HP iPAQ 512 Voice Messenger User's Guide
Contents
1. Call Hold Originator and Terminator hold Music on Hold and resume with Endpoint Extension and PSTN 7 5 COMPATIBILITY WITH ALCATEL OMNIPCX ENTERPRISE V6 1 Feature Brief Description Pass Fail Comments mea e fee Extension and PSTN Calls k PSTN 2 Nea andre a and PSTN send and receive DTMF tones 2 Nor and eoe Sooo and PSTN send and receive DTMF tones Call Forward Call Forward All CFA Busy Pass Requires SIM card and CFB and Not Answered accepts valid E 164 CFNA numbers only Indicator from server to Endpoint test system transfer to Endpoint Extension and PSTN Call Waiting On Originator and Terminator Pass Requires SIM card to waiting timeout release before configure answer CLIP Type and Il configure Call Barring Calling Line ID Restriction Requires SIM card to CLIR configure Direct Inward Dial DID from Endpoint to Local and Remote Extension and PSTN to Endpoint Call Hold Originator and Terminator hold Pass Music on Hold and resume with Endpoint Extension and PSTN 8 Accessory Support This category is related to headsets and other accessories that may be used with the handset in conjunction with VoIP Supported e Wired headsets e Mono Bluetooth headsets Not Supported e Stereo Bluetooth headsets 9 Configuring VoIP The Phone Dialer on the HP iPAQ 500 series Voice Messenger can be used to make VoIP calls in addition to normal cellular calls VoIP can be confi2. sip 1 2 3 host gt Please refer to the Windows Mobile 6 documentation for additional details about the VoIP Dial Plan and some examples 9 5 ADDITIONAL CONFIGURATION Windows Mobile 6 provides a set of registry keys to customize the WM6 SIP Client Some of the key customization features are provided by HP iPAQ Setup Assistant as explained in section 9 3 The other customization options can be set by modifying the registry Below are some examples of the configuration options provided using the registry Enable disable redirection of SIP calls Allow disallow listening for incoming SIP traffic on port 5060 Modify the registration expiry time default is 19 seconds Modify Session Timers specifically Session Expires and Min SE default is 90 seconds Please refer to the Windows Mobile 6 documentation for complete coverage of all available configuration options and the associated registry entries 10 Appendix A Standards Support 10 1 SIGNALING STANDARDS IMPLEMENTED Standard or Reference Title Decription RFC 3261 SIP Session Initiation Protocol RFC 3261 Call Waiting RFC 2327 SDP Session Description Protocol RFC 2246 TLS Protocol RFC 3263 Session Initiation Protocol SIP Locating SIP Servers RFC 3263 obsoletes 2543 RFC 3264 An Offer Answer Model with the Session Description Protocol SDP RFC 3262 Reliability of Provisional Responses Provisional Response ACKnowledgement P
3. HP iPAQ Setup Assistant cannot be used to make any significant changes to the dial plan It can only be used to add a prefix digit to 11 digit 10 digit and 7 digit outgoing phone numbers It is HIGHLY RECOMMENDED that any modifications additions deletions to the dial plan should be done using any of the provisioning methods mentioned above HP iPAQ Setup Assistant MUST only be used to configure modify SIP settings 9 4 1 IMPORTANT NOTE ABOUT EMERGENCY CALLING Emergency Calling over IP is not supported by the Voice Messenger All emergency calls must be placed over cellular To ensure this a dialing rule specifically for emergency calling MUST be added to the VoIP dial plan This rule will indicate what the emergency number is and will also indicate that this number should be placed on the cellular network only The emergency dialing rule for the U S is shown below lt rule pattern 911 display 911 restrict VoIP gt The base template for the VoIP Dial Plan is specified below It is HIGHLY RECOMMENDED that the rules included in the base template should not be removed lt wap provisioningdoc gt lt characteristic type VoIP gt lt parm name DialPlan value lt dialplan xmlins http schemas microsoft com embedded VoIP gt lt dialplan header gt lt host gt use_sipsrv_host_name lt host gt lt dialplan header gt lt Dial Plan rules gt lt IP address rules gt lt EQUIVALENT OF d 1 3 d 1
4. Reference Decription Dratft ietf core 23 Draft ietf xmpp cpim 04 RFC 4028 Session Timers in the SIP RFC 3611 RTP Control Protocol Extended Reports RTCP XR RFC 3389 RTP Payload for Comfort Noise CN RFC 3311 Draft ietf sip join xx SIP Join header Dratt ietf sipping cc conferencing xx SIP Call Control Conferencing for User Agents RFC 3891 SIP Replaces Header RFC 3266 Support for IPv6 in Session Description Protocol SDP RFC 3581 Symmetric Routing Call Park Call Pickup Call Park Call Pickup through Replaces header RFC 3903 Draft ietf sup publish 04 Extension to SIP event notification framewirking for aggregation of notification under the same AOR UPnP IDG Internet Gateway Device IGD Standardized Device Control Protocol V 1 0 RFC 3047 RTP Payload Format for ITU T Recommendation G 722 1 G 723 Supported if a 3rd party codec is used G 723 1 Supported if a 3rd party codec is used iLBC Supported if a 3rd party codec is used G 726 Supported if a 3rd party codec is used G 729 Supported if a 3 party codec is used SIP Auth AKA AKA Auth based on USIM
5. 3 d 1 3 d 1 3 gt lt rule pattern d d d d d qd d d d d d qd d d d d d qd d d d d d d restrict Cell SMS gt lt EQUIVALENT OF d 1 3 d 1 3 Ad 1 3 d 1 3 gt lt rule pattern d d d d d d d d d d d d d d d d d d d d d d d d dial 1 2 3 4 display 1 2 3 4 restrict Cell SMS gt lt Add Emergency Dialing Rules here gt lt Add Other Dialing Rules here gt lt SIP URI rules gt lt EQUIVALENT OF Ss li Pp Ss w d 3 d 3 d 4 gt lt rule pattern Ss li Po Ss w d d d d d d d d d d display 1 2 3 restrict Cell SMS gt lt rule pattern Ss li Pp Ss w a zA Z0 9_ display 2 restrict Cell SMS gt lt rule pattern Ss li Pp Ss w display 1 restrict Cell SMS gt lt Catch All gt lt rule pattern d dial sip 1 host display 1 transfer sip 1 host gt lt rule pattern a zA Z0 9_ dial sip 1 host display 1 transfer sip 1 host gt lt dialplan gt gt lt characteristic gt lt wap provisioningdoc gt A sample dialing rule for 10 digit phone numbers is shown below This rule pre pends a 9 to 10 digit dialed phone numbers lt rule pattern d 3 s d 3 s s d 4 s Xx s d dial sip 9 1 2 3 host display 1 2 3 transfer
6. RACK RFC 3361 DHCP Option for location the outbound SIP Proxy server RFC 3265 Session Initiation Protocol SIP Specific Event Notification SUBSCRIBE NOTIFY RFC 3842 A Message Summary and Message Waiting Indication RFC 3863 Presence Information Data Format PIDF RFC 2976 The SIP INFO Method RFC 3428 Session Initiation Protocol SIP Extension for Instant Messaging Secure RTP SRTP RFC 3856 Presence Event Package for the Session Initiation Protocol SIP RFC 3515 The Session Initiation Protocol SIP Refer Method RFC 3515 SIP Refer Method Call Transfer RFC 3323 Privacy Mechanism for SIP RFC 3324 Short term requirements for Network asserted identit Private extensions to SIP of asserted identify within trusted networks MD5 RTCP nSOP RFC 2605 RTCP in SDP RFC 1321 RFC 2474 RFC 2617 obsoletes 2069 An Extension to HTTP Digest Access Authentication 10 2 MEDIA OTHER STANDARDS Standard or Reference Description RFC 3550 obsoletes RFC 1889 RTP A Transport Protocol for Real Time Applications RFC 3551 obsoletes RFC 1889 RTP Profile for Audio and Video Conferences with Minimal Control draft wing behave symmetric rtprtcp 01 txt Symmetric RTP RTCP ports over UDP RFC 2833 RTP Payload for DTMF Digits Telephony Tones and Telephony Signals in band and out of band G711 WA RFC 2198 RTP Payload for Redundant Audio Data 10 3 STANDARDS NOT IMPLEMENTED Standard or
7. Voice over IP VoIP Application Note HP iPAQ 500 series Voice Messenger The goal of this document is to clearly and concisely state what HP iPAQ 500 series Voice Messenger is and is not capable of supporting for mobile IP telephony aka VoIP or Voice over IP All the capabilities and interoperability scenarios haven t been tested at this point and until then this document will be a work in progress As new information becomes available it will be added to this document Contents DON OE pate sais teers etna etna eters AE eee 2 2 Media and Signaling Protocols css sssessecccccccccsssssssssssssesceseacscccccccceesessessessees 3 3 Telephony PAU SS i ec nidncrrvanasioxavaneencndevacenscustcnedecssnanuivassiseavtonhivevereliacgalisesvesdseostorstes 3 A NNT RUN E EER 4 5 Hardware and Piri Ware csser ain E 5 GO WLAN TITAS UC LIS siccin e EE E i 6 7 IPPBX and SIP Server SUppoti senoir aE N 6 7 1 Compatibility with Cisco unified Call Manager v5 1 oossnoessesssessessessseserreerssees f 7 2 Compatibility with Avaya Communication Manager v4 0 SES v3 1 2 8 7 3 Compatibility with Nortel MCS 5100 V4 5 oo ceeccceesssseneseceeeeeeeeceeeeseneeees 9 7 4 Compatibility with Nortel CS1000 V4 5 wo ccccccccsesseeesenseneeeeeeeeeeeeeeeeeeees 10 7 5 Compatibility with Alcatel OmniPCX Enterprise V6 1 cccceeceeeeeeeeeeees 11 gt PCOS SS OR DUP PON eein E EE 12 D COE y O e E A E O 12 9 1 Enabling Internet Calling sirensis ENE a E 12 o
8. and CFB and Not Answered accepts valid E 164 CFNA numbers only Message Waiting Voicemail notification passed Pass Indicator from server to Endpoint Blind Transfer Originator and Terminator Pass transfer to Endpoint Extension and PSTN Call Waiting On Originator and Terminator Pass Requires SIM card to answer Caller ID Calling Line ID Presentation Pass Requires SIM card to CLIP Type and Il r configure Call Barring Calling Line ID Restriction Pass Requires SIM card to CLIR configure Remote Extension and PSTN to Endpoint Call Hold Originator and Terminator hold Pass Music on Hold and resume with Endpoint Extension and PSTN 7 4 COMPATIBILITY WITH NORTEL CS1000 V4 5 Feature Brief Description Pass Fail Comments ae Ete N al Tena Cals From iP and PSTN oe PSTN DTMF in band SIP End Point Extension and PSTN send and receive DTMF tones DTMF RFC 2833 SIP End Point Extension and PSTN send and receive DTMF tones Call Forward Call Forward All CFA Busy CFB and Not Answered CFNA Message Waiting Voicemail notification passed Indicator from server to Endpoint Blind Transfer Originator and Terminator transfer to Endpoint Extension and PSTN Call Waiting On Originator and Terminator waiting timeout release before answer Caller ID Calling Line ID Presentation CLIP Type and Il Call Barring Calling Line ID Restriction CLIR Remote Extension and PSTN to Endpoint
9. astructure testing and results may be available in the future For optimal performance all 802 11b and 802 11g data rates should be enabled on the APs Limiting the data rates on the APs may prevent iPAQ 500 series devices from connecting to the AP 7 IP PBX and SIP Server Support This category is related to the SIP based IP PBX system or server with which the handheld device must interact for VoIP services HP testing is underway with enterprise IP PBX products from Cisco Avaya Alcatel and Nortel Detailed test criteria and results will be added to this section as they become available Results of other additional testing with operator class SIP servers Broadsoft Sylantro Huawei Alcatel etc will be included when they are made available to HP 7 1 COMPATIBILITY WITH CISCO UNIFIED CALL MANAGER V5 1 Feature Brief Description Pass Fail Comments Extension and PSTN Terminate Calls From SIP End Point Extension Pass SIP End Point Extension and PSTN send and receive DTMF tones SIP End Point Extension and PSTN send and receive DTMF tones Call Forward Call Forward All CFA Busy Pass Requires SIM card and es CFB and Not Answered ail accepts valid E 164 CFNA numbers only Message Waiting Voicemail notification passed Cisco uses SIP Notify Indicator from server to Endpoint Enhancement request submitted to Microsoft Blind Transfer Originator and Terminator Pass transfer to Endpoint Extension and PSTN Call Waitin
10. elease e Certain features of 802 11e WMM are not currently supported o Automatic Power Save Delivery APSD and Unscheduled Automatic Power Save Delivery U APSD power saving mechanisms are not currently supported o Even though packet tagging 802 1p and Diffserv is inherently supported by Windows Mobile 6 there may not be any benefit by tagging voice packets with a higher priority 6 WLAN Infrastructure This category is related to the wireless local area network WLAN infrastructure to which the handset connects to send and receive IP traffic The key consideration from the mobile handset perspective is typically AP AP roaming Given the real time requirements of voice and the delays inherent in WLAN authentication 802 111 the strongly recommended answer is a VoIP enabled pervasive enterprise WLAN infrastructure Note that HP Services has a well developed practice for migrating customers WLANs to this model The baseline requirement here is a controller based AP deployment not standalone APs Examples of these controller based WLAN product offerings include e Cisco Aironet APs with Wireless LAN controller s e HP ProCurve Radio Ports plus Wireless Edge Services xl module s in 5300x1 switch chassis e Extreme Networks Altitude APs and Summit switches e Aruba Networks APs and Mobility Controller s e Meru Networks APs and Controller s Testing is underway with Cisco Aironet APs and wWireless LAN controllers Other WLAN infr
11. fied as the prefix digit it is automatically pre pended to all 11 digit 10 digit and 7 digit phone numbers By default no prefix digit is used The DSCP field in the IP header of voice packets contains a QoS number that determines the priority voice packets get over data packets in the network By default 56 is used as the DSCP QoS number This can be modified to increase or decrease the priority of voice traffic Once all the appropriate information is provided apply the configuration to the phone If Wi Fi is on and connected to an AP the phone will register with the specified SIP enabled IP PBX or the SIP Server If Wi Fi is off SIP registration will occur when the next time Wi Fi is turned on and connected to an AP The status of the Internet Calling plug in will either be Available or Selected Available means that the phone has Wi Fi access and is registered with the IP PBX but the next outgoing call will be over cellular Selected means that the phone has Wi Fi access is registered with the IP PBX and that VoIP will be used for the next outgoing call The Phone Dialer can be used to make and receive VoIP calls The Internet Calling status can be toggled between Available and Selected by pressing the Action Enter button on the Internet Calling plug in on the home screen If the status of the plug in is No Service it indicates that the registration was unsuccessful This could be due to
12. g On Originator and Terminator Pass Requires SIM card to waiting timeout release before configure answer Caller ID Calling Line ID Presentation Pass Requires SIM card to CLIP Type and Il configure CLIR configure Remote Extension and PSTN to Endpoint Call Hold Originator and Terminator hold Pass Music on Hold and resume with Endpoint Extension and PSTN 7 2 COMPATIBILITY WITH AVAYA COMMUNICATION MANAGER V4 0 SES V3 1 2 Feature Brief Description Pass Fail Comments pee Brenson ana psn ee Extension and PSTN mew pen and PSTN DTMF in band SIP End Point Extension and PSTN send and receive DTMF tones DTMF RFC 2833 SIP End Point Extension and PSTN send and receive DTMF tones Call Forward Call Forward All CFA Busy CFB and Not Answered CFNA mem aoe O Indicator from server to Endpoint transfer to Endpoint Extension and PSTN waiting timeout release before answer CLIP Type and Il case Gime FE CLIR Remote Extension and PSTN to Endpoint Music on Hold and resume with Endpoint Extension and PSTN 7 3 COMPATIBILITY WITH NORTEL MCS 5100 V4 5 Feature Brief Description Pass Fail Comments am Ee S oo Extension and PSTN and PSTN DTMF in band SIP End Point Extension Pass and PSTN send and receive DTMF tones DTMF RFC 2833 SIP End Point Extension Pass and PSTN send and receive DTMF tones Call Forward Call Forward All CFA Busy Pass Requires SIM card
13. gured to work with SIP enabled IP PBX or other SIP servers Refer to Section 7 above for specific interoperability test results 9 1 ENABLING INTERNET CALLING Before VoIP can be configured Internet Calling needs to be turned on Internet Calling is turned off by default The Internet Calling plug in on the Home screen displays Off indicating that the feature is turned off To turn the feature on select Internet Calling on the Home screen and press Enter This displays the Internet Calling Settings page Change the Use Internet Calling setting from Never to any of the other available options This changes the status of the plug in on the Home screen from Off to Not Available This indicates that even though VoIP is turned on it hasn t been configured yet 9 2 CHECKING WI FI W1 Fi connectivity is required for VoIP to work Before configuring VoIP please set up Wi Fi and make sure the phone can connect to an Access Point AP and that Internet Explorer Mobile can be used to browse web pages 9 3 USING HP IPAQ SETUP ASSISTANT Once Internet Calling is turned on and Wi Fi is set up VoIP can be configured using the HP iPAQ Setup Assistant software included on the Companion CD The VoIP tab under Setup Assistant provides three categories of settings Account Server and VoiceMail Account Please provide the username and password for the VoIP SIP account Also provide the D
14. hat manage their own IP PBX systems and provision devices for enterprise users Mobile enterprise VoIP services should be accessible anywhere the VoIP enabled enterprise wireless local area network WLAN provides coverage typically within company buildings and possibly around corporate campuses Using the built in VoIP client for mobile access to the company s IP PBX from remote locations homes hotels Internet hotspots etc is not a supported capability HP uses the native Microsoft SIP Session Initiation Protocol client in Windows Mobile 6 to access VoIP services and this SIP client s capabilities and limitations have been further split into the following categories for the purposes of this document e Media and Signaling Protocols e Telephony Features Audio Quality Hardware and Firmware WLAN Infrastructure IP PBX and SIP Server Support Accessory Support While Internet voice services delivered by Internet Telephony Service Providers ITSPs may also be based on SIP HP has done no testing or validation of interoperability with these services at this time 2 Media and Signaling Protocols This category is related to the industry protocols that exist today for carrying control messages and media messages Supported e SIP Session Initiation Protocol for Signaling RFC 3261 e G 711 Audio codec for compression decompression of voice Both A law and u law variants are included Not Supported e H 323 signali
15. l list of the RFCs supported and not supported is included in Appendix A of this document This category is related to the components that need to be in the device to handle audio quality Supported e Automatic Gain Control Adaptive Jitter Buffer Management Voice Activity Detection Silence Suppression Comfort Noise Generation Acoustic Echo Cancellation 3 party sourced not provided by Microsoft 5 Hardware and Firmware This category is related to WLAN and other hardware firmware components in the handset that are critical to the performance and usability of VoIP Supported e 802 11b g e 802 111 PEAPvO and EAP TLS with certificates e Encryption suites WEP64 WEP128 TKIP and AES CCMP Not Supported e Full 802 1X Supplicant Microsoft provides the limited set of EAP authentication noted above Additional EAP methods may be required by some customers A 3 party supplicant is currently under consideration for a future release e Fast Roaming A basic roaming agent has been implemented and specific roaming performance results will be published when they become available 802 1 1r Fast Roaming is currently unsupported Fast roaming enhancements are under consideration for a future release e Cisco Compatible Extensions CCX The WLAN module firmware and driver would have to be updated to support the CCX v4 ASD feature set A CCX v4 implementation is currently under consideration for a future r
16. lack of Wi Fi access the IP PBX being unavailable or some other failure 9 4 VOIP DIAL PLAN The dial plan for VoIP consists of dialing rules These dialing rules are constructed using regular expressions Please refer to the Windows Mobile 6 documentation for information on creating dialing rules The dial plan is an XML file based on the OMA provisioning standard The Voice Messenger comes with a default dial plan which may not necessarily work for a specific set up The dial plan will have to be modified and to do this a base XML template will be provided This should be used to add new dialing rules specific to the infrastructure being targeted Once the XML template is modified to add new rules it should be applied to the phone s to provision the phone s with the updated VoIP dial plan This can be done using any of the provisioning methods supported by Windows Mobile 6 1 OTA using OMA Client provisioning WAP push 2 OTA using OMA Device Management provisioning OMA DM server is required 3 Using a CAB Provisioning Format cpf file which can be delivered over HTTP or by using a SD MMC card or copied over to the phone directly using ActiveSync 4 Using Remote API RAPI in ActiveSync to push the provisioning XML file to the phone 5 Application Developers can use the DMProcessConfigXML API to provision Please refer to the Windows Mobile 6 documentation for additional details on the provisioning methods listed above
17. ng protocol used in some IP PB Xs e SCCP Cisco s proprietary call control protocol for Unified CallManager e G 729 Compressed audio codec more suitable for voice over WLAN compared to G 711 e Secure RTP SRTP secure version of the RTP protocol RFC 3711 A full list of the standards supported and not supported is included in Appendix A of this document 3 Telephony Features This category is related to the standard telephony features available in a VoIP implementation Supported e Originate and Terminate Calls e Caller ID e Call Waiting e Call Hold e Call Mute e Call Forwarding only with a SIM card present o The number the call is forwarded to must be in a format E 164 that can be validated on the GSM network e Configuring Caller ID only with a SIM card present e Configuring Call Waiting only with a SIM card present e Call Barring only with a SIM card present e Blind Transfer e In Band and out of band DTMF o Out of band DTMF is requested by default If the other end point declines in band DTMF is used e Emergency Calling over GSM only Interoperability testing with SIP enabled IP PBX systems is being done to confirm feature functionality with each vendor s SIP implementation Refer to subsequent sections of this document for test results Not Supported Conferencing a second line Consultative Transfer Call Park Pick up Do Not Disturb Emergency Calling over IP A ful
18. omain name for the account The username and Domain are used to construct the SIP URI for the account as in the example below User name johndoe Domain voipservice sip com SIP URI sip johndoe voipservice sip com The name of the VoIP Service Provider can also be specified this is optional Server This category is for the SIP settings associated with the SIP enabled IP PBX or the SIP Server The name s or IP Address es of the SIP Proxy and the SIP Registrar must be specified here By default Register with SIP Proxy is checked indicating that the SIP Proxy is also used as the SIP Registrar In this case the name or IP Address of the SIP Proxy only needs to be provided and the same name IP address is also used for the SIP Registrar If the Proxy and the Registrar are different please uncheck Register with SIP Proxy and provide the name IP address for the SIP Registrar in addition to the SIP Proxy Voice Mail If a Voice Mail number is provided with the account please include it here This is optional The Options button can be used to access some advanced VoIP settings The UDP signaling method can be toggled between symmetric and asymmetric Symmetric UDP signaling is used by default An option to enable early UDP packets on RTP and RTCP ports is available This enables opening of ports on a NAT allowing NAT traversal of traffic A prefix digit can be added to outgoing phone numbers For example if 9 is speci
19. r Checkit Wi Eise cman me ees note came ees aeestcncenene cane unemeieehases 12 9 3 Using HP iPAQ Setup Assistant vs idcsccssscesecesnscavavcevevesscvsasansesccnecscesaseanavesdpbaveasest 12 g WOT Dia PIA sena E accesetatesauansusiadessauenas 14 9 4 1 IMPORTANT NOTE ABOUT EMERGENCY CALLING 0 14 9 5 ADDITIONAL CONFIGURATION 0 00 ccccsccceeeseneceeeseneeccesesneceessneeeeees 16 10 Appendix A Standards Support c cc cecccccceccccccececceeseeessesssnssaaeeeeeeeeeeeeeeseseeeeeees 17 10 1 Signaling Standards Implemented cccccsccsscccecceceecesseseeseessnsneceeseeeeeeeees 17 10 2 MediwvOfher Stannard Secs conic svavesvevedvay cvs tondevavsvanveslenscdvevasvapecdevedacenesebrarevedsanwisnes 18 10 3 Standards Not Implemented sca ciescsiccpsuits tadsnihenmotieasstunnicedunesienleaienbinbeastossctuneintedinns 18 Mobile IP telephony services can be separated into three broad categories each with its own specific target markets and technologies used These categories are illustrated below Enterprise VolP Internet Voice Services Services PPBX extension Skype e Microsoft and 3 e Internet Telephony arty clients Service Providers ay ITSPs Mobile Operator IP Telephony Services Fixed Mobile Convergence with UMA GAN e 3GPP IMS Tested use cases for the built in VoIP client on the HP iPAQ 500 series smartphone cover the enterprise VoIP segment exclusively and expect the enterprise has IT staff t
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