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Versions: 00 01 02 03 04 05 06 07 08 RFC 4504

        SIPPING WG                      H. Sinnreich/pulver.com, editor
        Internet Draft                  S. Lass/MCI
                                        C. Stredicke/snom
                                        October 2005

            SIP Telephony Device Requirements and Configuration
                     draft-sinnreich-sipdev-req-08.txt

     Status of this Memo

        This memo provides information for the Internet community.
        It does not specify an Internet standard of any kind.
        Distribution of this memo is unlimited.

        By submitting this Internet-Draft, each author represents
        that any applicable patent or other IPR claims of which he
        or she is aware have been or will be disclosed, and any of
        which he or she becomes aware will be disclosed, in
        accordance with Section 6 of BCP 79.

        Internet-Drafts are working documents of the Internet
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        The list of current Internet-Drafts can be accessed at
             http://www.ietf.org/ietf/1id-abstracts.txt

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        This Internet-Draft will expire on May 2006.

     Abstract

        This document describes the requirements for SIP telephony
        devices, based on the deployment experience of large numbers
        of SIP phones and PC clients using different implementations
        in various networks. The objectives of the requirements are
        a well defined set of interoperability and multi-vendor
        supported core features, so as to enable similar ease of
        purchase, installation and operation as found for PCs, PDAs
        analog feature phones or mobile phones.
        We present a glossary of the most common settings and some
        of the more widely used values for some settings.



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        Conventions used in this document

        This document is informational and therefore the key words
        "MUST", "SHOULD", "SHOULD NOT", "MAY", in this document are
        not to be interpreted as described in RFC 2119 [2], but
        rather indicate the nature of the suggested requirement.

     Table of Contents

        1. Introduction...............................................3
        2. Generic Requirements.......................................4
           2.1. SIP Telephony Devices.................................4
           2.2. DNS and ENUM Support..................................5
           2.3. SIP Device Resident Telephony Features................6
           2.4. Support for SIP Services..............................9
           2.5. Basic Telephony and Presence Information Support.....10
           2.6. Emergency and Resource Priority Support..............11
           2.7. Multi-Line Requirements..............................12
           2.8. User Mobility........................................13
           2.9. Interactive Text Support.............................13
           2.10. Other Related Protocols.............................15
           2.11. SIP Device Security Requirements....................15
           2.12. Quality of Service..................................16
           2.13. Media Requirements..................................16
           2.14. Voice Codecs........................................17
           2.15. Telephony Sound Requirements........................18
           2.16. International Requirements..........................18
           2.17. Support for Related Applications....................19
           2.18. Web Based Feature Management........................19
           2.19. Firewall and NAT Traversal..........................19
           2.20. Device Interfaces...................................20
        3. Glossary and Usage for the Configuration Settings.........21
           3.1. Device ID............................................22
           3.2. Signaling Port.......................................22
           3.3. RTP Port Range.......................................22
           3.4. Quality of Service...................................22
           3.5. Default Call Handling................................23
              3.5.1. Outbound Proxy..................................23
              3.5.2. Default Outbound Proxy..........................23
              3.5.3. SIP Session Timer...............................23
           3.6. Telephone Dialing Functions..........................23
              3.6.1. Phone Number Representations....................23
              3.6.2. Digit Maps and/or the Dial/OK Key...............24
              3.6.3. Default Digit Map...............................25
           3.7. SIP Timer Settings...................................25
           3.8. Audio Codecs.........................................25
           3.9. DTMF Method..........................................25
           3.10. Local and Regional Parameters.......................26
           3.11. Time Server.........................................26


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           3.12. Language............................................26
           3.13. Inbound Authentication..............................27
           3.14. Voice Message Settings..............................27
           3.15. Phonebook and Call History..........................27
           3.16. User Related Settings and Mobility..................28
           3.17. AOR Related Settings................................28
           3.18. Maximum Connections.................................29
           3.19. Automatic Configuration and Upgrade.................29
           3.20. Security Configurations.............................29
        4. Security Considerations...................................30
           4.1. Threats and Problem Statement........................30
           4.2. SIP Telephony Device Security........................31
           4.3. Privacy..............................................32
           4.4. Support for NAT and Firewall Traversal...............32
        5. IANA Considerations.......................................33
        6. Acknowledgments...........................................33
        7. Changes from Previous Versions............................34
        8. References................................................36
        9. Author's Addresses........................................42
        10. Copyright Notice.........................................43

     1. Introduction

        This document has the objective of focusing the Internet
        communications community on requirements for telephony
        devices using SIP.

        We base this information from developing and using a large
        number of SIP telephony devices in carrier and private IP
        networks and on the Internet. This deployment has shown the
        need for generic requirements for SIP telephony devices and
        also the need for some specifics that can be used in SIP
        interoperability testing.

        SIP telephony devices, also referred to as SIP User Agents
        (UAs) can be any type of IP networked computing user device
        enabled for SIP based IP telephony. SIP telephony user
        devices can be SIP phones, adaptors for analog phones and
        for fax machines, conference speakerphones, software
        packages (soft clients) running on PCs, laptops, wireless
        connected PDAs, 'Wi-Fi' SIP mobile phones, as well as other
        mobile and cordless phones that support SIP signaling for
        real time communications. SIP-PSTN gateways are not the
        object of this memo, since they are network elements and not
        end user devices.

        SIP telephony devices can also be instant messaging (IM)
        applications that have a telephony option.



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        SIP devices MAY support various other media besides voice,
        such as text, video, games and other Internet applications;
        however the non-voice requirements are not specified in this
        document, except when providing enhanced telephony features.

        SIP telephony devices are highly complex IP endpoints that
        speak many Internet protocols, have audio and visual
        interfaces and require functionality targeted at several
        constituencies: (1) End users, (2) service providers and
        network administrators and (3) manufacturers, as well as (4)
        system integrators.

        The objectives of the requirements are a well defined set of
        interoperability and multi-vendor supported core features,
        so as to enable similar ease of purchase, installation and
        operation as found for standard PCs, analog feature phones
        or mobile phones. Given the cost of some feature rich
        display phones may approach the cost of PCs and PDAs,
        similar or even better ease of use as compared to personal
        computers and networked PDAs is expected by both end users
        and network administrators.

        While some of the recommendations of this document go beyond
        what is currently mandated for SIP implementations within
        the IETF, this is believed necessary to support the
        specified operational objectives.  However, it is also
        important to keep in mind that the SIP specifications are
        constantly being evolved, thus these recommendations need to
        be considered in the context of that change and evolution.
        Due to the evolution of IETF documents in the standards
        process, and the informational nature of this memo, the
        usual distinction between normative and informative
        references is not made here.

     2. Generic Requirements

        We present here a minimal set of requirements that MUST be
        met by all SIP [3] telephony devices, except where SHOULD or
        MAY is specified.

     2.1. SIP Telephony Devices

       This memo applies mainly to desktop phones and other special
       purpose SIP telephony hardware. Some of the requirements in
       this section are not applicable to PC/laptop or PDA software
       phones (soft phones) and mobile phones.




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       Req-1: SIP telephony devices MUST be able to acquire IP
               network settings by automatic configuration using
               DHCP [4].

       Req-2: SIP telephony devices MUST be able to acquire IP
               network settings by manual entry of settings from the
               device.

       Req-3: SIP telephony devices SHOULD support IPv6. Some newer
               wireless networks may mandate support for IPv6 and in
               such networks SIP telephony devices MUST support
               IPv6.

       Req-4: SIP telephony devices MUST support the Simple Network
               Time Protocol [5].

       Req-5: Desktop SIP phones and other special purpose SIP
               telephony devices MUST be able to upgrade their
               firmware to support additional features and the
               functionality.

       Req-6: Users SHOULD be able to upgrade the devices with no
               special applications or equipment; or a service
               provider SHOULD be able to push the upgrade down to
               the devices remotely.

     2.2. DNS and ENUM Support

       Req-7: SIP telephony devices MUST support RFC 3263 [6] for
               locating a SIP server and selecting a transport
               protocol.

       Req-8: SIP telephony devices MUST incorporate DNS resolvers
               that are configurable with at least two entries for
               DNS servers for redundancy. To provide efficient DNS
               resolution, SIP telephony devices SHOULD query
               responsive DNS servers and skip DNS servers that have
               been non-responsive to recent queries.



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       Req-9: To provide efficient DNS resolution and to limit post-
               dial delay, SIP telephony devices MUST cache DNS
               responses based on the DNS time-to-live.

       Req-10: For DNS efficiency, SIP telephony devices SHOULD use
               the additional information section of the DNS
               response instead of generating additional DNS
               queries.

       Req-11: SIP telephony devices MAY support ENUM [7] in case
               the end users prefer to have control over the ENUM
               lookup. Note: The ENUM resolver can also be placed in
               the outgoing SIP proxy to simplify the operation of
               the SIP telephony device.

     2.3. SIP Device Resident Telephony Features

       Req-12: SIP telephony devices MUST support RFC 3261 [3].

       Req-13: SIP telephony devices SHOULD support the SIP Privacy
               header by populating headers with values that reflect
               the privacy requirements and preferences as described
               in "Section 4 User Agent Behavior" in RFC 3323 [8].

       Req-14: SIP telephony devices MUST be able to place an
               existing call on hold, and initiate or receive
               another call, as specified in RFC 3264 [12] and
               SHOULD NOT omit the sendrecv attribute.

       Req-15: SIP telephony devices MUST provide a call waiting
               indicator. When participating in a call, the user
               MUST be alerted audibly and/or visually of another
               incoming call. The user MUST be able to
               enable/disable the call waiting indicator.

       Req-16: SIP telephony devices MUST support SIP message
               waiting [43] and the integration with message store
               platforms.





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       Req-17: SIP telephony devices MAY support a local dial plan.
       If a dial plan is supported, it MUST be able to match the
       user input to one of multiple pattern strings and transform
       the input to a URI, including an arbitrary scheme and URI
       parameters.

       Example: If a local dial plan is supported, it SHOULD be
       configurable to generate any of the following URIs when
       "5551234" is dialed:

       tel:+12125551234
       sip:+12125551234@example.net;user=phone
       sips:+12125551234@example.net;user=phone
       sip:5551234@example.net
       sips:5551234@example.net
       tel:5551234;phone-context=nyc1.example.net
       sip:5551234;phone-
       context=nyc1.example.net@example.net;user=phone
       sips:5551234;phone-
       context=nyc1.example.net@example.net;user=phone
       sip:5551234;phone-
       context=nyc1.example.net@example.net;user=dialstring
       sips:5551234;phone-
       context=nyc1.example.net@example.net;user=dialstring
       tel:5551234;phone-context=+1212
       sip:5551234;phone-context=+1212@example.net;user=phone
       sips:5551234;phone-context=+1212@example.net;user=phone
       sip:5551234;phone-context=+1212@example.net;user=dialstring
       sips:5551234;phone-context=+1212@example.net;user=dialstring

       If a local dial plan is not supported, the device SHOULD be
       configurable to generate any of the following URIs when
       "5551234" is dialed:

       sip:5551234@example.net
       sips:5551234@example.net
       sip:5551234;phone-
       context=nyc1.example.net@example.net;user=dialstring
       sips:5551234;phone-
       context=nyc1.example.net@example.net;user=dialstring
       sip:5551234;phone-context=+1212@example.net;user=dialstring
       sips:5551234;phone-context=+1212@example.net;user=dialstring"


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       Req-18: SIP telephony devices MUST support URIs for telephone
               numbers as per RFC 3966 [9]. This includes
               the reception as well as the sending of requests. The
               reception may be denied according to the configurable
               security policy of the device. It is a reasonable
               behavior to send a request to a preconfigured
               outbound proxy.

       Req-19: SIP telephony devices MUST support REFER and NOTIFY
               for call transfer [45], [46]. SIP telephony devices
               MUST support escaped Replaces-Header (RFC 3891) and
               SHOULD support other escaped headers in the Refer-To
               header.

       Req-20: SIP telephony devices MUST support the unattended
               call transfer flows as defined in [46].

       Req-21: SIP telephony devices MUST support the attended call
               transfer as defined in [46].

       Req-22: SIP telephony devices MAY support device based 3-way
               calling by mixing the audio streams and displaying
               the interactive text of at least 2 separate calls.

       Req-23: SIP telephony devices MUST be able to send DTMF named
               telephone events as specified by RFC 2833 [11].

       Req-24: Payload type negotiation MUST comply with RFC 3264
               [12] and with the registered MIME types for RTP
               payload formats in RFC 3555 [13].

       Req-25: The dynamic payload type MUST remain constant
               throughout the session. For example, if an endpoint
               decides to renegotiate codecs or put the call on
               hold, the payload type for the re-invite MUST be the
               same as the initial payload type. SIP devices MAY
               support Flow Identification as defined in RFC 3388
               [14].





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       Req-26: When acting as a UAC, SIP telephony devices SHOULD
               support the gateway model of RFC 3960 [71]. When
               acting as a UAS, SIP telephony devices SHOULD NOT
               send early media.

       Req-27: SIP telephony devices MUST be able to handle multiple
               early dialogs in the context of request forking. When
               a confirmed dialog has been established, it is an
               acceptable behavior to send a BYE request in response
               to additional 2xx responses that establish additional
               confirmed dialogs.

       Req-28: SIP devices with a suitable display SHOULD support
               the call-info header and depending on the display
               capabilities MAY for example display an icon or the
               image of the caller.

       Req-29: To provide additional information about call
               failures, SIP telephony devices with a suitable
               display MUST render the "Reason Phrase" of the SIP
               message or map the "Status-Code" to custom or default
               messages. This presumes the language for the reason
               phrase is the same as the negotiated language. The
               devices MAY use an internal "Status Code" table if
               there was a problem with the language negotiation.

       Req-30: SIP telephony devices MAY support music on hold, both
               in receive mode or locally generated. See also "SIP
               Service Examples" for a call flow with music on hold
               [46].

       Req-31: SIP telephony devices MAY ring after a call has been
               on hold for a predetermined period of time, typically
               3 minutes.

     2.4. Support for SIP Services

       Req-32: SIP telephony devices MUST support the SIP Basic Call
               Flow Examples as per RFC 3665 [47].





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       Req-33: SIP telephony devices MUST support the SIP-PSTN
               Service Examples as per RFC 3666 [16].

       Req-34: SIP telephony devices MUST support the Third Party
               Call Control model [17], in the sense that they may
               be the controlled device.

       Req-35: SIP telephony devices SHOULD support SIP call control
               and multiparty usage [42].

       Req-36: SIP telephony devices SHOULD support conferencing
               services for voice [48], [49] and interactive text
               [56] and if equipped with an adequate display MAY
               also support instant messaging (IM) and presence
               [50], [59].

       Req-37: SIP telephony devices SHOULD support the indication
               of the User Agent Capabilities and MUST support the
               caller capabilities and preferences as per RFC 3840
               [52].

       Req-38: SIP telephony devices MAY support service mobility:
               Devices MAY allow roaming users to input their
               identity so as to have access to their services and
               preferences from the home SIP server. Examples of
               user data to be available for roaming users are: User
               service ID, the dialing plan, personal directory and
               caller preferences.

     2.5. Basic Telephony and Presence Information Support

        The large color displays in some newer models make such SIP
        phones and applications attractive for a rich communication
        environment. This document is focused however only on
        telephony specific features enabled by SIP Presence and SIP
        Events.
        SIP telephony devices can also support presence status, such
        as the traditional Do Not Disturb, new event state based
        information, such as being in another call or being in a
        conference, typing a message, emoticons, etc. Some SIP
        telephony User Agents can support for example a voice
        session and several IM sessions with different parties.



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       Req-39: SIP telephony devices SHOULD support Presence
               information [50] and SHOULD support the Rich Presence
               Information Data Format [51] for the new IP
               communication services enabled by Presence.

       Req-40: Users MUST be able to set the state of the SIP
               telephony device to "Do Not Disturb", and this MAY be
               manifested as a Presence state across the network if
               the UA can support Presence information

       Req-41: SIP telephony devices with "Do Not Disturb" enabled
               MUST respond to new sessions with "486 Busy Here".

     2.6. Emergency and Resource Priority Support

       Req-42: Emergency calling: For emergency numbers (e.g. 911,
               SOS URL), SIP telephony devices SHOULD support the
               work of the ECRIT WG [54].

       Req-43: Priority header: SIP devices SHOULD support the
               setting by the user of the Priority header specified
               in RFC 3261 for such applications as emergency calls
               or for selective call acceptance.

       Req-44: Resource Priority header: SIP telephony devices that
               are used in environments that support emergency
               preparedness MUST also support the sending and
               receiving of the Resource-Priority header as
               specified in [55]. The Resource Priority header
               influences the behavior for message routing in SIP
               proxies and PSTN telephony gateways and is different
               from the SIP Priority header specified in RFC 3261.
               Users of SIP telephony devices may want to be
               interrupted in their lower-priority communications
               activities if such an emergency communication request
               arrives.

       Note: As of this writing we recommend implementers to follow
       the work of the Working Group on Emergency Context Resolution
       with Internet Technologies (ecrit) in the IETF. The complete



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       solution is for further study at this time. There is also
       work on the requirements for location conveyance in the
       SIPPING WG, see [77].

     2.7. Multi-Line Requirements

        A SIP telephony device can have multiple lines: One SIP
        telephony device can be registered simultaneously with
        different SIP registrars from different service providers,
        using different names and credentials for each line. The
        different sets of names and credentials are also called 'SIP
        accounts'. The "line" terminology has been borrowed from
        multi-line PSTN/PBX phones, except that for SIP telephony
        devices there can be different SIP registrar/proxies for
        each line, each of which may belong to a different service
        provider, whereas this would be an exceptional case for the
        PSTN and certainly not the case for PBX phones. Multi-line
        SIP telephony devices resemble more closely e-mail clients
        that can support several e-mail accounts.

        Note: Each SIP account can usually support different
        Addresses of Record (AOR) with a different list of contact
        addresses (CA), as may be convenient for example when having
        different SIP accounts for business and for the private
        life. Some of the CAs in different SIP accounts may though
        point to the same devices.

       Req-45: Multi-line SIP telephony devices MUST support a
               unique authentication username, authentication
               password, registrar, and identity to be provisioned
               for each line. The authentication username MAY be
               identical with the user name of the AOR and the
               domain name MAY be identical with the host name of
               the registrar.

       Req-46: Multi-line SIP telephony devices MUST be able to
               support the state of the client to Do Not Disturb on
               a per line basis.

       Req-47: Multi-line SIP telephony devices MUST support multi-
               line call waiting indicators. Devices MUST allow the
               call waiting indicator to be set on a per line basis.





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       Req-48: Multi-line SIP telephony devices MUST be able to
               support a few different ring tones for different
               lines. We specify here "a few", since provisioning
               different tones for all lines may be difficult for
               phones with many lines.

     2.8.  User Mobility

       The following requirements allow users with a set of
       credentials to use any SIP telephony device that can support
       personal credentials from several users, distinct from the
       identity of the device.

       Req-49: User mobility enabled SIP telephony devices MUST
               store static credentials associated with the device
               in non-volatile memory. This static profile is used
               during the power up sequence.

       Req-50: User mobility enabled SIP telephony devices SHOULD
               allow a user to walk up to a device and input their
               personal credentials. All user features and settings
               stored in home SIP proxy and the associated policy
               server SHOULD be available to the user.

       Req-51: User mobility enabled SIP telephony devices
               registered as fixed desktop with network
               administrator MUST use the local static location data
               associated with the device for emergency calls.

      2.9. Interactive Text Support

        SIP telephony devices supporting Instant Messaging based on
        SIMPLE [50] support text conversation based on blocks of
        text. However, continuous interactive text conversation may
        be sometimes preferred as a parallel to voice, due to its
        interactive and more streaming-like nature, thus more
        appropriate for real time conversation. It also allows for
        text captioning of voice in noisy environments and for those
        who cannot hear well or cannot hear at all.

        Finally continuous, character by character text is what is
        preferred by emergency and public safety programs (e.g. 112
        and 911) because of its immediacy, efficiency, lack of
        crossed messages problem, better ability to interact with a


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        confused person, and the additional information that can be
        observed from watching the message as it is composed.

        Req-52: SIP telephony devices such as SIP display phones and
               IP-analog adapters SHOULD support the accessibility
               requirements for the deaf, hard of hearing and speech
               impaired individuals as per RCF 3351 [18] and also
               for interactive text conversation [56], [70].

        Req-53: SIP telephony devices SHOULD provide a way to input
               text and to display text through any reasonable
               method. Built-in user interfaces, standard wired or
               wireless interfaces, and/or support for text through
               a web interface are all considered reasonable
               mechanisms.

        Req-54: SIP telephony devices SHOULD provide an external
               standard wired or wireless link to connect external
               input (keyboard, mouse) and display devices.

        Req-55: SIP telephony devices which include a display, or
               have a facility for connecting an external display,
               MUST include protocol support as described in RFC
               2793 for real-time interactive text.

        Req-56: There may be value of having RFC 2793 support in a
               terminal also without a visual display. A synthetic
               voice output for the text conversation may be of
               value for all who can hear, and thereby having the
               opportunity to have a text conversation with other
               users.

        Req-57: SIP telephony devices MAY provide analog adaptor
               functionality through an RJ-11 FXS port to support
               FXS devices. If an RJ-11 (FXS) port is provided, then
               it MAY support a gateway function from all text-
               telephone protocols according to ITU-T Recommendation
               V.18 to RFC 2793 text conversation (in fact this is
               encouraged in the near term during the transition to
               widespread use of SIP telephony devices). If this
               gateway function is not included or fails, the device



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               MUST pass-through all text-telephone protocols
               according to ITU-T Recommendation V.18, November
               2000, in a transparent fashion.

        Req-58: SIP telephony devices MAY provide a 2.5 mm audio
               port, in portable SIP devices, such as PDAs and
               various wireless SIP phones.

     2.10. Other Related Protocols

       Req-59: SIP telephony devices MUST support the Real-Time
               Protocol and the Real-Time Control Protocol, RFC 3550
               [20]. SIP devices SHOULD use RTCP Extended Reports
               for logging and reporting on network support for
               voice quality, RFC 2611 [21] and MAY also support the
               RTCP summary report delivery [57].

     2.11. SIP Device Security Requirements

       Req-60: SIP telephony devices MUST support digest
               authentication as per RFC3261. In addition, SIP
               telephony devices MUST support TLS for secure
               transport [36] for scenarios where the SIP registrar
               is located outside the secure, private IP network in
               which the SIP UA may reside. Note: TLS need not be
               used in every call though.

       Req-61: SIP telephony devices MUST be able to password
               protect configuration information and administrative
               functions.

       Req-62: SIP telephony devices MUST NOT display the password
               to the user or administrator after it has been
               entered.

       Req-63: SIP clients MUST be able to disable remote access,
               i.e. block incoming SNMP (where this is supported),
               HTTP, and other services not necessary for basic
               operation.





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       Req-64: SIP telephony devices MUST support the option to
               reject an incoming INVITE where the user-portion of
               the SIP request URI is blank or does not match a
               provisioned contact. This provides protection against
               war-dialer attacks, unwanted telemarketing and spam.
               The setting to reject MUST be configurable.

       Req-65: When TLS is not used, SIP telephony devices MUST be
               able to reject an incoming INVITE when the message
               does not come from the proxy or proxies where the
               client is registered. This prevents callers from
               bypassing terminating call features on the proxy. For
               DNS SRV specified proxy addresses, the client must
               accept an INVITE from all of the resolved proxy IP
               addresses.

     2.12. Quality of Service

       Req-66: SIP devices MUST support the IPv4 DSCP field for RTP
               streams as per RFC 2597 [22]. The DSCP setting MUST
               be configurable to conform with the local network
               policy.

       Req-67: If not specifically provisioned, SIP telephony
               devices SHOULD mark RTP packets with the recommended
               DSCP for expedited forwarding (codepoint 101110); and
               mark SIP packets with DSCP AF31 (codepoint 011010).

       Req-68: SIP telephony devices MAY support RSVP [23].

     2.13. Media Requirements

       Req-69: To simplify the interoperability issues, SIP
               telephony devices MUST use the first matching codec
               listed by the receiver if the requested codec is
               available in the called device. See the offer/answer
               model in RFC 3261.

       Req-70: To reduce overall bandwidth, SIP telephony devices
               MAY support active voice detection and comfort noise
               generation.



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     2.14. Voice Codecs

        Internet telephony devices face the problem of supporting
        multiple codecs due to various historic reasons, on how
        telecom industry players have approached codec
        implementations and the serious intellectual property and
        licensing problems associated with most codec types.
        RFC 3551 for example [24] lists 17 registered MIME subtypes
        for audio codecs.

        Ideally, the more codecs can be supported in a SIP telephony
        device, the better, since it enhances the chances of success
        during the codec negotiation at call setup and avoids media
        intermediaries used for codec mediation.

        Implementers interested in a short list MAY however support
        a minimal number of codecs used in wireline VoIP, and also
        codecs found in mobile networks for which the SIP UA is
        targeted. An ordered short list of preferences may look as
        this:

       Req-71: SIP telephony devices SHOULD support AVT payload type
               0 (G.711 uLaw) as in reference [25] and its Annexes 1
               and 2.

       Req-72: SIP telephony devices SHOULD support the Internet Low
               Bit Rate codec (iLBC) [26], [27].

       Req-73: Mobile SIP telephony devices MAY support codecs found
               in various wireless mobile networks. This can avoid
               codec conversion in network based intermediaries.

       Req-74: SIP telephony devices MAY support a small set of
               special purpose codecs, such as G.723.1, where low
               bandwidth usage is needed (for dial-up Internet
               access), SPEEX [28] or G.722 for high quality audio
               conferences.

       Req-75: SIP telephony devices MAY support G.729 and its
               annexes. Note: The G.729 codec is included here for
               backward compatibility only, since the iLBC and the
               G.723.1 codecs are preferable in bandwidth
               constrained environments.



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               Note: The authors believe the Internet Low Bit Rate
               codec (iLBC) should be the default codec for Internet
               telephony.

              A summary count reveals up to 25 and more voice codec
               types currently in use. The authors believe there is
               also a need for a single multi-rate Internet codec,
               such as Speex [28] or similar that can effectively be
               substituted for all of the multiple legacy G.7xx
               codec types, such as G. 711, G.729, G.723.1, G.722,
               etc. for various data rates, thus avoiding the
               complexity and cost to implementers and service
               providers alike who are burdened by supporting so
               many codec types, besides the licensing costs.

     2.15. Telephony Sound Requirements

       Req-76: SIP telephony devices SHOULD comply with the handset
               receive comfort noise requirements outlined in the
               ANSI standards [29], [30].

       Req-77: SIP telephony devices SHOULD comply with the
               stability or minimum loss defined in ITU-T G.177
               [31].

       Req-78: SIP telephony devices MAY support a full-duplex
               speakerphone function with echo and side tone
               cancellation. The design of high quality side tone
               cancellation for desktop IP phones, laptop computers
               and PDAs is outside the scope of this memo.

       Req-79: SIP telephony device MAY support different ring-tones
               based on the caller identity.

     2.16. International Requirements

       Req-80: SIP telephony devices SHOULD indicate the preferred
               language [34] using User Agent Capabilities [52].

       Req-81: SIP telephony devices intended to be used in various
               language settings [34], MUST support other languages
               for menus, help, and labels.




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     2.17. Support for Related Applications

        The following requirements apply to functions placed in the
        SIP telephony device.

       Req-82: SIP telephony devices that have a large display and
                support presence SHOULD display a buddy list [50].

       Req-83: SIP telephony devices MAY support LDAP for client-
                based directory lookup.

       Req-84: SIP telephony devices MAY support a phone setup where
                a URL is automatically dialed when the phone goes
                off-hook.

     2.18. Web Based Feature Management

       Req-85: SIP telephony devices SHOULD support an internal web
                server to allow users the option to manually
                configure the phone and to set up personal phone
                applications such as the address book, speed-dial,
                ring tones, and last but not least the call handling
                options for the various lines, aliases, in a user
                friendly fashion. Web pages to manage the SIP
                telephony device SHOULD be supported by the
                individual device, or MAY be supported in managed
                networks from centralized web servers linked from a
                URI.

               Managing SIP telephony devices SHOULD NOT require
                special client software on the PC or require a
                dedicated management console. SIP telephony devices
                SHOULD support https transport for this purpose.

                In addition to the Web Based Feature Management
                Requirement the device MAY have an SNMP interface
                for monitoring and management purposes.

     2.19. Firewall and NAT Traversal

        The following requirements allow SIP clients to properly
        function behind various firewall architectures.


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       Req-86: SIP telephony devices SHOULD be able to operate
               behind a static NAPT (Network Address
               Translation/Port Address Translation) device. This
               implies the SIP telephony device SHOULD be able to 1)
               populate SIP messages with the public, external
               address of the NAPT device, 2) use symmetric UDP or
               TCP for signaling, and 3) Use symmetric RTP [72].

       Req-87: SIP telephony devices SHOULD support the STUN
               protocol [32] for determining the NAPT public
               external address. A classification of scenarios and
               NATs where STUN is effective is reported in [58].
               Detailed call flows for interactive connectivity
               establishment (ICE) [76] are given in [63].

               Note: Developers are strongly advised to follow the
               document on best current practices for NAT traversal
               for SIP [63].

       Req-88: SIP telephony devices MAY support UPnP
               (http://www.upnp.org/) for local NAPT traversal. Note
               that UPnP does not help if there are NAPT in the
               network of the services provider.

       Req-89: SIP telephony devices MUST be able to limit the ports
               used for RTP to a provisioned range.

     2.20. Device Interfaces

       Req-90: SIP telephony devices MUST support two types of
                addressing capabilities, to enable end users to
                "dial" either phone numbers or URIs.

       Req-91: SIP telephony devices MUST have a telephony-like
                dial-pad and MAY have telephony style buttons like
                mute, redial, transfer, conference, hold, etc. The
                traditional telephony dial-pad interface MAY appear
                as an option in large screen telephony devices using
                other interface models, such as Push-To-Talk in
                mobile phones and the Presence and IM GUI found in
                PCs, PDAs, in mobile phones and in cordless phones.


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       Req-92: SIP telephony devices MUST have a convenient way for
                entering SIP URIs and phone numbers. This includes
                all alphanumeric characters allowed in legal SIP
                URIs. Possible approaches include using a web page,
                display and keyboard entry, type-ahead or graffiti
                for PDAs.

       Req-93: SIP telephony devices should allow phone number entry
                in human friendly fashion, with the usual separators
                and brackets between digits and digit groups.

     3. Glossary and Usage for the Configuration Settings

        SIP telephony devices are quite complex and their
       configuration is made more difficult by the widely diverse
       use of technical terms for the settings. We present here a
       glossary of the most common settings and some of the more
       widely used values for some settings.
       Settings are the information on a SIP UA that it needs so as
       to be a functional SIP endpoint. The settings defined in this
       document are not intended to be a complete listing of all
       possible settings. It MUST be possible to add vendor specific
       settings.
        The list of available settings includes settings that MUST,
       SHOULD or MAY be used by all devices (when present) and that
       make up the common denominator that is used and understood by
       all devices. However, the list is open to vendor specific
       extensions that support additional settings, which enable a
       rich and valuable set of features.
        Settings MAY be read-only on the device. This avoids the
       misconfiguration of important settings by inexperienced users
       generating service cost for operators. The settings
       provisioning process SHOULD indicate which settings can be
       changed by the end-user and which settings should be
       protected.
        In order to achieve wide adoption of any settings format it
       is important that it should not be excessive in size for
       modest devices to use it. Any format SHOULD be structured
       enough to allow flexible extensions to it by vendors.
       Settings may belong to the device or to a SIP service
       provider and the address of record (AOR) registered there.
       When the device acts in the context of an AOR, it will first
       try to look up a setting in the AOR context. If the setting
       can not be found in that context, the device will try to find
       the setting in the device context. If that also fails, the
       device MAY use a default value for the setting.



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        The examples shown here are just of informational nature.
       Other documents may specify the syntax and semantics for the
       respective settings.

     3.1. Device ID

               A device setting MAY include some unique identifier
               for the device it represents. This MAY be an
               arbitrary device name chosen by the user, the MAC
               address, some manufacturer serial number or some
               other unique piece of data. The Device ID SHOULD also
               indicate the ID type.
               Example: DeviceId="000413100A10;type=MAC"

     3.2. Signaling Port

               The port that will be used for a specific transport
               protocol for SIP MAY be indicated with the SIP ports
               setting. If this setting is omitted, the device MAY
               choose any port within a range as specified in 3.3.
               For UDP, the port may also be used for sending
               requests so that NAT devices will be able to route
               the responses back to the UA.
               Example: SIPPort="5060;transport=UDP"

     3.3. RTP Port Range

               A range of port numbers MUST be used by a device for
               the consecutive pairs of ports which MUST be used to
               receive audio and control information (RTP and RTCP)
               for each concurrent connection. Sometimes this is
               required to support firewall traversal and it helps
               network operators to identify voice packets.
               Example: RTPPorts="50000-51000"

     3.4. Quality of Service

               The QoS settings for outbound packets SHOULD be
               configurable for network packets associated with call
               signaling (SIP) and media transport (RTP/RTCP). These
               settings help network operators identifying voice
               packets in their network and allow them to transport
               them with the required QoS. The settings are
               independently configurable for the different
               transport layers and signaling, media or
               administration. The QoS settings SHOULD also include
               the QoS mechanism.
               For both categories of network traffic, the device
               SHOULD permit configuration of the type of service


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               settings for both layer 3 (IP DiffServ) and layer 2
               (for example IEEE 802.1D/Q) of the network protocol
               stack.
               Example: RTPQoS="0xA0;type=DiffSrv,
               5;type=802.1DQ;vlan=324"

     3.5. Default Call Handling

               All of the call handling settings defined below can
               be defined here as default behaviors.

     3.5.1. Outbound Proxy

               The outbound proxy for a device MAY be set. The
               setting MAY require that all signaling packets MUST
               be sent to the outbound proxy or that only in the
               case when no route has been received the outbound
               proxy MUST be used. This ensures that application
               layer gateways are in the signaling path. The second
               requirement allows the optimization of the routing by
               the outbound proxy.
               Example: OutboundProxy="sip:nat.proxy.com"

     3.5.2. Default Outbound Proxy

               The default outbound proxy SHOULD be a global setting
               (not related to a specific line).
               Example: DefaultProxy="sip:123@proxy.com"

     3.5.3. SIP Session Timer

               The re-invite timer allows user agents to detect
               broken sessions caused by network failures. A value
               indicating the number of seconds for the next re-
               invite SHOULD be used if provided.
               Example: SessionTimer="600;unit=seconds"

     3.6. Telephone Dialing Functions

               As most telephone users are used to dialing digits to
               indicate the address of the destination, there is a
               need for specifying the rule by which digits are
               transformed into a URI (usually SIP URI or TEL URI).

     3.6.1. Phone Number Representations

               SIP phones need to understand entries in the phone
               book of the most common separators used between



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               dialed digits, such as spaces, angle and round
               brackets, dashes and dots.
               Example: A phonebook entry of "+49(30)398.33-401"
               should be translated into "+493039833401".

     3.6.2. Digit Maps and/or the Dial/OK Key

               A SIP UA needs to translate user input before it can
               generate a valid request. Digit maps are settings
               that describe the parameters of this process.
               If present, digit maps define patterns that when
               matched define:

               1) A rule by which the end point can judge that the
               user has completed dialing, and
               2) A rule to construct a URI from the dialed digits,
               and optionally
               3) An outbound proxy to be used in routing the SIP
               INVITE.

               A critical timer MAY be provided which determines how
               long the device SHOULD wait before dialing if a dial
               plan contains a T (Timer) character. It MAY also
               provide a timer for the maximum elapsed time which
               SHOULD pass before dialing if the digits entered by
               the user match no dial plan. If the UA has a Dial or
               Ok key, pressing this key will override the timer
               setting.
               SIP telephony devices SHOULD have a Dial/OK key.
               After sending a request, UA SHOULD be prepared to
               receive a 484 Address Incomplete response. In this
               case, the user agent should accept more user input
               and try again to dial the number.
               An example digit map could use regular expressions
               like in DNS NAPTR (RFC2915) to translate user input
               into a SIP URL. Additional replacement patterns like
               "d" could insert the domain name of the used AOR.
               Additional parameters could be inserted in the flags
               portion of the substitution expression. A list of
               those patterns would make up the dial plan:

               |^([0-9]*)#$|sip:\1@\d;user=phone|outbound=proxy.com
               |^([a-zA-Z0-9&=+\$,;?\-_.!~*'()%]+@.+)|sip:\1|
               |^([a-zA-Z0-9&=+\$,;?\-_.!~*'()%]+)$|sip:\1@\d|
               |^(.*)$|sip:\1@\d|timeout=5






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     3.6.3. Default Digit Map

               The SIP telephony device SHOULD support the
               configuration of a default digit map. If the SIP
               telephony device does not support digit maps, it
               SHOULD at least support a default digit map rule to
               construct a URI from digits. If the end point does
               support digit maps, this rule applies if none of the
               digit maps match.
               For example, when a user enters "12345", the UA might
               send the request to "sip:12345@proxy.com;user=phone"
               after the user presses the OK key.

     3.7. SIP Timer Settings

               The parameters for SIP (like timer T1) and other
               related settings MAY be indicated. An example of
               usage would be the reduction of the DNS SRV failover
               time.
               Example: SIPTimer="t1=100;unit=ms"
               Note: The timer settings can be included in the digit
               map.

     3.8. Audio Codecs

               In some cases operators want to control which codecs
               may be used in their network. The desired subset of
               codecs supported by the device SHOULD be configurable
               along with the order of preference. Service providers
               SHOULD have the possibility of plugging in their own
               codecs of choice. The codec settings MAY include the
               packet length and other parameters like silence
               suppression or comfort noise generation.
               The set of available codecs will be used in the codec
               negotiation according to RFC 3264 [12].

               Example: Codecs="speex/8000;ptime=20;cng=on,
               gsm;ptime=30"

               The settings MUST include hints about privacy for
               audio using SRTP that either mandate or encourage the
               usage of secure RTP.
               Example: SRTP="mandatory"

     3.9. DTMF Method

               Keyboard interaction can be indicated with in-band
               tones or preferable with out-of-band RTP packets (RFC
               2833) [11]. The method for sending these events


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               SHOULD be configurable with the order of precedence.
               Settings MAY include additional parameters like the
               content-type that should be used.
               Example: DTMFMethod="INFO;type=application/dtmf,
               RFC2833", [11].

     3.10. Local and Regional Parameters

               Certain settings are dependent upon the regional
               location for the daylight saving time rules and for
               the time zone.
               Time Zone and UTC Offset: A time zone MAY be
               specified for the user. Where one is specified; it
               SHOULD use the schema used by the Olson Time One
               database [33].
               Examples of the database naming scheme are Asia/Dubai
               or America/Los Angeles where the first part of the
               name is the continent or ocean and the second part is
               normally the largest city on that time-zone. Optional
               parameters like the UTC offset may provide additional
               information for UA that are not able to map the time
               zone information to a internal database.
               Example: TimeZone="Asia/Dubai;offset=7200"

     3.11. Time Server

               A time server SHOULD be used. DHCP is the preferred
               way to provide this setting. Optional parameters may
               indicate the protocol that SHOULD be used for
               determining the time. If present, the DHCP time
               server setting has higher precedence than the time
               server Setting.
               Example: TimeServer="12.34.5.2;protocol=NTP"

     3.12. Language

               Setting the correct language is important for simple
               installation around the globe.
               A language Setting SHOULD be specified for the whole
               device. Where it is specified it MUST use the codes
               defined in RFC 3066 [34] to provide some
               predictability.
               Example: Language="de"
               It is recommended to set the Language as writable, so
               that the user MAY change this. This setting SHOULD
               NOT be AOR related.
               A SIP UA MUST be able to parse and accept requests
               containing international characters encoded as UTF-8



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               even if it cannot display those characters in the
               user interface.

     3.13. Inbound Authentication

               SIP allows a device to limit incoming signaling to
               those made by a predefined set of authorized users
               from a list and/or with valid passwords. Note that
               the inbound proxy from most service providers may
               also support the screening of incoming calls, but in
               some cases users may want to have control in the SIP
               telephony device for the screening.
               A device SHOULD support the setting as to whether
               authentication (on the device) is required and what
               type of authentication is required.
               Example: InboundAuthentication="digest;pattern=*"
               If inbound authentication is enabled then a list of
               allowed users and credentials to call this device MAY
               be used by the device. The credentials MAY contain
               the same data as the credentials for an AOR (i.e.
               URL, user, password digest and domain). This applies
               to SIP control signaling as well as call initiation.

     3.14. Voice Message Settings

               Various voice message settings require the use of
               URI's for the service context as specified in RFC
               3087 [35].
               The message waiting indicator (MWI) address setting
               controls where the client SHOULD SUBSCRIBE to a voice
               message server and what MWI summaries MAY be
               displayed [43].
               Example: MWISubscribe="sip:mailbox01@media.proxy.com"
               User Agents SHOULD accept MWI information carried by
               SIP MESSAGE without prior subscription. This way the
               setup of voice message settings can be avoided.

     3.15. Phonebook and Call History

               UA SHOULD have a phonebook and keep a history of
               recent calls. The phonebook SHOULD save the
               information in permanent memory that keeps the
               information even after restarting the device or save
               the information in an external database that
               permanently stores the information.






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     3.16. User Related Settings and Mobility

               A device MAY specify the user which is currently
               registered on the device. This SHOULD be an address-
               of-record URL specified in an AOR definition.
               The purpose of specifying which user is currently
               assigned to this device is to provide the device with
               the identity of the user whose settings are defined
               in the user section. This is primarily interesting
               with regards to user roaming. Devices MAY allow users
               to sign-on to them and then request that their
               particular settings be retrieved. Likewise a user MAY
               stop using a device and want to disable their AOR
               while not present. For the device to understand what
               to do it MUST have some way of identifying users and
               knowing which user is currently using it. By
               separating the user and device properties it becomes
               clear what the user wishes to enable or to disable.
               Providing an identifier in the configuration for the
               user gives an explicit handle for the user. For this
               to work the device MUST have some way of identifying
               users and knowing which user is currently assigned to
               it.
               One possible scenario for roaming is an agent who has
               definitions for several AOR (e.g. one or more
               personal AOR and one for each executive for whom the
               administrator takes calls) that they are registered
               for. If the agent goes to the copy room they would
               sign-on to a device in that room and their user
               settings including their AOR would roam with them.
               The alternative to this is to require the agent to
               individually configure all of the AORs individually
               (this would be particularly irksome using standard
               telephone button entry).
               The management of user profiles, aggregation of user
               or device AOR and profile information from multiple
               management sources are configuration server concerns
               which are out of the scope of this document. However
               the ability to uniquely identify the device and user
               within the configuration data enables easier server
               based as well as local (i.e. on the device)
               configuration management of the configuration data.

     3.17. AOR Related Settings

               SIP telephony devices MUST use the Address of Record
               (AOR) related settings, as specified here.
               There are many properties which MAY be associated
               with or SHOULD be applied to the AOR or signaling


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               addressed to or from the AOR. AORs MAY be defined for
               a device or a user of the device. At least one AOR
               MUST be defined in the settings, this MAY pertain to
               either the device itself or the user.
               Example: AOR="sip:12345@proxy.com"
               It MUST be possible to specify at least one set of
               domain, user name and authentication credentials for
               each AOR. The user name and authentication
               credentials are used for authentication challenges.

     3.18. Maximum Connections

               A setting defining the maximum number of simultaneous
               connections that a device can support MUST be used by
               the device. The end point might have some maximum
               limit, most likely determined by the media handling
               capability. The number of simultaneous connections
               may be also limited by the access bandwidth, such as
               of DSL, cable and wireless users. Other optional
               settings MAY include the enabling or disabling of
               call waiting indication.
               A SIP telephony device MAY support at least two
               connections for three-way conference calls that are
               locally hosted.
               Example: MaximumConnections="2;cwi=false;bw=128". See
               the recent work on connection reuse [74] and the
               guidelines for connection oriented transport for SIP
               [75].

     3.19. Automatic Configuration and Upgrade

               Automatic SIP telephony device configuration SHOULD
               use the processes and requirements described in [60].
               The user name or the realm in the domain name SHOULD
               be used by the configuration server to automatically
               configure the device for individual or group specific
               settings, without any configuration by the user.
               Image and service data upgrades SHOULD also not
               require any settings by the user.

     3.20. Security Configurations

               The device configuration usually contains sensitive
               information that MUST be protected. Examples include
               authentication information, private address books and
               call history entries. Because of this, it is
               RECOMMENDED to use an encrypted transport mechanism
               for configuration data. Where devices use HTTP this
               could be TLS [36].


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               For devices which use FTP or TFTP for content
               delivery this can be achieved using symmetric key
               encryption.
               Access to retrieving configuration information is
               also an important issue. A configuration server
               SHOULD challenge a subscriber before sending
               configuration information.
               The configuration server SHOULD NOT include passwords
               through the automatic configuration process. Users
               SHOULD enter the passwords locally.

     4. Security Considerations

     4.1. Threats and Problem Statement

        While section 2.11  states the minimal security requirements
        and NAT/firewall traversal that have to be met respectively
        by SIP telephony devices, developers and network managers
        have to be aware of the larger context of security for IP
        telephony, especially for those scenarios where security may
        reside in other parts of SIP enabled networks.
        Users of SIP telephony devices are exposed to many threats
        [61] that include but are not limited to fake identity of
        callers, telemarketing, spam in IM, hijacking of calls,
        eavesdropping, learning of private information such as the
        personal phone directory, user accounts and passwords and
        the personal calling history. Various DOS attacks are
        possible, such as hanging up on other people's conversations
        or contributing to DOS attacks of others.
        Service providers are also exposed to many types of attacks
        that include but are not limited to theft of service by
        users with fake identities, DOS attacks and the liabilities
        due to theft of private customer data and eavesdropping in
        which poorly secured SIP telephony devices or especially
        intermediaries such as stateful back-to-back user agents
        with media (B2BUA) may be implicated.
        SIP security is a hard problem for several reasons:

          . Peers can communicate across domains without any pre-
             arranged trust relationship,
          . There may be many intermediaries in the signaling path,
          . Multiple endpoints can be involved in such telephony
             operations as forwarding, forking, transfer or
             conferencing,
          . There are seemingly conflicting service requirements
             when supporting anonymity, legal intercept, call trace
             and privacy,
          . Complications arise from the need to traverse NATs and
             firewalls.


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        There are a large number of deployment scenarios in
        enterprise networks, using residential networks and
        employees using VPN access to the corporate network when
        working from home or on travel. There are different security
        scenarios for each. The security expectations are also very
        different, say within an enterprise network or when using a
        laptop in a public wireless hotspot and it is beyond the
        scope of this memo to describe all possible scenarios in
        detail.
        The authors believe that adequate security for SIP telephony
        devices can be best implemented within protected networks,
        be they private IP networks or service provider SIP enabled
        networks where a large part of the security threats listed
        here are dealt with in the protected network. A more general
        security discussion that includes network based security
        features, such as network based assertion of identity [37]
        and privacy services [38] are outside the scope of this
        memo, but must be well understood by developers, network
        managers and service providers.
        In the following some basic security considerations as
        specified in RFC 3261 are discussed as they apply for SIP
        telephony devices.

     4.2. SIP Telephony Device Security

        Transport Level Security
               SIP telephony devices that operate outside the
               perimeter of secure private IP networks (this
               includes telecommuters and roaming users) MUST use
               TLS [36] to the outgoing SIP proxy for protection on
               the first hop. SIP telephony devices that use TLS
               must support SIPS in the SIP headers.
               Supporting large numbers of TLS channels to endpoints
               is quite a burden for service providers and may
               therefore constitute a premium service feature.

        Digest Authentication
               SIP telephony devices MUST support digest
               authentication to register with the outgoing SIP
               registrar. This assures proper identity credentials
               that can be conveyed by the network to the called
               party. It is assumed that the service provider
               operating the outgoing SIP registrar has an adequate
               trust relationship with their users and knows its
               customers well enough (identity, address, billing
               relationship, etc.). The exceptions are users of
               prepaid service. SIP telephony devices that accept



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               prepaid calls MUST place "unknown" in the "From"
               header.

        End User Certificates
               SIP telephony devices MAY store personal end user
               certificates that are part of some PKI [39] service
               for high security identification to the outgoing SIP
               registrar as well as for end to end authentication.
               SIP telephony devices equipped for certificate based
               authentication MUST also store a key ring of
               certificates from public certificate authorities
               (CA"s).
               Note the recent work in the IETF on certificate
               services that do not require the telephony devices to
               store certificates [69].

        End-to-End Security Using S/MIME
               S/MIME [40] MUST be supported by SIP telephony
               devices to sign and encrypt portions of the SIP
               message that are not strictly required for routing by
               intermediaries. S/MIME protects private information
               in the SIP bodies and in some SIP headers from
               intermediaries. The end user certificates required
               for S/MIME assure the identity of the parties to each
               other. Note: S/MIME need not be used though in every
               call.

     4.3. Privacy

        Media Encryption
               Secure RTP (SRTP) [41] MAY be used for the encryption
               of media such as audio, text and video, after the
               keying information has been passed by SIP signaling.
               Instant messaging MAY be protected end-to-end using
               S/MIME.

     4.4. Support for NAT and Firewall Traversal

               The various NAT and firewall traversal scenarios
               require support in telephony SIP devices. The best
               current practices for NAT traversal for SIP are
               reviewed in [63]. Most scenarios where there are no
               SIP enabled network edge NAT/firewalls or gateways in
               the enterprise can be managed if there is a STUN [32]
               client in the SIP telephony device and a STUN server
               on the Internet, maintained by a service provider. In
               some exceptional cases (legacy symmetric NAT) an
               external media relay must also be provided that can
               support the TURN protocol exchange [62] with SIP


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               telephony devices. Media relays such as TURN come at
               a high bandwidth cost to the service provider, since
               the bandwidth for many active SIP telephony devices
               must be supported. Media relays may also introduce
               longer paths with additional delays for voice.
               Due to these disadvantages of media relays, it is
               preferable to avoid symmetric and non-deterministic
               NAT"s in the network, so that only STUN can be used,
               where required. Reference [73] deals in more detail
               how NAT has to 'behave'.
               It is not always obvious to determine the specific
               NAT and firewall scenario under which a SIP telephony
               device may operate.
               For this reason, the support for Interactive
               Connectivity Establishment (ICE) [76] has been
               defined to be deployed in all devices that required
               end-to-end connectivity for SIP signaling and RTP
               media streams, as well as for streaming media using
               RTSP. ICE makes use of existing protocols, such as
               STUN and TURN.

        Call flows using SIP security mechanisms
               The high level security aspects described here are
               best illustrated by inspecting the detailed call
               flows using SIP security, such as in [64].

       Security enhancements, certificates and identity management
               As of this writing, recent work in the IETF deals
               with the SIP authenticated body (AIB) format [66],
               new S/MIME requirements [67] enhancements for the
               authenticated identity [68], and Certificate
               Management Services for SIP [69]. We recommend
               developers and network managers to follow this work
               as it will develop into IETF standards.

     5. IANA Considerations

        This document has no actions for IANA.

     6. Acknowledgments

        Paul Kyzivat and Francois Audet have made useful comments
        how to support to the dial plan requirements in Req-17.
        Mary Barnes has kindly made a very detailed review on
        version 04 that has contributed to significantly improving
        the document. Useful comments on version 05 have also been
        made by Ted Hardie, David Kessens, Russ Housley and Harald
        Alvestrand that are reflected in this version of the
        document.


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        We would like to thank Jon Peterson for very detailed
        comments on the previous version 0.3 that has prompted the
        rewriting of much of this document. John Elwell has
        contributed with many detailed comments to version of the 04
        of the draft. Rohan Mahy has contributed several
        clarifications to the document and leadership in the
        discussions on support for the hearing disabled. These
        discussions have been concluded during the BOF on SIP
        Devices held during the 57th IETF and the conclusions are
        reflected in the section on interactive text support for
        hearing or speech disabled users.
        Gunnar Hellstrom, Arnoud van Wijk and Guido Gybels have been
        instrumental in driving the specification for support of the
        hearing disabled.
        The authors would also like to thank numerous persons for
        contributions and comments to this work: Henning
        Schulzrinne, Jorgen Bjorkner, Jay Batson, Eric Tremblay,
        David Oran and Denise Caballero McCann, Brian Rosen, Jean
        Brierre, Kai Miao, Adrian Lewis and Franz Edler. Jonathan
        Knight has contributed significantly to earlier versions of
        the requirements for SIP phones. Peter Baker has also
        provided valuable pointers to TIA/EIA IS 811 requirements to
        IP phones that are referenced here.
        Last but not least, the co-authors of the previous versions,
        Daniel Petrie and Ian Butcher have provided support and
        guidance all along in the development of these requirements.
        Their contributions are now the focus of separate documents.

     7. Changes from Previous Versions

        Changes from draft-sinnreich-sipdev-req-07

           . Updated the references from the IETF and explained in
             the intro that due to the informational nature of this
             memo, no distinction is made here between normative and
             informative references.

           . Specified that ideally, the more codecs can be
             supported the better and given the reasons. Also
             specified for implementers interested in a short list,
             what an ordered short list of codecs may look like.

           . Deleted any specifics on codecs for various mobile
             networks.

           . Specified that G.729 codecs may be supported for
             backward compatibility but that iLBC and G.723.1 codecs
             will perform better in low bandwidth environments.



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           . Clarified Req-85 that instead of an internal web server
             in the device, a URI may link to the web page for
             manual configuration.

           . Changed in Section 3.2 the "MUST be indicated" for
             ports for the transport protocol to MAY.

           . Changed Req-90 for clarity on the support for "dialing"
             both phone numbers and URIs.

        Changes from draft-sinnreich-sipdev-req-06

        We have updated a number of requirements based on
        discussions on the SIPPING WG list (sipping.ietf.org) and
        helpful comments by Paul Kyzivat and Francois Audet.

          .
Edited and added example for a dial plan in Req-17,

          .
Edited Req-18, Req-19, Req-26, Req-27 and Req-64 so as
            to match recently issued RFCs that are quoted in the
            Reference.

        Changes from draft-sinnreich-sipdev-req-05

        Updated the references and made edits as suggested by Mary
        Barnes and from comments by Russ Housley, David Kessen and
        Ted Hardie.

        Changes from draft-sinnreich-sipdev-req-05

          . Added edits on text over IP has suggested by Gunnar
             Hellstrom and Jon Peterson.

        Changes from draft-sinnreich-sipdev-req-04

          . Removed the section on IANA Considerations that was
             meant to register the event package for automatic
             configuration, since this topic is now dealt elsewhere
             in [60].

          . Removed the reference to RFC 791, since that is implied
             by referencing the DiffServ code points in RFC 2597
             [22].

          . Reviewed and tightened the language based on comments
             by John Elwell.

        Changes from draft-sinnreich-sipdev-req-03



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           . Version 03 of the memo is focused more narrowly on SIP
           telephony device requirements and configuration only.

           . Automatic configuration over the network has been
           ommitted since it is addressed separately in [60].

           . The section with the example with XML based
           configuration data has been omitted, since such data
           formats are different topic altogether.

           . The section on security considerations has been re-
           written from scratch so as to keep up with recent work on
           SIP security, and such items as user identity,
           certificates, S/MIME and the SIP Authenticated Body (AIB)
           format.

        Changes to -02 since draft-sinnreich-sipdev-req-01

           . Re-edited the section on Interactive text support for
           hearing or speech disabled users.

           . Shortened the sections on phonebook, call history and
           line related settings.

           . Deleted the section on ringer behavior.

           . Updated and added references.


     8. References

        Note: The references provided here should be considered
        informative, since this is an informational memo. Also, as
        of this writing, some references are work in progress at the
        IETF. As a result the version number on some key draft may
        be obsolete at the time of reading this memo and other
        Internet Drafts are advanced to RFC status.

        [1] Scott Bradner: "The Internet Standards Process, Revision
        3", RFC 2026. IETF, October 1996.

        [2] Scott Bradner: "Key words for use in RFCs to Indicate
        Requirement Levels", RFC 2119, IETF, 1997.

        [3] J. Rosenberg et. al: "SIP: Session Initiation Protocol",
        RFC 3261. IETF, June 2002.

        [4] T. Lemon et al: "Encoding Long Options in the DHCP", RFC
        3396. IETF, November 2002.


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        [5] D. Mills: "Simple Network Time Protocol (SNTP) Version 4
        for IPv4 and IPv6 and OSI" RFC 2030. IETF, October 1996.

        [6] J. Rosenberg and H. Schulzrinne: "Session Initiation
        Protocol (SIP): Locating SIP Servers", RFC 3263. IETF, June
        2002.

        [7] J.Peterson "ENUM Service Registration for Session
        Initiation Protocol (SIP) Address of Record", RFC 3764.
        IETF, April 2004.

        [8] R J. Peterson: "A Privacy Mechanism for the Session
        Initiation Protocol", RFC 3323. IETF, November 2002.

        [9] H. Schulzrinne: "The tel URI for Telephone Numbers", RFC
        3966. IETF, December 2004.

        [10] R. Sparks: "The Session Initiation Protocol (SIP) Refer
        Method", RFC 3515. IETF, April 2003.

        [11] H. Schulzrinne and S. Petrack: RTP Payload for DTM
        Digits, Telephony Tones and Telephony Signals", RFC 2833.
        IETF, May 2000.

        [12] J. Rosenberg and H. Schulzrinne: "An Offer/Answer Model
        with the Session Description Protocol (SDP", RFC 3264. IETF,
        June 2002.

        [13] S. Casner and P. Hoschka: S. "MIME Type Registration of
        RTP Payload Formats", RFC 3555. IETF, July 2003. Updated by
        RFC 3625.

        [15] A. Johnston et al: "Session Initiation Protocol (SIP)
        Basic Call Flow Examples", RFC 3665. IETF, December 2003.

        [14] G. Camarillo et al: "Grouping of Media Lines in the
        Session Description Protocol (SDP)" RFC 3388. IETF, December
        2002.

        [16] A. Johnston: "Session Initiation Protocol (SIP) Public
        Switched Telephone Network (PSTN) Call Flows", RFC 3666.
        IETF, December 2003.

        [17] J. Rosenberg et al: "Best Current Practices for Third
        Party Call Control (3pcc) in the Session Initiation Protocol
        (SIP)", RFC 3725. IETF, April 2004.




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        [18] N. Charlton et al: "User Requirements for the Session
        Initiation Protocol (SIP) in Support of Deaf, Hard of
        Hearing and Speech-impaired Individuals". RFC 3351. IETF,
        August 2002.

        [19] M. Handley and V. Jacobson: "SDP: Session Description
        Protocol", RFC 2327. IETF, April 1998.

        [20] H. Schulzrinne et al: "RTP: A Transport Protocol for
        Real-Time Applications", RFC 3550. IETF, July 2003.

        [21] T. Friedman et al: "RTP Control Protocol Extended
        Reports (RTCP XR)", RFC 3611. IETF, November 2003.

        [22] J. Heinanen et al: "Assured Forwarding PHB Group", RFC
        2597. IETF, June 1999.

        [23] R. Braden et al: "Resource ReSerVation Protocol (RSVP)-
        Version 1 Functional Specification", RFC 2205. IETF,
        September 1997.

        [24] H. Schulzrinne and S. Casner: "RTP Profile for Audio
        and Video Conferences with Minimal Control", RFC 3551. IETF,
        July 2003.

        [25] ITU-T Recommendation G.711 available online from the
        ITU bookstore at http://www.itu.int.

        [26] S.V. Anderson et al: "Internet Low Bit Rate Codec", RFC
        3951. IETF, December 2004.

        [27] R A. Duric: "RTP Payload Format for iLBC Speech", RFC
        3952. IETF, December 2004.

        [28] G. Herlein et al.: "RTP Payload Format for the Speex
        Codec", draft-herlein-speex-rtp-profile-02, IETF, April
        2005.

        [29] TIA/EIA-810-A, "Transmission Requirements for
        Narrowband Voice over IP and Voice over PCM Digital Wireline
        Telephones", July 2000.

        [30] TIA-EIA-IS-811, "Terminal Equipment - Performance and
        Interoperability Requirements for Voice-over-IP (VoIP)
        Feature Telephones", July 2000.

        [31] ITU-T Recommendation G.177 available online from the
        ITU bookstore at http://www.itu.int



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        [32] J. Rosenberg et al: "STUN - Simple Traversal of User
        Datagram Protocol (UDP) Through Network Address Translators
        (NATs)" RFC 3489. IETF, March 2003.

        [33] P. Eggert, "Sources for time zone and daylight saving
        time data." Available at http://www.twinsun.com/tz/tz-
        link.htm

        [34] H. Alvestrand: "Tags for the Identification of
        Languages" RFC 3066. IETF, January 2001.

        [35] B. Campbell and R. Sparks: "Control of Service Context
        using SIP Request-URI" RFC 3087. IETF, April 2001.

        [36] T. Dierks: "The TLS protocol Version 1.0" RFC 2246.
        IETF, January 1999. Updated by RFC 3546.

        [37] C. Jennings et al: "Private Extensions to the Session
        Initiation Protocol (SIP) for Asserted Identity within
        Trusted Networks", RFC 3325. IETF, November 2002.

        [38] J. Peterson: "A Privacy Mechanism for the Session
        Initiation Protocol (SIP)", RFC 3323. IETF, Nov. 2002.

        [39] S. Chokhani et al: "Internet X.509 Public Key
        Infrastructure, Certificate Policy and Certification
        Practices Framework" RFC 3647. IETF, Nov. 2003.

        [40] B. Ramsdell: "S/MIME Version 3.1 Message Specification"
        RFC 3851. IETF, July 2004.

        [41] M. Baugher et al: "The Secure Real-time Transport
        Protocol (SRTP)", RFC 3711. IETF March 2004.

        [42] Mahy, R. et al: "A Call Control and Multi-party usage
        framework for the Session Initiation  Protocol (SIP)",
        draft-ietf-sipping-cc-framework-02. March 2003.
        http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-cc-
        framework-02.html

        [43] R. Mahy: "A Message Summary and Message Waiting
        Indication Event Package for the Session Initiation Protocol
        (SIP)", RFC 3842. IETF, August 2004.

        [44] J. Peterson: "Telephone Number Mapping (ENUM) Service
        Registration for Presence Services". RFC 3953. IETF, January
        2005.




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        [45] O. Levin and A. Johnston: "Conveying Feature Tags with
        Session Initiation Protocol REFER Method", draft-ietf-sip-
        refer-feature-param-00,IETF July 2005.

        [46] A. Johnston: "SIP Service Examples", draft-ietf-
        sipping-service-examples-09, IETF July 2005. Work in
        progress.

        [47] A. Johnston et al: "Session Initiation Protocol (SIP)
        Basic Call Flow Examples" RFC 3665. IETF, December 2003.

        [48] A. Johnston and O. Levin: "Session Initiation Protocol
        Call Control - Conferencing for User Agents", draft-ietf-
        sipping-cc-conferencing-06.txt, IETF, November 2004, work in
        progress.

        [49] R. Even and N. Ismail: "Conferencing Scenarios" draft-
        ietf-xcon-conference-scenarios-05.txt, IETF, September 2005.

        [50] J. Rosenberg et al: "Session Initiation Protocol (SIP)
        Extension for Instant Messaging", RFC 3428. IETF, December
        2002.

        [51] H. Schulzrinne et al.: "RPID: Rich Presence Extensions
        to the Presence Information Data Format (PIDF)", draft-ietf-
        simple-rpid-09, IETF, September 2005.

        [52] J. Rosenberg et al: "Indicating User Agent Capabilities
        in the Session Initiation Protocol (SIP)" RFC 3840. IETF,
        August 2004.

        [53] H. Schulzrinne and B. Rosen: "Emergency Services for
        Internet Telephony Systems", draft-schulzrinne-sipping-
        emergency-arch-02, IETF, October 2004. IETF, work in
        progress.

        [54] See the Working Group on Emergency Context Resolution
        with Internet Technologies at
        http://www.ietf.org/html.charters/ecrit-charter.html

        [55] H. Schulzrinne and J. Polk: "Communications Resource
        Priority for the Session Initiation Protocol", IETF, draft-
        ietf-sip-resource-priority-10, July 2005, work in progress.

        [56] G. Hellstrom and P. Jones: "RTP Payload for Text
        Conversation", RFC 4103, IETF, June 2005.





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        [57] A. Johnston: "A Performance Report Event Package For
        SIP", draft-johnston-sipping-rtcp-summary-07, IETF, July
        2005. Work in progress.

        [58] C. Jennings: " NAT Classification Test Results", draft-
        jennings-behave-test-results-01, IETF, July 2005. Work in
        progress.

        [59] J. Rosenberg: "A Presence Event Package for the Session
        Initiation Protocol (SIP)", RFC 3856. IETF, October 2004.

        [60] D. Petrie: "A Framework for SIP User Agent Profile
        Delivery", draft-ietf-sipping-config-framework-07.txt, IETF,
        July 2005, work in progress.

        [61] C. Jennings: "SIP Tutorial: SIP Security" presented at
        the VON Spring 2004 conference, March 29, 2004, Santa Clara,
        CA.

        [62] J. Rosenberg et al.: "Traversal Using Relay NAT
        (TURN)", draft-rosenberg-midcom-turn-08.txt, IETF, September
        2005, work in progress.

        [63] C. Boulton and J. Rosenberg: "Best Current Practices
        for NAT Traversal for SIP", IETF, October 2004, work in
        progress.

        [64] C. Jennings: "Example call flows using SIP security
        mechanisms", draft-jennings-sip-sec-flows-03, IETF, July
        2005.

        [65] J. Rosenberg et al: "Caller Preferences for the Session
        Initiation Protocol (SIP)", RFC 3841. IETF, August 2004.

        [66] J. Peterson: "Session Initiation Protocol (SIP)
        Authenticated Identity Body (AIB) Format", RFC 3893. IETF,
        September 2004.

        [67] J. Peterson: "S/MIME Advanced Encryption Standard (AES)
        Requirement for the Session Initiation Protocol (SIP), RFC
        3853. IETF, July 2004.

        [68] J. Peterson and C. Jennings: "Enhancements for
        Authenticated Identity Management in the Session Initiation
        Protocol (SIP)", draft-ietf-sip-identity-03, September 2004.

        [69] J. Peterson and C. Jennings: "Certificate Management
        Services for SIP", IETF, October 2004.



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        [70] A. van Wijk: "Framework of requirements for real-time
        text conversation using SIP", draft-ietf-sipping-toip-
        03.txt, IETF, September 2005, work in progress.

       [71] G. Camarillo and H. Schulzrinne: "Early Media and
       Ringing Tone Generation in the Session Initiation Protocol
       (SIP)", RFC 3960. IETF, December 2004.

        [72] "D. Wing: "Symmetric RTP and RTCP Considered Helpful".
        IETF, October 2004, work in progress.

        [73] F. Audet and C. Jennings: "NAT Behavioral Requirements
        for Unicast UDP", draft-ietf-behave-nat-udp-02, IETF, June
        2005, work in progress.

        [74] R. Mahy: "Connection Reuse in the Session Initiation
        Protocol (SIP)", draft-ietf-sip-connect-reuse-04.txt, IETF,
        July 2005, work in progress.

        [75] C. Jennings and R. Mahy: "Managing Client Initiated
        Connections in the Session Initiation Protocol", draft-ietf-
        sip-outbound-00, IETF, July 2005, work in progress.

        [76] J. Rosenberg: "Interactive Connectivity Establishment
        (ICE): A Methodology for Network Address Translator (NAT)
        Traversal for Offer/Answer Protocols", draft-ietf-mmusic-
        ice-05, Internet Draft, IETF, July 2005, work in progress.

        [77] J. Polk and B. Rosen: "Session Initiation Protocol
        Location Conveyance", draft-ietf-sip-location-conveyance-
        01.txt, Internet Draft. October 2004, work in progress.

     9. Author's Addresses

          Henry Sinnreich
          Pulver.com
          115 Broadhollow Road
          Melville, NY 11747, USA
          Email: henry@pulver.com
          Phone: +1-631-961-8950

          Steven Lass
          MCI
          1201 East Arapaho Road
          Richardson, TX 75081, USA
          Email: steven.lass@mci.com
          Phone: +1-972-728-2363



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          Christian Stredicke
          snom technology AG
          Gradestrasse, 46
          D-12347 Berlin, Germany
          Email: cs@snom.de
          Phone: +49(30)39833-0

     10. Copyright Notice


        Copyright (C) The Internet Society (2005).  This document is
        subject to the rights, licenses and restrictions contained
        in BCP 78, and except as set forth therein, the authors
        retain all their rights.

        This document and the information contained herein are
        provided on an "AS IS" basis and THE CONTRIBUTOR, THE
        ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY),
        THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE
        DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT
        NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
        HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
        WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR
        PURPOSE.

        The IETF takes no position regarding the validity or scope
        of any Intellectual Property Rights or other rights that
        might be claimed to pertain to the implementation or use of
        the technology described in this document or the extent to
        which any license under such rights might or might not be
        available; nor does it represent that it has made any
        independent effort to identify any such rights.  Information
        on the procedures with respect to rights in RFC documents
        can be found in BCP 78 and BCP 79.

              Copies of IPR disclosures made to the IETF Secretariat
        and any assurances of licenses to be made available, or the
        result of an attempt made to obtain a general license or
        permission for the use of such proprietary rights by
        implementers or users of this specification can be obtained
        from the IETF on-line IPR repository at
        http://www.ietf.org/ipr.

              The IETF invites any interested party to bring to its
        attention any copyrights, patents or patent applications, or
        other proprietary rights that may cover technology that may
        be required to implement this standard.  Please address the
        information to the IETF at ietf-ipr@ietf.org.



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