings, if IAP-specific values are defined.

It is recommended to use TCP as the transport instead of UDP since even doubled battery life can beachieved with a UI always connected to a SIP service.

For information on provisioning NAT/FW settings, see [6] and [2].

7.1 Domain-specific settings Domain:

o SIP domain to specify domain-specific NAT/FW traversal values. NAT refresh TCP:

o This parameter defines the NAT refresh interval for TCP in the domain-specific NAT-FWsettings. The unit of the refresh interval is seconds. If an IAP-specific value for this interval isdefined, it overrides this value. Optional.

o Default value: 1200. NAT refresh UDP:

o This parameter defines the NAT refresh interval for UDP in the domain-specific NAT-FWsettings. The unit of the refresh interval is seconds. If an IAP-specific value for this interval isdefined, it overrides this value. Optional.

o Default value: 28. Enable CRLF refresh:

o This parameter defines the usage of CRLF-based NAT binding refresh. This attribute enablesthe CRLF refresh to the outbound proxy (or to the registrar if no proxy is defined) over anytransport. Optional, but enabling is strongly recommended if it is known that there is either aNAT or firewall on the route, or if the SIP proxy requires refresh to keep the persistent TCP/TLSconnection alive.

o 0: CRLF refresh is disabled.o 1: CRLF refresh is enabled (CRLF over UDP is sent even when there is no STUN server).o Default value: 0.

Used NAT protocol:o This parameter defines the NAT protocol used. Optional. Supported from Nokia S60 VoIP

Release 3.0 onwards.

o Default value: nokia.stun.



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7.1.1 STUN-specific settings STUN server addresses:

o This parameter defines the STUN server addresses in the domain-specific NAT-FW settings.Optional. By default, the DNS SRV query tries to find the STUN servers.

o Apply value 0.0.0.0 to disable the STUN servers, for example, if a SBC is taking care of the NATtraversal.

STUN server ports:o This parameter defines the STUN server ports in the domain-specific NAT-FW settings.

Optional.

o Default value: 3478. STUN server user name:

o Needed for STUN server authentication with long-term credentials. Short-term credentialsthat do not need to be provisioned are generally used. Optional. Supported from Nokia S60VoIP Release 3.0 onwards.

STUN server password:o Needed for STUN server authentication with long-term credentials. Optional. Supported from

Nokia S60 VoIP Release 3.0 onwards.

STUN shared secret:o This parameter indicates whether the STUN server supports TLS and shared secret or not.

Optional. Supported from Nokia S60 VoIP Release 3.0 onwards.

o 0: STUN shared secret mechanism is not used.o 1: STUN shared secret mechanism is used.o Default value: 0.

Start port number:o A number in range of 4915265535.o The lower limit of the domain port range. Do not edit this value unless for a specific reason. In

any case, do not set the port range to zero because using the same value for start and endport numbers would prevent all VoIP calls. Optional. Supported from Nokia S60 VoIP Release3.0 onwards.

o Default value: 49152. End port number:

o A number in range of 4915265535.o The upper limit of the domain port range. Do not edit this value unless for a specific reason. In

any case, do not set the port range to zero because using the same value for start and endport numbers would prevent all VoIP calls. Optional. Supported from Nokia S60 VoIP Release3.0 onwards.

o Default value: 49200.7.2 IAP-specific settings

Access point ID:o Access point ID of the access point-specific NAT/FW traversal settings. Optional.

NAT refresh TCP:o This parameter defines the NAT refresh interval for TCP in the IAP-specific NAT-FW settings.

The unit of the refresh interval is seconds. The value overrides the domain-specific NAT

Refresh TCP value, if it is defined. Optional.o Default value: 1200.



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NAT refresh UDP:o This parameter defines the NAT refresh interval for UDP in the IAP-specific NAT-FW settings.

The unit of the refresh interval is seconds. The value overrides the domain-specific NATRefresh UDP value, if it is defined. Optional.

o Default value: 28. STUN retransmit:

o This parameter defines the STUN request retransmit timer (time in milliseconds) in the IAP-specific NAT-FW settings. Optional.

o Default value: 250.



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8 VoIP settings not visible from the standard UIThe following VoIP settings are not visible from the terminal standard user interface, but can beedited through the Advanced Settings of the SIP VoIP Settings application. For more information onthe application, see the SIP/VoIP Settings User Guide for Nokia S60 VoIP Release 3.0[7].

Speech codec selection, order, and codec-specific settings Voice mailbox-specific VoIP settings NAT/Firewall settings The following VoIP profile settings:

o Provider of the settingso Start media port numbero End media port numbero Media QOSo Whether to generate DTMF in-band signallingo Whether to generate DTMF out-band signallingo RTCP reportingo SIP VoIP User Agent header: terminal typeo SIP VoIP User Agent header: WLAN MAC addresso SIP VoIP User Agent header: free stringo Secure call preferenceo Caller ID characterso URI domain ignore ruleo Add user phoneo Service provider bookmark URIo NAT/FW profile IDo Min session intervalo Session expireso Presence settings IDo Destination network IDo Auto-enable

http://forum.nokia.com/info/sw.nokia.com/id/8c94bef3-bc2f-4723-8552-3e95b1c90206/SIP_VoIP_Settings_User_Guide.htmlhttp://forum.nokia.com/info/sw.nokia.com/id/8c94bef3-bc2f-4723-8552-3e95b1c90206/SIP_VoIP_Settings_User_Guide.html


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9 UsageAll VoIP service settings are installed and managed from the Internet communications settings. A

new service tab is created in the phonebook for each VoIP service installed. From the phonebooksVoIP service tab the user can for example:

Enable service Search for WLAN Make an Internet call Edit contact details Handle presence information (subscribe, unsubscribe or block contacts), if supported Edit settings, such as the connectivity preferencesWhen the settings have been provisioned or set up manually, the user may still need to set up the

access network: If the service provider has not provisioned the access points, the user can add them in the

phonebooks VoIP service tab by selecting the Search for WLAN option which starts a WLANnetwork scan. Selecting an access point from the list displayed will enable service and establish anetwork connection. The selected network is saved to the services destination.

Nokia S60 VoIP Release 3.0 terminal may be registered simultaneously to cellular network andmultiple VoIP services. Mobile originated (MO) calls are made with the following rules:

1. If multiple VoIP services are available, the call is made to the service the user selects.2. If the user specifically selects the Internet or Voice call from the phonebook or log, the call is made

respectively as VoIP or cellular.

3. If the user calls to a SIP URI, the call is made as VoIP.4. If the user is not registered to a VoIP service and makes a call to a number containing only 09, *,

and #, the call will be made as cellular.

5. If the user tries to make an Internet call without an active VoIP service registration, theregistration is first prompted and then performed if a known VoIP-capable access point isavailable.



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10Terms and abbreviationsTerm or abbreviation Meaning

3GPP The 3rd

Generation Partnership Project

a-law Name of G.711 PCMU algorithm (European)

AMR Adaptive Multi Rate

CLIP Calling Line Identification Presentation

CLIR Calling Line Identification Restriction

CN Comfort Noise

CP Client Provisioning

CRLF Carrier Line Feed

DM Device Management

DND Do Not Disturb

DSCP DiffServ Code Point

DTMF Dual-Tone Multi Frequency

DTX Discontinuous Transmission

FW Firewall

HTTP Hyper Text Transport Protocol

IAP Internet Access Point

ID Identity

IEEE Institute of Electrical and Electronics Engineers (www.ieee.org)

IETF The Internet Engineering Task Force (www.ietf.org)

iLBC Internet Low Bitrate Codec

IM Instant Messaging

IMS IP Multimedia System

IP Internet Protocol

Maxptime The maximum amount of media which can be encapsulated in apayload packet.

MWI Message Waiting Indication

NAT Network Address Translation

NB Narrow Band

OMA Open Mobile Alliance (www.openmobilealliance.org)

OTA Over-the-Air

PCMA Pulse Code Modulation a-law

PCMU Pulse Code Modulation -law

Ptime Packetization interval

PHB Per-Hop forwarding Behaviour

QoS Quality of Service

http://www.ieee.org/http://www.ietf.org/http://www.openmobilealliance.org/http://www.openmobilealliance.org/http://www.ietf.org/http://www.ieee.org/


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Term or abbreviation Meaning

RFC Request For Comments

RTP Real-Time Transport Protocol

SBC Session Border Controller

SID Silence Description

SIP Session Initiation Protocol

SNAP Service Network Access Point

SW Software

STUN Simple Traversal of UDP through NAT; a protocol that allowsapplications to detect that network address translation (NAT) is beingused.

TEL Telephony

TC Traffic Class

TLS Transport Layer Security

TOS Type of Service

U-APSD Unsolicited Automatic Power Save Delivery

URI Uniform Resource Identifier

VAD Voice Activation Detection

VoIP Voice over IP

VPN Virtual Private Network

WLAN Wireless LAN, Wireless Local Area Network

WMM Wireless MultimediaXCAP XML Configuration Access Protocol

XDM XML Document Management

-law Name of G.711 PCMU algorithm (North American)



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11References[1] OMA DM: Device Management Object for Generic VoIP Settings, available at

www.forum.nokia.com

[2] Client Provisioning Registration, available at www.forum.nokia.com[3] IETF RFC 2598, An Expedited Forwarding PHB, available at www.ietf.org[4] IEEE Standard 802.11e, Wireless Multimedia (WMM) Specification, available at

www.ieee.org

[5] OMA DM: Device Management Object for SIP, available at www.forum.nokia.com[6] OMA DM: Device Management Object for NATFW, available at www.forum.nokia.com[7] SIP VoIP Settings User Guide for Nokia S60 VoIP Release 3.0, available at

www.forum.nokia.com

[8]

IETF RFC 3389, available at RTP Payload for Comfort Noise, www.ietf.org[9] IETF RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals,

available at www.ietf.org

[10] IETF RFC 3550, RTP: A Transport Protocol for Real-Time Applications, available atwww.ietf.org

[11] IETF RFC 3261, SIP: Session Initiation Protocol, available at www.ietf.org[12] IETF RFC 4867, RTP Payload Format for AMR and AMR-WB, available at www.ietf.org[13] IETF RFC 2327, SDP: Session Description Protocol, available at www.ietf.org[14] IETF RFC 3555, MIME Type Registration of RTP Payload Formats, available at www.ietf.org[15] IETF RFC 3966, The tel URI for Telephone Numbers, available at www.ietf.org[16] IETF RFC 2597, Assured Forwarding PHB Group, available at www.ietf.org[17] IETF RFC 3260, New Terminology and Clarifications for Diffserv, available at www.ietf.org[18] IETF RFC 4594, Configuration Guidelines for DiffServ Service Classes, available at

www.ietf.org

[19] IETF RFC 3263, SIP: Locating SIP Servers, available at www.ietf.org

http://www.forum.nokia.com/info/sw.nokia.com/id/3b0dcd32-e8e1-42fb-919a-6a57fe4a82ad/OMA_DM_Management_Object_for_Generic_VoIP_Settings_v1_1_en.pdf.htmlhttp://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/ea58757f-e5c8-4d0a-97ae-edfd62c41db0/Client_Provisioning_Registration_v1_8.zip.htmlhttp://www.forum.nokia.com/http://www.ietf.org/http://www.ieee.org/http://www.forum.nokia.com/info/sw.nokia.com/id/9b6100d7-0993-4ab6-9d10-eb047cfe5210/OMA_DM_Management_Object_for_SIP.htmlhttp://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/a0cf7076-9741-4510-911a-8f80fedcf57a/OMA_DM_Management_Object_for_NATFW_v1_1_en.pdf.htmlhttp://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/e0764893-0513-416a-98a7-76abd6934270/SIP_VoIP_Settings_User_Guide_for_Nokia_S60_VoIP_Rel_3_v1_0_en.pdfhttp://www.forum.nokia.com/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/http://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/e0764893-0513-416a-98a7-76abd6934270/SIP_VoIP_Settings_User_Guide_for_Nokia_S60_VoIP_Rel_3_v1_0_en.pdfhttp://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/a0cf7076-9741-4510-911a-8f80fedcf57a/OMA_DM_Management_Object_for_NATFW_v1_1_en.pdf.htmlhttp://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/9b6100d7-0993-4ab6-9d10-eb047cfe5210/OMA_DM_Management_Object_for_SIP.htmlhttp://www.ieee.org/http://www.ietf.org/http://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/ea58757f-e5c8-4d0a-97ae-edfd62c41db0/Client_Provisioning_Registration_v1_8.zip.htmlhttp://www.forum.nokia.com/http://www.forum.nokia.com/info/sw.nokia.com/id/3b0dcd32-e8e1-42fb-919a-6a57fe4a82ad/OMA_DM_Management_Object_for_Generic_VoIP_Settings_v1_1_en.pdf.html


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