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SIP VoIP Release 3.x Settings Application User Guide
Version 2.1; 26 November 2009

VoIP

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Copyright © 2008, 2009 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 of merchantability, 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. Licence A licence is hereby granted to download and print a copy of this document for personal use only. No other licence to any other intellectual property rights is granted herein.

SIP VoIP Release 3.x Settings Application User Guide

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Contents
1 2 3 Introduction................................................................................................................................................ 5 SIP VoIP Settings application.................................................................................................................. 6 VoIP services ............................................................................................................................................... 9 3.1 3.2 3.3 3.4 3.5 Creating, editing, and removing VoIP settings............................................................................ 10 Exporting and importing VoIP services..........................................................................................12 Profile settings ....................................................................................................................................... 16 Used SIP profiles .................................................................................................................................... 20 Codecs ....................................................................................................................................................... 20 3.5.1 3.5.2 3.5.3 3.5.4 3.5.5 3.5.6 3.5.7 4 4.1 4.2 5 6 5.1 6.1 6.2 6.3 7 8 9 AMR NB codec ....................................................................................................................... 22 AMR-WB codec...................................................................................................................... 24 PCMU (G.711 µ-law) codec................................................................................................ 26 PCMA (G.711 A-law) codec ................................................................................................ 26 iLBC codec .............................................................................................................................. 27 G.729 codec ........................................................................................................................... 28 Comfort Noise (CN) codec ................................................................................................. 28

NAT firewall settings...............................................................................................................................29 Domain parameters.............................................................................................................................. 29 4.1.1 STUN settings ....................................................................................................................... 31 IAP parameters....................................................................................................................................... 32 VCC parameters ...................................................................................................................................... 35 SIP profile settings................................................................................................................................ 37 SIP proxy settings.................................................................................................................................. 38 SIP registrar server settings............................................................................................................... 39

VCC settings ...............................................................................................................................................34 SIP settings................................................................................................................................................37

SIMPLE Presence settings.......................................................................................................................40 XDM settings..............................................................................................................................................41 Terms and abbreviations.......................................................................................................................42

10 References .................................................................................................................................................45 11 Evaluate this resource ............................................................................................................................46

SIP VoIP Release 3.x Settings Application User Guide

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Change history
1 October 2008 7 May 2009 26 November 2009 Version 1.0 Version 2.0 Version 2.1 Initial document release Updates throughout the document to describe the use of SIP VoIP Release 3.x Settings 2.0 Updated to describe the use of the settings application on S60 5th Edition devices from Nokia

SIP VoIP Release 3.x Settings Application User Guide

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The guide provides:    An introduction to the SIP VoIP Release 3.x Settings 2. SIP VoIP Release 3. Note: The modification of SIP VoIP settings described in this document should be done using a Nokia S60 VoIP-enabled device.1.0 Configuration Tutorial [1].0 of the application is available in two variants: one for S60 3rd Edition devices that implement Nokia S60 VoIP Release 3. All screen shots and menus are examples only. For more detailed information about the configuration of settings for Nokia S60 VoIP implementations.x Settings Application User Guide 5 .0 and 3. Version 2. A description of application settings.0 application and how to use it.Forum.Nokia.com 1 Introduction This user guide describes the SIP VoIP Release 3. This document is intended for very experienced application developers. A list of features. see the Nokia S60 VoIP Release 3.x Settings application.1 and a second for S60 5th Edition devices that implement Nokia S60 VoIP Release 3. Menu contents and the location of applications and folders may vary according to the device.

The plug-in can be used to both create new VoIP services and modify existing ones.Forum. It can also be used to create NAT firewall (FW) settings and to modify the VoIP parameters. Menu > Control panel > Net settings (see Figure 1). select Menu > Connectivity > Net settings. in some devices. Figure 1: The Net settings application on an S60 3rd Edition device. including the codec and NAT firewall settings. The SIP VoIP Release 3.Nokia. which are normally not visible on the device UI.x Settings application also makes it possible to export the VoIP profile settings to a file or send the settings to another device via Bluetooth connectivity. SIP VoIP Release 3.com 2 SIP VoIP Settings application SIP VoIP Settings is a UI plug-in for the S60 internet communication settings application that can be used to configure VoIP settings.x Settings application on S60 3rd Edition devices. or.x Settings Application User Guide 6 . To access the SIP VoIP Release 3.

com On touch screen devices. Figure 2: The Net settings application on a touch screen device. The user can export VoIP profile settings to a file or send them to another device via Bluetooth connectivity. The user can change the order (priority) of the codecs. The user can manually delete an audio codec. The user can view and modify audio codec settings. select Menu > Settings > Connectivity > Administrative settings > Net settings (see Figure 2). The features provided by the application include: VoIP services Create VoIP profile: Delete VoIP profile: Export settings: Import settings: Edit VoIP profile settings: Select SIP profile: The user can manually create a VoIP profile by using a default or an existing VoIP profile. The user can import VoIP profile settings from a file or from another device via Bluetooth connectivity. Add audio codec: Delete audio codec: Move audio codec: Edit audio codec settings: SIP VoIP Release 3.x Settings Application User Guide 7 .Forum. The user can manually delete a VoIP profile. The user can view the name of the SIP profile used in the VoIP profile and select another SIP profile to be used instead of the current one. The user can view and modify VoIP profile-specific settings. The user can manually add an audio codec.Nokia.

The user can view and modify domain-specific settings.Nokia.Forum. SIP VoIP Release 3. The user can manually delete the domain-specific settings. The user can manually delete the IAP-specific settings.com NAT firewall settings Create domain-specific settings: Edit domain-specific settings: Delete domain-specific settings: Create IAP-specific settings: Edit IAP-specific settings: Delete IAP-specific settings: The user can manually create domain-specific settings.x Settings Application User Guide 8 . The user can view and modify IAP-specific settings. The user can manually create Internet Access Point (IAP)specific settings.

Speech codec settings.Forum. The VoIP profile settings are linked to a network destination created automatically when the user has successfully activated the service from the Contacts application.x Settings Application User Guide 9 . Figure 3: Select Sip VoIP settings Figure 4: Select VoIP services Figure 5: VoIP services available SIP VoIP Release 3.com 3 VoIP services The VoIP profile includes VoIP service-specific settings. select Net Settings > Advanced VoIP settings > VoIP services > Options > Open (see Figure 3 – Figure 5). such as:     VoIP service name.Nokia. To access the VoIP service settings. The VoIP provider name is the same as the VoIP service name shown on the device. VoIP service parameters. SIP profiles used.

the user can add presence settings with default values to the service (see Figure 12 – Figure 14). and removing VoIP settings To create a new VoIP service. the user must create one (see Figure 9 – Figure 11). a SIP profile must be selected for the VoIP profile. If using an existing VoIP profile.Nokia. select Create new service. When creating a VoIP service. Finally. when a SIP profile is selected or created.com 3. or VoIP services > Options > New service > Use default profile/Use existing profile. select one to be copied (see Figure 6 – Figure 8).1 Creating.Forum. Figure 6: Create a new VoIP service Figure 7: Select creation option Figure 8: Select the VoIP profile Figure 9: Create a new SIP profile (or select an existing one) Figure 10: Type full SIP user name and password Figure 11: Ok button becomes visible when full user name with domain is given (Password field is optional) SIP VoIP Release 3. editing.x Settings Application User Guide 10 . If no SIP profiles are available.

select VoIP services > X VoIP profile > Options > Remove (see Figure 15 – Figure 17). Seed value http://localhost is automatically filled into URI field Figure 14: Type XDM URI To delete a VoIP profile.x Settings Application User Guide 11 .Forum.com Figure 12: Create new presence settings Figure 13: XDM URI is needed. Figure 15: Select VoIP profile and Remove Figure 16: Confirm removal by selecting Ok Figure 17: VoIP services available SIP VoIP Release 3.Nokia.

com 3. select the memory and location in which the file will be saved (see Figure 20 – Figure 23). Figure 18: Export settings Figure 19: Select To file If settings are to be exported to a file. The saved file can be used for importing settings later.x device via Bluetooth connectivity.Forum. select VoIP services > X VoIP profile > Options > Export settings (see Figure 18 – Figure 19).x Settings Application User Guide 12 . Figure 20: Select the memory Figure 21: Select the location for the file to be saved Figure 22: Enter name for the file to be saved SIP VoIP Release 3.Nokia.2 Exporting and importing VoIP services To export VoIP service to a file or send to another Nokia S60 VoIP Release 3.

com Figure 23: Export successful If settings are exported to another device via Bluetooth connectivity.Forum.x Settings Application User Guide 13 . select a Bluetooth device from the list (see Figure 24 – Figure 26).Nokia. Figure 24: Select Via Bluetooth Figure 25: Select device from list Figure 26: Settings are being sent SIP VoIP Release 3.

x Settings Application User Guide 14 . Figure 27: Select Import settings If settings are to be imported from a file.com To import VoIP settings from a file or directly from another device. select the memory and location of the file (see Figure 28 – Figure 30).Forum. Figure 28: Select From file Figure 29: Select the memory Figure 30: Select the file to be imported SIP VoIP Release 3.Nokia. select VoIP services > X VoIP profile > Options > Import (see Figure 27).

Nokia. Figure 31: Select Via Bluetooth Figure 32: Settings are being received After a successful import. the Contacts application is automatically launched.x Settings Application User Guide 15 .com If settings are to be imported from another device.Forum. select a Bluetooth device from the list (see Figure 31 – Figure 32). SIP VoIP Release 3.

A number in the range of 0–63.Forum. Quality of Service for VoIP media. DSCP bits) QoS values used in IP headers (IPv4 TOS and IPv6 TC). To modify the VoIP profile-specific settings.1/ OMA-WAP-ProvContv1_1-20021112-C. then it can be edited. the maximum length is 64. The IETF [2] and WMM [3] specifications conflict on the QoS values used for voice packets. IETF RFC 2598. If the VoIP profile was created using the SIP VoIP Release 3. The U-APSD powersave scheme of WMM is also enabled with the IETF default value (46) if the feature is supported by the terminal and the WLAN access point. the user can edit profile settings for provider-specific VoIP service.1/ OMA-WAP-ProvContv1_1-20021112-C.3 Profile settings In this view. an Expedited Forwarding PHB. This text is displayed on the terminal UI as the sender of the settings and cannot be edited.Nokia. Default value: ‘46’. DiffServ Code Point (DiffServ.x Settings application. select X VoIP profile > Profile settings > X parameter > Options > Change (see Figure 33 – Figure 35). Based on the provisioning parameter NAME as described in /R.com 3.x Settings Application User Guide 16 . Figure 33: VoIP profile settings Figure 34: Select the profile parameter to be modified Figure 35: Modify the profile parameter value  Provider name: o o o o o Text.  Profile name: o o  Media QoS: o o o SIP VoIP Release 3. This text is displayed on the device as the service name. Text. Provider of the VoIP profile settings as described in /R. the maximum length is 32.

and out-band DTMF signalling methods are enabled (setting value ‘1’).Nokia. ‘Off’: VoIP over WCDMA is not allowed. Default value: ‘49152’.729.Forum. DTMF tones are sent as compressed audio. the DTMF tones are sent as out-band. as specified in IETF RFC 2833. ‘On’: DTMF digits out-band are generated. Default value: ‘On’. ‘Off’: Disabled. It is not recommended to change this value because if enabled (see below) and if supported by the other peer in the VoIP call.and out-band DTMF should be enabled. If the other end does not support security. Default value: ‘65534’. ‘Use secure only’: Security is mandatory for mobile-originated (MO) call establishment. G. or iLBC) is in use for a VoIP call. DTMF tones are sent as RTP payload. ‘On’: Enabled. ‘Prefer non-secure’: A nonsecure call is preferred. Default value: ‘Prefer nonsecure’. Default value: ‘On’.x Settings Application User Guide . ‘Prefer secure’: A secure call is preferred. they are part of the actual VoIP call audio stream.com  Start media port: o o o An even number in the range of 1024–65530. The value must be at least four over the ‘Start media port number’ to guarantee two simultaneous calls. This setting enables the use of VoIP over WCDMA. The upper limit for the allocated RTP ports. both in. disabling the out-band signalling is required in some special cases. the DTMF out-band mode is used if the peer supports it. if requested by the remote side. Note that the DTMF tones may be degraded if a high-compression rate codec (AMR-NB. ‘On’: VoIP over WCDMA is allowed. Default value: ‘Off’. Parameter that enables media security (secure RTP) if SIP TLS has been used for signalling. ‘Off’: DTMF digits out-band are not generated. a fallback to a nonsecure call takes place. The lower limit of the RTP port range. however. Typically. If both in. An even number in the range of 1028–65534.  End media port: o o o  DTMF in-band: o o o o o  DTMF out-band: o o o o o  Secure call preference (SCP): o o o o o  Allow VoIP over WCDMA (AWCDMA): o o o o  RTCP reporting: 17 SIP VoIP Release 3.

for example.x Settings Application User Guide .164 numbers are used in the user part of the URI. Parameter that defines the meaningful count of caller ID characters for caller identification. ‘On’: The MAC address is appended to the User-Agent header. Parameter that defines the rule for displaying the domain part of an address (URI) in the Logs application. Internet Access Point (IAP). ‘On’: The domain part is never displayed. the maximum length is 32. ‘On’: RTCP reporting is enabled. Parameter that defines the interval for sending a re-SUBSCRIBE message in seconds in the range of 0 – 86400.com o o o o  This setting enables the Real-Time Transport Control Protocol (RTCP) reports defined in RFC 3550.Forum. ‘On’: The terminal type is appended to the User-Agent header. the IAP/SNAP used by the SIP settings is used. to separate two different configurations using different IAPs. Default value: 600. If this parameter is not present. If this parameter is not present. Default value: ‘Off’. ‘0’: All caller ID characters are meaningful. Default value: ‘Off’. ‘Off’: RTCP reporting is disabled. ‘Off’: The terminal type is not appended to the User-Agent header. ‘Numbers only’: The domain part is not displayed if only E. Text. ‘Off’: The MAC address is not appended to the User-Agent header. Default value: ‘On’. the SIP profile used by the VoIP profile is used.Nokia. 18 SIP VoIP User-Agent header: Terminal type display (UATT): o o o  SIP VoIP User-Agent header: WLAN MAC address display (UAWMAC): o o o  SIP VoIP User-Agent header: Free string (UAHS): o o  Count of VoIP digits: o o o o  Ignoring domain part: o o o o o  Voice mailbox settings ID: o  Voice mailbox preferred IAP ID: o  Voice mailbox resubscribe: o o SIP VoIP Release 3. ‘3–20’: Number of meaningful caller ID characters. ‘Off’: The domain part is displayed. Default value: ‘0’. SIP settings name. Default value: ‘Off’. User agent information string that is appended to the SIP User-Agent header.

Parameter that defines the URI for the service provider bookmark. Defines the address to be used in SIP connectivity test calls. Parameter that defines the URI for branding data.example. For example.example. Parameter that defines the session interval for a SIP session in seconds.  Voice mailbox listen address (Vmbx listening URI): o   Presence settings ID: o Auto-accept buddy requests: o o o o  Add ‘user=phone’ to all numbers: o o o o   SIP connectivity test URI: o Provider bookmark URI: o  Minimum value for session interval: o o  SIP session interval: o o  Branding URI: o SIP VoIP Release 3. for example. a value in the range of 0-3600. Parameter that defines the minimum value for a session interval in delta seconds in the range of 90 – 9999. Default value: ‘90’.Nokia. Parameter that defines the rule for automatic acceptance of presence subscription requests.x Settings Application User Guide 19 . A SIP or tel URI (IETF RFC 3261 or 3966) defining the listening address of the user’s IP voice mailbox. Default value: ‘Off’. sip:alice@voicemailbox. ‘On’: Buddy requests are accepted automatically. ‘Off’: Buddy requests are not accepted automatically. For example.com  Voice mailbox address (Vmbx MWI URI): o A SIP or tel URI (IETF RFC 3261 or 3966) defining the Message Waiting Indication (MWI) address of the user’s IP voice mailbox. ‘On’: User=phone is added to all numbers. Default value: ‘1800’. Default value: ‘Off’.Forum.com. sip:alice@listen. Name of the presence settings. ‘Off’: User=phone is not added to all numbers. http://www. Parameter that defines the rule for adding the text string ‘user=phone’ to all SIP URIs containing only numbers as a user name.com.example.com.

Figure 36: View the used SIP profiles Figure 37: Select a SIP profile 3. For codec order.5 Codecs The VoIP profile includes settings for one or multiple speech codecs.Forum.0 Configuration Tutorial [1].1 in the Nokia S60 VoIP Release 3. see Section 3.4 Used SIP profiles This is the SIP profile used by the VoIP profile being created or edited. The VoIP profile can only refer to one SIP profile and one SIP profile can only be used by one VoIP profile. select the profile to be used from the list. select VoIP services > X VoIP profile > Codecs > X codec > X parameter > Options > Change (see Figure 38 – Figure 43). select VoIP services > X VoIP profile > Used SIP profiles (see Figure 36 and Figure 37).Nokia.x Settings Application User Guide 20 . If there are several SIP profiles. To modify the speech codec settings.com 3. To view the name of the SIP profile used in the VoIP profile or to select another SIP profile to be used instead of the current one. Figure 38: Speech codecs Figure 39: Codecs available Figure 40: Open the X codec SIP VoIP Release 3.

Select a new position for the codec (see Figure 45 and Figure 46). select Codecs > X codec > Options > Move (see Figure 44).com Figure 41: Available codec parameters Figure 42: Select the codec parameter to be modified Figure 43: Modify the codec parameter value To change the order of the speech codecs.Forum. Figure 44: Move a speech codec Figure 45: Select a new position for the codec Figure 46: Codec position changed SIP VoIP Release 3.x Settings Application User Guide 21 .Nokia.

Default value: ‘200’. If this parameter is not present. the sender may encapsulate any number of speech frames into one RTP packet. o  MaxPTime: o o o  Jitter buffer: o o SIP VoIP Release 3.Nokia.1  AMR NB codec PTime: o The length of time in milliseconds represented by the media in a packet.x Settings Application User Guide 22 . Default value: ‘200’. The time shall be calculated as the sum of the time the media present in the packet represents. expressed as time in milliseconds. The time should be a multiple of the frame size. and implementations that have not been updated will ignore this attribute. select Codecs > Options > Add/Remove (see Figure 47). Select the codec to be added (see Figure 48 and Figure 49).5. Time in milliseconds. Note that this attribute was introduced after RFC 2327. Figure 47: Add a new speech codec Figure 48: Select the codec to be added Figure 49: A new codec added 3. Default value: ‘20’. which means a 20-ms speech block in one RTP packet. The maximum amount of media that can be encapsulated in each packet. A positive integer (milliseconds) in the range of 20–200. The ptime may vary between the codec’s default ptime and maxptime so that the ptime is increased by the multiples of its allowed values. If other allowed values are not mentioned.Forum. a value in the range of 20–200. the default value and its multiples should be considered as the allowed value.com To create a new speech codec or to delete one.

Default value: ‘Off’. that is. ‘1’: Mode changes are allowed at any time during the session.75 kbit/s ‘1’: AMR 5. ‘2’: Mode changes are allowed every second frame-block.x Settings Application User Guide 23 . a value of two allows the sender to change the mode every second frame-block. Default value: ‘Off’. ‘On’: Enabled. A possible value is a list of modes from the set 0–7 in ascending order. ‘Off’: Disabled.2 kbit/s ‘7’: AMR 12. The initial phase of the interval is arbitrary.90 kbit/s ‘3’: AMR 6.com  Voice activation detection (VAD): o Enables VoIP Discontinuous Transmission (DTX). The value of N shall be either 1 or 2.70 kbit/s (PDC-EFR) ‘4’: AMR 7. Octet-aligned framing used according to RFC 4867. RTP packets are not sent during silent periods.Forum. Neighbouring modes are the ones closest in bit rate to the current mode. ‘Off’: No restriction for used modes. that is. Parameter that defines the requested AMR mode set. mode changes are allowed at any time during the session. N (1 or 2). that is. ‘0’: AMR 4. either the next higher or the next lower rate.15 kbit/s ‘2’: AMR 5. Restricts the active codec mode set to a subset of all modes. but changes must be separated by a period of N frame-blocks.95 kbit/s ‘6’: AMR 10.2 kbit/s (GSM-EFR) Parameter that defines a number of frame-blocks. ‘On’: Enabled. AMR generates Silence Description (SID) packets also during inactivity. that is. ‘1’: Mode changes will only be made to the neighbouring modes in the active codec mode set. o o o  Octet align: o o o  Mode-set: o o o o o o o o o o  Mode-change-period: o o o  Mode-change-neighbour: o o SIP VoIP Release 3. Bandwidth efficient framing is employed. N=1. but the packet frequency is reduced.40 kbit/s ‘5’: AMR 7.Nokia. ‘0’: Change is allowed between any two modes in the active codec mode set. the frame-block period at which codec mode changes are allowed for the sender. If this parameter is not present. ‘Off’: Disabled.

A positive integer (milliseconds) in the range of 20–200. o  MaxPTime: o o o  Jitter buffer: o o  Voice activation detection (VAD): o o o o  Octet align: o o o SIP VoIP Release 3.x Settings Application User Guide 24 . but the packet frequency is reduced. expressed as time in milliseconds. Default value: ‘Off’. no limitation on the use of redundancy is present. The time will be calculated as the sum of the time the media present in the packet represents. The value must be a multiple of the used frame time. Octet-aligned framing used according to RFC 4867. ‘Off’: Disabled. If other allowed values are not mentioned. The ptime may vary between the codec’s default ptime and maxptime so that the ptime is increased by the multiples of its allowed values. Bandwidth efficient framing is employed. The allowed values are between 0 and 100.2 AMR-WB codec The AMR-WB codec is supported from Nokia S60 VoIP Release 3. the default value and its multiples should be considered as the allowed value.Nokia. which means a 20-ms speech block in one RTP packet. AMR generates Silence Description (SID) packets also during inactivity.com  Maxred: o Parameter that 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. Default value: Not used. that is. Default value: ‘200’.Forum. If this parameter is not present. ‘On’: Enabled. RTP packets are not sent during silent periods. the sender may encapsulate any number of speech frames into one RTP packet. ‘20–100’: 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. The time should be a multiple of the frame size. Default value: ‘Off’.1 onwards. Default value: ‘200’. The maximum amount of media that can be encapsulated in each packet. ‘On’: Enabled. and implementations that have not been updated will ignore this attribute. ‘Off’: Disabled.  PTime: o The length of time in milliseconds represented by the media in a packet.5. This parameter allows the receiver to have a bounded delay when redundancy is used. Time in milliseconds. Note that this attribute was introduced after RFC 2327. ‘0’: No redundancy will be used. Default value: ‘20’. o o o 3. If the parameter is omitted. Enabling VoIP Discontinuous Transmission (DTX). a value in the range of 20–200.

Parameter that defines the maximum length of time in milliseconds that elapses between the first transmission of a frame and any redundant transmission that the sender uses.85 kbit/s Parameter that defines a number of frame-blocks. that is.Nokia. ‘0’: Change between any two modes in the active codec mode set is allowed. N=1.  Mode-change-period: o o o  Mode-change-neighbour: o o  Maxred: o o o o SIP VoIP Release 3. Restricts the active codec mode set to a subset of all modes.60 kbit/s ‘1’: AMR-WB 8. either the next higher or the next lower rate. ‘Off’: No restriction for used modes. but changes must be separated by a period of N frame-blocks.25 kbit/s ‘4’: AMR-WB 15. If the parameter is omitted. The initial phase of the interval is arbitrary. ‘1’: Mode changes are allowed at any time during the session.85 kbit/s ‘5’: AMR-WB 18. Neighbouring modes are the ones closest in bit rate to the current mode.x Settings Application User Guide 25 .25 kbit/s ‘6’: AMR-WB 19. The value of N shall be either 1 or 2.05 kbit/s ‘8’: AMR-WB 23. a value of two allows the sender to change the mode every second frame-block. the frame-block period at which codec mode changes are allowed for the sender. The value must be a multiple of the used frame time. mode changes are allowed at any time during the session. ‘0’: AMR-WB 6. N (1 or 2). ‘2’: Mode changes are allowed every second frame-block.65 kbit/s ‘3’: AMR-WB 14. ‘1’: Mode changes will only be made to the neighbouring modes in the active codec mode set. that is. ‘20–100’: The maximum length of time in milliseconds that elapses between the first transmission of a frame and any redundant transmission that the sender uses. If this parameter is not present. ‘0’: No redundancy will be used. that is. The allowed values are between 0 and 100.Forum.85 kbit/s ‘7’: AMR-WB 23.85 kbit/s ‘2’: AMR-WB 12. A possible value is a list of modes from the set 0–8 in ascending order. Default value: Not used. no limitation on the use of redundancy is present. It allows the receiver to have a bounded delay when redundancy is used.com  Mode-set: o o o o o o o o o o o Parameter that defines the requested AMR mode-set.

If this parameter is not present. The ptime may vary between the codec’s default ptime and maxptime so that the ptime is increased by the multiples of its allowed values. Default value: ‘200’. a value in the range of 20–200. but the packet frequency is reduced. The ptime may vary between the codec’s default ptime and maxptime so that the ptime is increased by the multiples of its allowed values. Default value: ‘20’. The time shall be calculated as the sum of the time the media present in the packet represents. expressed as time in milliseconds. Default value: ‘20’. ‘Off’: Disabled. The time shall be calculated as the sum of the time the media present in the packet represents.711 µ-law) codec PTime: o The length of time in milliseconds represented by the media in a packet.5.711 A-law) codec PTime: o The length of time in milliseconds represented by the media in a packet. Default value: ‘200’. The maximum amount of media that can be encapsulated in each packet. o  MaxPTime: o o o SIP VoIP Release 3. o  MaxPTime: o o o  Jitter buffer: o o  Voice activation detection (VAD): o o o o 3. the sender may encapsulate any number of speech frames into one RTP packet. Enabling VoIP DTX.com 3. and implementations that have not been updated will ignore it. If this parameter is not present. The allowed values for this codec are 10 or its multiples. Note that this attribute was introduced after RFC 2327. The maximum amount of media that can be encapsulated in each packet. The time should be a multiple of the frame size.4  PCMA (G. expressed as time in milliseconds. a value in the range of 20–200.5. RTP packets are not sent during silent periods.x Settings Application User Guide 26 . Note that this attribute was introduced after RFC 2327. Time in milliseconds. that is. Default value: ‘200’. Default value: ‘Off’.Nokia.Forum. A positive integer (milliseconds) in the range of 20–200. ‘On’: Enabled. and implementations that have not been updated will ignore it. the sender may encapsulate any number of speech frames into one RTP packet.3  PCMU (G. The allowed values for this codec are 10 or its multiples. Comfort Noise packets are also generated during inactivity if enabled as CN codec. The time should be a multiple of the frame size. Time in milliseconds.

The iLBC codec has support for two basic frame lengths (modes): 20 ms at 15.5  iLBC codec PTime: o The length of time in milliseconds represented by the media in a packet. The time shall be calculated as the sum of the time the media present in the packet represents. the terminal will offer the iLBC with the mode=20 (15. the default value and its multiples should be considered as the allowed value. the sender may encapsulate any number of speech frames into one RTP packet. The ptime may vary between the codec’s default ptime and maxptime so that the ptime is increased by the multiples of its allowed values. Default value: ‘Off’. Default value: ‘30’. a value in the range of 20–200. When the codec operates at block lengths of 20 ms. expressed as time in milliseconds. it produces 304 bits per block.33 kbit/s) if the remote end does not include the mode=20 in its answer. ‘On’: Enabled. ‘Off’: Disabled. Time in milliseconds. Note that this attribute was introduced after RFC 2327.x Settings Application User Guide . and implementations that have not been updated will ignore it.2 kbit/s). Default value: ‘180’. RTP packets are not sent during silent periods. Default value: ‘200’. Default value: ‘200’.5.33 kbit/s.Nokia. Similarly. but will fall back to using the mode=30 (13.com  Jitter buffer: o o A positive integer (milliseconds) in the range of 20–200. ‘On’: Enabled. Default value: ‘30’. The time should be a multiple of the frame size. The allowed values for this codec are 20 and 30 or their multiples. Enables VoIP DTX. A positive integer (milliseconds) in the range of 20–200.2 kbit/s and 30 ms at 13. 27 o  MaxPTime: o o o  Jitter buffer: o o  Voice activation detection (VAD): o o o  Mode-set: o o SIP VoIP Release 3. If the mode-set=20 is configured. Default value: ‘Off’. If this parameter is not present. that is. ‘Off’: Disabled. If other allowed values are not mentioned.Forum. it produces 400 bits per block. Comfort Noise packets are also generated during inactivity if enabled as CN codec. but the packet frequency is reduced. The maximum amount of media that can be encapsulated in each packet. The mode value is configured in the same way as the mode-set in the AMR configuration.  Voice activation detection (VAD): o o o o 3. for block lengths of 30 ms.

o  MaxPTime: o o o  Jitter buffer: o o  Voice activation detection (VAD): o o o  AnnexB: o o o o 3.x Settings Application User Guide 28 . Default value: ‘20’. Enable enhancement according to IETF RFC 3555 annex-b. The maximum amount of media that can be encapsulated in each packet. The time shall be calculated as the sum of the time the media present in the packet represents. Default value: ‘200’. ‘On’: Yes.Nokia. Default value: ‘200’.Forum.5. ‘Off’: Disabled. the sender may encapsulate any number of speech frames into one RTP packet. a value in the range of 10–200. The allowed values for this codec are 10 or its multiples. ‘On’: Enabled. The time should be a multiple of the frame size. and implementations that have not been updated will ignore it. ‘Off’: No. SIP VoIP Release 3. A number. expressed as time in milliseconds.5. Default value: ‘Off’. Time in milliseconds.6  G. Note that this attribute was introduced after RFC 2327.729 codec PTime: o The length of time in milliseconds represented by the media in a packet. If other allowed values are not mentioned. If this parameter is not present. Default value: ‘Off’. The ptime may vary between the codec’s default ptime and maxptime so that the ptime is increased by the multiples of its allowed values. PCMA or iLBC is enabled.com 3. the default value and its multiples should be considered as the allowed value. A positive integer (milliseconds) in the range of 20–200.7 Comfort Noise (CN) codec This codec is typically included if PCMU.

x Settings Application User Guide 29 . select Advanced VoIP Settings > NAT firewall settings > Options > Open (see Figure 50 and Figure 51). select Domain parameters > Create parameters > Select domain (see Figure 52 – Figure 54). NAT firewall settings by default refer to SIP domain-specific settings. Figure 50: NAT firewall settings Figure 51: Domain and IAP parameters 4. With the SIP VoIP Settings application.Nokia. if IAP-specific values are defined. Refresh timers are overridden with IAP-specific NAT firewall settings. Figure 52: Create domain parameters Figure 53: Select the SIP domain Figure 54: X domain parameters created SIP VoIP Release 3. the user can also create the STUN settings. To access the NAT firewall domain or IAP-specific settings.com 4 NAT firewall settings Nokia S60 VoIP implementation has Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) (STUN) support for NAT traversal and NAT binding refresh features.1 Domain parameters To create domain parameters.Forum.

A number in the range of 0–9999. the user can change STUN settings. TCP NAT bind. select X domain > X parameter > Options > Change (see Figure 55 –Figure 57) or X domain > STUN Settings > X parameter > Options > Change (see Figure 58 – Figure 60).com To modify the domain-specific settings. Parameter that defines the NAT refresh interval for TCP in the domain-specific NAT-FW settings. The unit of the refresh interval is seconds. it overrides this value.x Settings Application User Guide 30 . Default value: ‘1200’. If an IAP-specific value for this interval is defined. Figure 55: Open the X domain Figure 56: Select the domain parameter to be modified Figure 57: Modify the domain parameter value Figure 58: Open the STUN settings Figure 59: Select the parameter to be modified Figure 60: Modify the parameter value   STUN settings: o In this view. refresh: o o o SIP VoIP Release 3.Forum.Nokia.

Though optional. Needed for STUN server authentication with long-term credentials. If an IAP-specific value for this interval is defined. ‘On’: STUN shared secret mechanism is used.0 to disable the STUN servers.1. Default value: ‘nokia.stun’. Short-term credentials that do not need to be configured are generally used.x Settings Application User Guide 31 .com  UDP NAT bind. Default value: ‘Off’. Needed for STUN server authentication with long-term credentials. If no address is defined. Parameter that defines the STUN server port in the domain-specific NAT-FW settings.Forum. Parameter that indicates whether the STUN server supports TLS and shared secret or not. o  CRLF refresh: o o o o  Used NAT protocol: o o 4. ‘On’: Enabled.0.  STUN server port: o o  STUN server user name: o   STUN server password: o STUN shared secret: o o o o SIP VoIP Release 3. it overrides this value. ‘Off’: Disabled.Nokia. refresh: o o A number in the range of 0–9999. Parameter that defines the usage of CRLF-based NAT binding refresh. Optional. for example. The unit of the refresh interval is seconds. the UE tries to resolve the STUN server using the DNS SRV query. or if the SIP proxy requires refresh to keep the persistent TCP/TLS connection alive. Default value: ‘3478’.1  STUN settings STUN server address: o o Parameter that defines the STUN server address in the domain-specific NAT-FW settings.0. ‘Off’: STUN shared secret mechanism is not used. Apply value 0. enabling is strongly recommended if there is either a NAT or firewall on the route. Parameter that defines the NAT protocol used. Default value: ‘28’. Default value: ‘Off’. Parameter that defines the NAT refresh interval for UDP in the domain-specific NAT-FW settings. The attribute enables the CRLF refresh to the outbound proxy (or to the registrar if no proxy is defined) over any transport. Optional. if an SBC is taking care of the NAT traversal.

The upper limit of the domain port range. The lower limit of the domain port range. Default value: ‘49152’. In any case. o  End port number: o o o 4. To create IAP parameters or to modify the IAP-specific settings. In any case.com  Start port number: o o A number in the range of 49152–65535. Default value: ‘65535’.Nokia. only those IAPs with defined parameters are available. A number in the range of 49152–65535. Do not edit this value unless for a specific reason. select IAP parameters > X access point > X parameter > Options > Create parameters/Open (see Figure 61 – Figure 66). do not set the port range to zero because using the same value for start and end port numbers will prevent all VoIP calls. Do not edit this value unless for a specific reason.2 IAP parameters As the default.x Settings Application User Guide 32 . Figure 61: Select the IAP parameters Figure 62: Create new IAP-specific NAT-FW parameters Figure 63: Select the access point for which settings are to be created SIP VoIP Release 3. do not set the port range to zero because using the same value for start and end port numbers will prevent all VoIP calls.Forum.

Nokia.com Figure 64: Open the settings to be modified Figure 65: Select the IAP parameter to be modified Figure 66: Modify the IAP parameter value  TCP NAT bind. Default value: ‘28’. The unit of the refresh interval is seconds. o  UDP NAT bind. if it is defined. Default value: ‘250’. Parameter that defines the NAT refresh interval for UDP in the IAP-specific NAT-FW settings. A number in the range of 0–9999. refresh: o o o  STUN retransmission: o o o SIP VoIP Release 3.x Settings Application User Guide 33 .Forum. if it is defined. The unit of the refresh interval is seconds. Parameter that defines the STUN request retransmit timer (time in milliseconds) in the IAPspecific NAT-FW settings. The value overrides the domain-specific NAT Refresh TCP value. The value overrides the domain-specific NAT Refresh UDP value. Parameter that defines the NAT refresh interval for TCP in the IAP-specific NAT-FW settings. A number in the range of 0–9999. Default value: ‘1200’. refresh: o o A number in the range of 0–9999.

To access VCC settings.x Settings Application User Guide 34 .1 introduces VCC (Voice Call Continuity) settings.Forum.Nokia. Figure 67: Select the VCC settings Figure 68: Create new VCC settings Figure 69: Select editable setting Figure 70: Edit a setting SIP VoIP Release 3.com 5 VCC settings Nokia S60 VoIP Release 3. the plug-in asks to create settings with default values when the VCC settings view is opened (see Figure 67 – Figure 68). these settings can be edited using SIP VoIP Settings for the Nokia S60 VoIP Release 3 plug-in. In devices that have VCC enabled. If there are no VCC settings available. select Options > Change or hit the middle softkey (see Figure 69 – Figure 70). select Advanced VoIP settings > VCC settings (see Figure 67. To edit a single setting.

The operator’s preferred domain for UE-originated calls/sessions. ‘Off’ = Not allowed. Tells whether VCC domain transfer from CS to PS is allowed or not.com 5. VDN: o  Name of the VoIP service. ‘3’ = Preference for IM CN subsystem only. Default value: ‘On’. Tells whether VCC domain transfer from PS to CS is allowed or not. ‘1’ = Preference for IM CN subsystem.Nokia. Indicates whether to initiate a VCC domain transfer immediately to the operator’s preferred domain when that domain becomes available.1  VCC parameters VoIP service: o   VDI: o Current VCC transfer URI that is included in the SIP INVITE request to initiate domain transfer. then it is up to the VCC UE to decide when to perform the domain transfer if the preferred domain is available. This policy only affects ongoing sessions.Forum. ‘On’ = Allowed. ‘2’ = Preference for CS domain only. Current VCC transfer number that the user includes in a circuit switched call setup to initiate domain transfer. Preferred domain: o o o o o o  Immediate DT: o o o o o  CS to PS allowed: o o o o  PS to CS allowed: o o o o SIP VoIP Release 3. One VoIP service at a time can use VCC settings. ‘Off’ = Not allowed. Default value: ‘On’.x Settings Application User Guide 35 . ‘0’ = Preference for CS domain. ‘On’ = Initiate domain transfer immediately to the preferred domain when that domain becomes available. and the selected one is shown here. Default value: ‘On’. ‘On’ = Allowed. If the Immediate Domain Transfer operator policy indicates that the domain transfer is not immediately required. ‘Off’ = Do not initiate domain transfer immediately to the preferred domain when that domain becomes available. Default value: ‘1’.

Forum.Nokia. DT will be committed. Indicates the threshold value in dBm for CS signal strength from which DT starts. Default value: ‘Off’.com  Waiting calls allowed: o o o o Indicates whether VCC domain transfer is allowed during a waiting call. DT will be committed. DT will not be committed. The amount of time in microseconds GSM signal must be above CS HO threshold for the signal to be interpreted as ‘good’. The amount of time in microseconds WLAN signal must be below WLAN HO threshold for the signal to be interpreted as ‘bad’. The amount of time in microseconds WLAN signal must be above WLAN HO threshold for the signal to be interpreted as ‘good’. Hysteresis value in dBm for CS signal. ‘On’ = Allowed ‘Off’ = Not allowed. Defines the hysteresis value in dBm for WLAN signal. DT will not be committed. Indicates the threshold value in dBm for WLAN signal strength from which domain transfer starts. Default value: ‘On’. ‘Off’ = DT is not allowed.  WLAN HO threshold: o   WLAN HO hysteresis: o WLAN HO hysteresis low: o  WLAN HO hysteresis high: o    CS HO threshold: o CS HO hysteresis: o CS HO hysteresis low: o  CS HO hysteresis high: o  Allow CS originated: o o o o SIP VoIP Release 3. ‘On’ = DT is allowed.x Settings Application User Guide 36 . The amount of time in microseconds GSM signal must be below CS HO threshold for the signal to be interpreted as ‘bad’. Indicates whether domain transfer is allowed when the call is CS originated.

Forum.Nokia. Service profile: o o  Default destination: o  Default access point: o  Public username: o o  Use compression: o  Registration mode: o  Security negotiation: o o   Proxy server: o Registrar server: o SIP VoIP Release 3. Tells the registration mode. With 3GPP. If used. this setting is not in use. More information about configuring SIP can be found in [1]. use the prefix ‘sips:’. Signalling compression can be used with a cellular radio to reduce the data generated by SIP signalling. A selection between IETF and 3GPP SIP dialects. Access point used with this SIP profile. Configuration is needed if an outbound proxy is used. 6. A destination network can contain multiple access points. For example.com.doe@example.x Settings Application User Guide 37 . the SIP domain part. ‘Always on’ is highly recommended with VoIP service because using ‘Always on’ affects only SIP registration and not VoIP service activation. Requires a sec-agree support from the SIP server side.com 6 SIP settings Below is a list of configurable SIP settings. use ‘IMS’. sip:john. A SIP URI for the user including the host-name part. It requires an outbound proxy support and may cause error situations in poor WLAN coverage.1   SIP profile settings Profile name: o Name of the SIP profile. this setting overrides the default access point setting. Note that always when registering a SIPS URI. that is. Default value: ‘Internet’ (IETF). Used mainly with IMS in 3GPP mode. Destination network used with this SIP profile. a persistent TLS connection will be created towards the proxy (or registrar if no proxy is defined). If a default destination is defined. To enable registering a SIPS URI. Configuration is always needed. and is thus not recommended with WLAN. Default value: ‘Inactive’.

resolving the related IP address will be done with the procedures specified within RFC 3263 [4]. the terminal will try to resolve it using only A and AAAA queries. If the FQDN is supplied without a port parameter but with a transport parameter. This value is only needed if the proxy and registrar require different credentials. ‘UDP’ – UDP transport forced to be used. ‘Auto’ – Transport selected according to RFC 3261 [5]. the terminal will try to resolve it using the DNS SRV and. o o o o  Apply value 0. A possible transport parameter on the next hop in the Record-Route or Contact overdrives this setting.doe john. If the proxy server is defined as FQDN. The proxy has to insert Record-Route headers to keep itself in the route set on later requests inside a SIP session.0 to enable the discovery of SIP proxy using DHCP option 120.2  SIP proxy settings Proxy address: o o Address of the SIP outbound proxy. finally. A and AAAA queries. SRV.doe@example. the terminal will try to resolve it using the DNS NATRP.Nokia.com 6.x Settings Application User Guide 38 .If the FQDN is supplied without port and transport parameters. Used in proxy authentication (not necessarily the same as the public user name). This setting affects all initial requests. A and AAAA queries. the Route header is inserted in the initial request. To enable SIP over TLS towards the proxy. Needed if proxy authentication is used. This setting creates a preroute set according to RFC 3261 [5]. that is.com Depends on the proxy vendor and configuration. and. If the FQDN is supplied with the port parameter.0. finally.Forum.0.doe@example. Username: o o  Password: o o  Loose routing: o o  Transport protocol: o o o o SIP VoIP Release 3. this setting must be left empty. UDP is used for ≤ 1300 bytes. use the prefix ‘sips:’. and TCP for > 1300-byte-long initial requests. The transport is TLS (RFC 3263 [4]) for SIPS. but it may also contain the SIP domain or the SIP scheme as a prefix. This value is only needed if the proxy and registrar require different credentials. If outbound proxy is not used. The authentication user name may simply be the user name part of the public SIP URI. ‘No’ for SIP 1. ‘Yes’ for RFC 3261-compliant SIP proxy. A short summary is as follows: . ‘TCP’ – TCP transport forced to be used. that is: john.com sip:john.0-specification-compatible strict routing mode.

User name: o  Password: o  Transport protocol: o o o  Port: o SIP VoIP Release 3. that is. Used in user authentication. ‘Auto’ – Transport selected according to RFC 3261 [5]. UDP is used for ≤ 1300 bytes. Typically the same value is also used for the proxy authentication. The transport may also be TLS (RFC 3263 [4]). 6. ‘TCP’ – TCP transport used for the REGISTER request sent to the registrar and forces the use of TCP transport for all initial requests in cases where an outbound proxy is not defined. Often. Used in user authentication. the default value of 5060 is typically used. that is. ‘UDP’ – UDP transport is forced to be used if selected and if an outbound proxy is not defined.Forum. the same value is also used for the proxy authentication. TCP and/or UDP port to which the SIP registrar is listening. This parameter is often called a private user ID. use the prefix ‘sips:’.com  Port: o TCP and/or UDP port to which the SIP proxy is listening. the host-name part of the user’s SIP URI. Normally. the same as the user name of the public SIP URI.3  SIP registrar server settings Registrar address: o o  IP address or FQDN of the registrar server on the SIP domain. To enable SIP over TLS towards the registrar.x Settings Application User Guide 39 . the default value of 5060 is typically used. but not necessarily always.Nokia. and TCP for > 1300-byte-long initial requests.

The SIP profile used for the presence settings. Default value of 3600 is recommended. A value in the range of 0–999999. Default value of 5120 is recommended. @domain. If this value is left empty. for example. The XDM settings used for the presence settings. A value in the range of 0–9999.    Server name: o Name of the presence server. The available profiles are presented on a list. A value in the range of 0–999999999.Nokia. If this value is left empty. contacts in list: o  Domain syntax: o SIP VoIP Release 3. The maximum number of contacts in the subscription list. The syntax of the service URI. The maximum number of presence subscriptions. the number of contacts is unlimited. the number of subscriptions is unlimited.Forum. The available XDM settings are presented on a list.com 7 SIMPLE Presence settings Below is a list of configurable SIMPLE Presence settings. A value in the range of 0–9999. The maximum size of the MIME object data in bytes. The minimum time interval in seconds between two consecutive publications. More information about configuring SIMPLE Presence can be found in [1]. SIP profile: o XDM settings: o  Object size: o o  Publish interval (sec): o o  Maximum subscriptions: o  Max.x Settings Application User Guide 40 .

x Settings Application User Guide 41 .   XDM set name: o The name of the XDM setting set. Required for the XDM settings authentication. Access point to be used with these XDM settings. It is recommended that this be left empty and that the VoIP service be linked to a destination network. More information about configuring XDM can be found in [1].Forum. URI of the XDM server. Access point: o   XCAP service address: o HTTP user name: o SIP VoIP Release 3.com 8 XDM settings Below is a list of configurable XDM settings.Nokia.

x Settings Application User Guide Meaning Name of G.com 9 Terms and abbreviations Term or abbreviation a-law AMR CRLF CN CN CRLF CS dBm DHCP DiffServ DNS DSCP DT DTMF DTX FQDN FW GSM GSM-EFR HO HTTP IAP ID IEEE IETF IM CN iLBC IMS IP IPv4 IPv6 kbit/s SIP VoIP Release 3. version 6 Kilobits per second 42 . Dynamic Host Configuration Protocol Differentiated Service Domain Name System DiffServ Code Point Domain Transfer Dual-Tone Multi Frequency Discontinuous Transmission Fully Qualified Domain Name Firewall Global System for Mobile communications Global System for Mobile communications – Enhanced Full Rate Handover Hypertext Transport Protocol Internet Access Point Identity Institute of Electrical and Electronics Engineers (www.Forum.org) The Internet Engineering Task Force (www. version 4 Internet Protocol.org) IP Multimedia Core Network Internet Low Bitrate Codec IP Multimedia System Internet Protocol Internet Protocol.ieee.711 PCMU algorithm (European) Adaptive Multi Rate Carrier Line Feed Comfort Noise (in codec settings) Core Network (in VCC settings) Carrier Line Feed Circuit Switched Power ratio in decibels (dB) of the measured power referenced to one milliwatt (mW).ietf.Nokia.

Nokia.openmobilealliance. Telephony Traffic Class Transport Control Protocol Transport Layer Security Type of Service Unsolicited Automatic Power Save Delivery User Datagram Protocol User Equipment User Interface Uniform Resource Identifier Voice Activation Detection Voice Call Continuity VCC Domain transfer URI SIP VoIP Release 3.org) Pulse Code Modulation a-law Pulse Code Modulation µ-law Personal Digital Cellular – Enhanced Full Rate Packetization interval Per-Hop forwarding Behaviour Packet Switched Quality of Service Request For Comments Real-time Transport Control Protocol Real-Time Transport Protocol Session Border Controller Silence Description SIP Instant Message and Presence Leveraging Extensions Session Initiation Protocol Session Traversal Utilities for NAT.Forum.x Settings Application User Guide 43 .com Term or abbreviation Maxptime MIME NAT NB OMA PCMA PCMU PDC-EFR Ptime PHB PS QoS RFC RTCP RTP SBC SID SIMPLE SIP STUN tel TC TCP TLS TOS U-APSD UDP UE UI URI VAD VCC VDI Meaning Maximum amount of media that can be encapsulated in a payload packet Multipurpose Internet Mai Extensions Network Address Translation / Network Address Translator Narrow Band Open Mobile Alliance (www. a protocol that allows applications to detect that network address translation (NAT) is being used.

711 PCMU algorithm (North American) SIP VoIP Release 3. Wireless Local Area Network Wireless Multimedia XML Configuration Access Protocol XML Document Management Name of G.Nokia.x Settings Application User Guide 44 .Forum.com Term or abbreviation VDN VoIP WB WLAN WMM XCAP XDM µ-law Meaning VCC Domain transfer Number Voice over IP Wide Band Wireless LAN.

Forum.0 Configuration Tutorial.ietf.11e.org SIP VoIP Release 3. www.x Settings Application User Guide 45 .forum.com IETF RFC 2598. http://www. SIP: Locating SIP Servers. www.org IETF RFC 3263. http://www.Nokia.com 10 References [1] [2] [3] [4] [5] Nokia S60 VoIP Release 3.org IEEE Standard 802. An Expedited Forwarding PHB. SIP: Session Initiation Protocol.nokia. Wireless Multimedia (WMM) Specification.ieee.ietf.ietf.org IETF RFC 3261. www.

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