SIP Telephony Device Requirements June 2004 SIPPING WG H. Sinnreich/MCI,editor Internet Draft S. Lass/MCI C. Stredicke/snom Expires: February 2005 June 2004 SIP Telephony Device Requirements and Configuration draft-sinnreich-sipdev-req-04.txt Status of this Memo By submitting this Internet-Draft, we certify that any applicable patent or other IPR claims of which we are aware have been disclosed, or will be disclosed, and any of which we become aware will be disclosed, in accordance with RFC 3668. This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026 [1]. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on February 2005. Copyright Notice Copyright (C) The Internet Society (2004). All Rights Reserved. Abstract This informational I-D 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 minimum set of interoperability and multi-vendor supported core features, so as to Sinnreich et al. Expires February 2005 [Page 1]
SIP Telephony Device Requirements June 2004 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. Table of Contents 1. Introduction...................................................3 2. Conventions Used In This Document..............................4 3. Generic Requirements...........................................4 3.1 IP Host Requirements.......................................4 3.2 DNS and ENUM Support.......................................5 3.3 SIP Device Resident Telephony Features.....................5 3.4 Support for SIP Services...................................7 3.5 Basic Telephony and Presence Information Support...........8 3.6 Emergency and Resource Priority Support....................9 3.7 Multi-Line Requirements....................................9 3.8 User Mobility.............................................10 3.9 Interactive Text Support..................................11 3.10 Other Related Protocols..................................12 3.11 SIP Device Security Requirements.........................12 3.12 Quality of Service.......................................13 3.13 Media Requirements.......................................13 3.14 Voice Codecs.............................................13 3.15 Telephony Sound Requirements.............................14 3.16 International Requirements...............................15 3.17 Support for Applications.................................15 3.18 Web Based Feature Management.............................15 3.19 Firewall and NAT Traversal...............................16 3.20 Device Interfaces........................................16 4. Glossary and Usage for the Configuration Settings.............17 4.1 Device ID.................................................18 4.2 Signaling Port............................................18 4.3 RTP Port Range............................................18 4.4 Quality of Service........................................18 4.5 Default Call Handling.....................................19 4.6 Outbound Proxy............................................19 4.7 Default Outbound Proxy....................................19 4.8 SIP Session Timer.........................................19 4.9 Telephone Dialing Functions...............................19 4.10 Phone Number Representations.............................20 4.11 Digit Maps and/or the Dial/OK Key........................20 4.12 Default Digit Map........................................21 4.13 SIP Timer Settings.......................................21 4.14 Audio Codecs.............................................21 4.15 DTMF Method..............................................21 4.16 Local and Regional Parameters............................22 4.17 Time Server..............................................22 Sinnreich et al. Expires February 2005 [Page 2]
SIP Telephony Device Requirements June 2004 4.18 Language.................................................22 4.19 Inbound Authentication...................................23 4.20 Voice Message Settings...................................23 4.21 Phonebook and Call History...............................23 4.22 User Related Settings and Mobility.......................24 4.23 AOR Related Settings.....................................24 4.24 Maximum Connections......................................25 4.25 Automatic Configuration and Upgrade......................25 4.26 Security Configurations..................................25 5. Security Considerations.......................................26 5.1 Threats and Problem Statement.............................26 5.2 SIP Telephony Device Security.............................27 6. IANA Considerations...........................................29 7. Changes from draft-sinnreich-sipdev-req-03....................30 8. References....................................................30 8.1 Normative References......................................30 8.2 Informative References....................................33 9. Acknowledgments...............................................35 10. Author's Addresses...........................................36 11. Intellectual Property and Validity Statement.................36 1. Introduction This informational I-D 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 device enabled for SIP based IP telephony. SIP telephony 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 telephony devices can also be instant messaging (IM) applications that have a telephony option. SIP devices MAY also 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, as will be shown. Sinnreich et al. Expires February 2005 [Page 3]
SIP Telephony Device Requirements June 2004 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 minimum 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. 2. 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. 3. 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. 3.1 IP Host Requirements This section applies mainly to desktop phones and other special purpose SIP hardware. Some of the requirements in this section are not applicable to PC/laptop or PDA software phones (soft phones) and mobile phones. 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 that display the time MUST support the Simple Network Time Protocol [5]. Sinnreich et al. Expires February 2005 [Page 4]
SIP Telephony Device Requirements June 2004 REQ-5: Desktop SIP phones and other special purpose SIP telephony devices MUST be able to upgrade their firmware to support additional features and improved 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 upgrades down to the devices remotely. 3.2 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 support 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. 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 SHOULD support ENUM [7]. 3.3 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 [8]. REQ-14: SIP telephony devices SHOULD 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. Sinnreich et al. Expires February 2005 [Page 5]
SIP Telephony Device Requirements June 2004 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. REQ-17: SIP telephony devices MAY support a local dial plan. The dial plan MUST consist of a pattern string to match dial digits, and the ability to strip and also append prefix digits, and also append suffix digits. REQ-18: SIP telephony devices MUST support the URLs for Telephone numbers [9]. See also the amended version [44]. REQ-19: SIP telephony devices MUST support REFER and NOTIFY as required to support call transfer [45], [46]. SIP telephony devices MUST support 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 attended call transfer as defined in [46]. REQ-22: SIP telephony devices MAY support device based 3-way calling by mixing the audio streams 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 Sinnreich et al. Expires February 2005 [Page 6]
SIP Telephony Device Requirements June 2004 devices SHOULD support Flow Identification as defined in RFC 3388 [11]. REQ-26: SIP telephony devices MUST generate local ringing and MUST ignore any early RTP media when a 180 Ringing response is received. REQ-27: SIP telephony devices MUST play the first RTP stream and ignore any other RTP media streams when a 183 Session Progress response is received. REQ-28: SIP telephony devices MUST obey the last 18x message received when multiple 18x responses are received. If the last response is 180 Ringing, the client MUST generate local ringing. If the last response is 183 Session Progress, the client MUST play the RTP stream. REQ-29: SIP devices with a display MUST support the call-info header and depending on the display capabilities MAY for example display an icon or the image of the caller. REQ-30: To provide additional information about call failures, SIP telephony devices MUST display the Reason Phrase of the SIP message instead of mapping the Status-Code to custom or default messages. The devices MAY use an internal Status Code table if there was a problem with the language negotiation. REQ-31: SIP telephony devices MUST support music on hold as shown in "SIP Service Examples" [46]. REQ-32: SIP telephony devices MAY ring after a call has been on hold for a predetermined period of time, typically 3 minutes. 3.4 Support for SIP Services REQ-33: SIP telephony devices MUST support the SIP Basic Call Flow Examples [47]. Sinnreich et al. Expires February 2005 [Page 7]
SIP Telephony Device Requirements June 2004 REQ-34: SIP telephony devices MUST support the SIP-PSTN Service Examples [16]. REQ-35: SIP telephony devices MUST support the Third Party Call Control model [17]. REQ-36: SIP telephony devices SHOULD support other SIP call control and multiparty usage [43]. REQ-37: SIP telephony devices SHOULD support conferencing services for voice and IM [48], [49]. REQ-38: SIP telephony devices MUST support caller and called party preferences [52]. REQ-39: SIP telephony devices MAY support service mobility: Devices MAY allow roaming users to upload 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/called party preferences. 3.5 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 for example presence states, 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 devices can support for example a voice session and several IM sessions with different parties. REQ-40: 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. Sinnreich et al. Expires February 2005 [Page 8]
SIP Telephony Device Requirements June 2004 REQ-41: Users MUST be able to set the state of the SIP telephony device to Do Not Disturb, manifested as a Presence state across the network. REQ-42: SIP telephony devices with Do Not Disturb enabled MUST respond to new sessions with 486 Busy Here. 3.6 Emergency and Resource Priority Support REQ-43: Emergency calling: For emergency numbers (e.g. 911, SOS URL) the client MUST send the location information acquired by various means as detailed in [53]. SIP telephony devices MUST be provisioned with a URL or phone number of the emergency call center and MUST support the SOS URL [54]. REQ-44: Priority header: SIP devices MUST support the Priority header specified in RFC 3261 for such applications as emergency calls or for selective call acceptance. REQ-45: 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. 3.7 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 Sinnreich et al. Expires February 2005 [Page 9]
SIP Telephony Device Requirements June 2004 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-46: 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-47: 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-48: 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. REQ-49: Multi-line SIP telephony devices MUST be able to provision a different ring tone for each line. 3.8 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-50: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-51: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 SIP proxy and the associated policy server SHOULD be available to the user. Sinnreich et al. Expires February 2005 [Page 10]
SIP Telephony Device Requirements June 2004 REQ-52:User mobility enabled SIP telephony devices for the desktop MUST use the local static location data associated with the device for emergency calls. 3.9 Interactive Text Support REQ-53: SIP telephony devices such as SIP display phones and IP-analog adapters SHOULD support the accessibility for user requirements for the deaf, hard of hearing and speech impaired individuals as per RCF 3351 [18] and also for interactive text conversation [56], [70]. Note: SIP telephony devices supporting Instant Messaging based on SIMPLE [50] support text conversation based on blocks of text. However, interactive text conversation is often preferred here due to its interactive and more streaming-like nature. REQ-54: 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-55: SIP telephony devices SHOULD provide an external standard wired or wireless link to connect external input (keyboard, mouse) and display devices. REQ-56: SIP-33: 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-57: 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-58: SIP telephony devices MAY provide analog adaptor functionality through an RJ-11 FXO port to support FXS devices. If an RJ-11 Sinnreich et al. Expires February 2005 [Page 11]
SIP Telephony Device Requirements June 2004 (FXO) 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 MUST pass- through all text-telephone protocols according to ITU-T Recommendation V.18, November 2000, in a transparent fashion. REQ-59: SIP telephony devices MAY provide a 2.5 mm audio port, in portable SIP devices, such as PDAs and various wireless SIP phones. 3.10 Other Related Protocols REQ-60: SIP telephony devices MUST support 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 [21] and MAY also support the RTCP summary report delivery [57]. 3.11 SIP Device Security Requirements This paragraph is focused on the requirements for SIP device security only. Section 4 on Security Consideration deals with some wider security issues. REQ-61: SIP telephony devices MUST support digest authentication as per RFC3261. In addition, SIP telephony devices SHOULD 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. REQ-62: SIP telephony devices MUST be able to password protect configuration information and administrative functions. REQ-63: SIP telephony devices MUST NOT display the password to the user or administrator after it has been entered. REQ-64: 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. Sinnreich et al. Expires February 2005 [Page 12]
SIP Telephony Device Requirements June 2004 REQ-65: 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 AOR. This provides protection against war-dialer attacks. The setting to accept/reject MUST be configurable. REQ-66: 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. 3.12 Quality of Service REQ-67: SIP devices MUST support the IPv4 DSCP field for RTP streams as per RFC 2597 [22] that supersedes the Type of Service (ToS) bits described in RFC 791. The DSCP setting MUST be configurable to complement the local network policy. REQ-68: 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-69: SIP telephony devices MAY support RSVP [23]. 3.13 Media Requirements REQ-70: 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. REQ-71: To reduce overall bandwidth, SIP telephony devices MAY support active voice detection and comfort noise generation. 3.14 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 Sinnreich et al. Expires February 2005 [Page 13]
SIP Telephony Device Requirements June 2004 intellectual property and licensing problems associated with most codec types. RFC 3551 [24] lists 17 registered MIME subtypes for audio codecs. This memo however requires the support of a minimal number of five codecs used in wireline VoIP, besides various codecs found in mobile phones. REQ-72: SIP telephony devices SHOULD support AVT payload type 0 (G.711 uLaw) as the default codec [25] and its Annexes 1 and 2. REQ-73: SIP telephony devices MUST support the Internet Low Bit Rate codec (iLBC) [26], [27]. REQ-74: SIP telephony devices SHOULD support GSM codecs found in various 3G wireless phones. REQ-75: SIP telephony devices MAY support G.723.1, where low bandwidth is needed. REQ-76: SIP telephony devices MAY support G.729 and its annexes. Note: A summary count reveals up to 25 and more voice codec types currently in use. The authors believe there is a need for a single multi-rate Internet codec, such as Speex [28] that can effectively be substituted for all of the multiple legacy narrow band compressed G.xx codec types, such as G. 711, G.729, G.723.1, G.722, etc., thus avoiding the complexity and cost to implementers and service providers alike who are burdened by supporting so many codec types, besides the additional licensing costs. The authors believe the Internet Low Bit Rate codec (iLBC) should be the default codec for Internet telephony. 3.15 Telephony Sound Requirements REQ-77: SIP telephony devices SHOULD comply with the handset receive comfort noise requirements outlined in the ANSI standards [29], [30]. REQ-78: SIP telephony devices SHOULD comply with the stability or minimum loss defined in ITU-T G.177 [31]. Sinnreich et al. Expires February 2005 [Page 14]
SIP Telephony Device Requirements June 2004 REQ-79: SIP telephony devices MAY provide a full-duplex speakerphone 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-80: SIP telephony device MAY support different ring-tones based on the caller identity. 3.16 International Requirements REQ-81: SIP telephony devices SHOULD indicate the preferred language [34] using Caller Preferences [52]. REQ-82: SIP telephony devices intended to be used in various language settings [34], MUST support other languages for menus, help, and labels. 3.17 Support for Applications The following requirements apply to functions placed in the SIP telephony device. REQ-83: SIP telephony devices that have a large display and support presence SHOULD display a buddy list [50]. REQ-84: SIP telephony devices MAY support LDAP for client-based directory lookup. REQ-85: SIP telephony devices MAY support a phone setup where a URL is automatically dialed when the phone goes off-hook. 3.18 Web Based Feature Management REQ-86: SIP telephony devices SHOULD support an internal web server to allow users to manually configure the phone and to set up personal phone services such as the address book, speed- dial, ringer 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. Managing SIP telephony devices SHOULD NOT require special client Sinnreich et al. Expires February 2005 [Page 15]
SIP Telephony Device Requirements June 2004 software on the PC or require a dedicated management console. SIP telephony devices SHOULD support https transport for this purpose. 3.19 Firewall and NAT Traversal The following requirements allow SIP clients to properly function behind various firewall architectures. REQ-87: 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. REQ-88: 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]. Note: Developers are advised to follow the standards process for ICE [63] and eventually support ICE in SIP telephony devices. REQ-89: SIP telephony devices SHOULD support UPnP (http://www.upnp.org/) for NAPT device traversal. REQ-90: SIP telephony devices MUST be able to limit the ports used for RTP to a provisioned range. 3.20 Device Interfaces SIP telephony devices MAY have various types of interfaces, such as resembling a desktop phone, cordless phone, mobile phone, handheld computer, laptop computer and MAY have various interface models, such as for phones, IM GUI or other. Given the variety of possible interfaces, only the generic requirements can be listed here. REQ-91: SIP telephony devices MUST have two types of interface capabilities, for both phone numbers and URLs, both accessible to the end user. Sinnreich et al. Expires February 2005 [Page 16]
SIP Telephony Device Requirements June 2004 REQ-92: 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 and mobile phones and wireless phones. REQ-93: SIP telephony devices MUST have a convenient way for entering SIP URLs and phone numbers. This includes all alphanumeric characters allowed in legal SIP URLs. Possible approaches include using a web page, display and keyboard entry, type-ahead or graffiti for PDAs. REQ-94: SIP telephony devices should allow phone number entry in human friendly fashion, with the usual separators and brackets between digits and digit groups. 4. 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. Sinnreich et al. Expires February 2005 [Page 17]
SIP Telephony Device Requirements June 2004 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. The examples shown here are just of informational nature. Other documents may specify the syntax and semantics for the respective settings. 4.1 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 4.2 Signaling Port The port that MUST 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. For UDP, the port must 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 4.3 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 4.4 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 Sinnreich et al. Expires February 2005 [Page 18]
SIP Telephony Device Requirements June 2004 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 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 4.5 Default Call Handling All of the call handling settings defined below can be defined here as default behaviors. 4.6 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 NAT application layer gateways are always in the signaling path. The second requirement allows the optimization of the routing by the outbound proxy. Example: OutboundProxy=sip:nat.proxy.com 4.7 Default Outbound Proxy The default outbound proxy SHOULD be a global setting (not related to a specific line). Example: DefaultProxy=sip:123@proxy.com 4.8 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 4.9 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 Sinnreich et al. Expires February 2005 [Page 19]
SIP Telephony Device Requirements June 2004 the rule by which digits are transformed into a URL (usually SIP URL or TEL URL). 4.10 Phone Number Representations SIP phones need to understand entries in the phone book of the most common separators used between 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. 4.11 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 URL 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]*)#$|sip:\1@\d;user=phone|outbound=proxy.com Sinnreich et al. Expires February 2005 [Page 20]
SIP Telephony Device Requirements June 2004 |^([a-zA-Z0-9&=+\$,;?\-_.!~*'()%]+@.+)|sip:\1| |^([a-zA-Z0-9&=+\$,;?\-_.!~*'()%]+)$|sip:\1@\d| |^(.*)$|sip:\1@\d|timeout=5 4.12 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 URL 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. 4.13 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. 4.14 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 MUST be configurable along with the order of preference. Service providers MUST 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 MAY include hints about privacy for audio using SRTP that either mandate or encourage the usage of secure RTP. Example: SRTP=mandatory 4.15 DTMF Method Sinnreich et al. Expires February 2005 [Page 21]
SIP Telephony Device Requirements June 2004 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 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]. 4.16 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 4.17 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 4.18 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 Sinnreich et al. Expires February 2005 [Page 22]
SIP Telephony Device Requirements June 2004 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 even if it cannot display those characters in the user interface. 4.19 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. 4.20 Voice Message Settings Various voice message settings require the use of URL's 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 notifications without prior subscription. This way the setup of voice message settings can be avoided. 4.21 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 Sinnreich et al. Expires February 2005 [Page 23]
SIP Telephony Device Requirements June 2004 that keeps the information even after restarting the device or save the information in an external database that permanently stores the information. 4.22 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 a 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. 4.23 AOR Related Settings Sinnreich et al. Expires February 2005 [Page 24]
SIP Telephony Device Requirements June 2004 SIP telephony devices MUST use the AOR related settings, as specified here. AOR Identification There are many properties which MAY be associated with or SHOULD be applied to the AOR or signaling 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. 4.24 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 4.25 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 settings by the user. Image and service data upgrades SHOULD also not require any settings by the user. 4.26 Security Configurations Sinnreich et al. Expires February 2005 [Page 25]
SIP Telephony Device Requirements June 2004 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]. 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. It is RECOMMENDED not to include passwords through the automatic configuration process. Users SHOULD enter the passwords locally. 5. Security Considerations 5.1 Threats and Problem Statement While section 2.12 and 2.20 state 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 in other parts of SIP enabled networks may reside. 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 peoples 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, Sinnreich et al. Expires February 2005 [Page 26]
SIP Telephony Device Requirements June 2004 . 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. 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. 5.2 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 connected via a VPN channel to the private IP network) SHOULD 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 that operates 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 prepaid calls MUST place unknown in the From header. Sinnreich et al. Expires February 2005 [Page 27]
SIP Telephony Device Requirements June 2004 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 (CAs). End-to-End Security Using S/MIME S/MIME [40] MAY be used 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. Privacy Media Encryption Secure RTP (SRTP) [41] MAY be used for the encryption of media such as audio and video, after the keying information has been passed by SIP signaling. Instant messaging MAY be protected end-to-end using S/MIME. Support for NAT and Firewall Traversal The various NAT and firewall traversal scenarios require support in telephony SIP devices. 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 cases an external media relay must also be provided that can support the TURN protocol exchange [62] with SIP telephony devices. Media relays such as TURN come at a high bandwidth cost to the service provider, since the bandwidth for all 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 NATs in the network, so that only STUN can be used, where required. 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 ICE [63] has been proposed to be deployed in all devices that required end-to- end connectivity for SIP signaling and RTP media streams, as Sinnreich et al. Expires February 2005 [Page 28]
SIP Telephony Device Requirements June 2004 well as for streaming media using RTSP. ICE makes use of the STUN, TURN and RSIP protocols by using extensions to SDP. 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 [69]. We recommend developers and network managers to follow this work as it will develop into IETF standards. 6. IANA Considerations SIP Event Package Registration for Configuration Package name: SIP Telephony Device Configuration Type: package Contact: [Christian Stredicke] Published Specification: This document. MIME Registration for Application The MIME Registration for application/sip-endpoint-configuration is: MIME media type name: application MIME subtype name: sip-endpoint-configuration Required parameters: none. Optional parameters: none. Encoding considerations: See SIP [3] message header definition. Security considerations: See the "Security Considerations" in Section 8 n this document. Interoperability considerations: none Published specification: This document. Sinnreich et al. Expires February 2005 [Page 29]
SIP Telephony Device Requirements June 2004 Applications which use this media: SIP configuration server and clients subscribing to these servers. Additional information: 1. Magic number(s): N/A 2. File extension(s): N/A 3. Macintosh file type code: N/A. 7. Changes from draft-sinnreich-sipdev-req-03 . This version (0.5) 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 8.1 Normative References [1] RFC2026: "The Internet Standards Process, Revision 3" by Scott Bradner, IETF, October 1996. [2] RFC 2119: "Key words for use in RFCs to Indicate Requirement Levels" by Scott Bradner, IETF, 1997. [3] RFC 3261: "SIP: Session Initiation Protocol by J. Rosenberg et. al, IETF, June 2002. [4] RFC 2131: Dynamic Host Configuration Protocol by R. Droms, IETF, March 1997. Sinnreich et al. Expires February 2005 [Page 30]
SIP Telephony Device Requirements June 2004 [5] RFC 2030: Simple Network Time Protocol (SNTP) Version 4 for IPv4 and IPv6 and OSI by D. Mills, IETF, October 1996. [6] RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers by J. Rosenberg and H. Schulzrinne, IETF, June 2002. [7] RFC 3764: ENUM Service Registration for Session Initiation Protocol (SIP) Address of Record by J. Peterson, IETF, April 2004. [8] RFC 3323: A Privacy Mechanism for the Session Initiation Protocol by J. Peterson, IETF, November 2002. [9] RFC 2806: URLs for Telephone Calls by A. Vaha-Sipila, IETF, April 2000.3 [10] RFC 3515: The Session Initiation Protocol (SIP) Refer Method by R. Sparks. IETF, April 2003. [11] RFC 2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, by H. Schulzrinne and S. Petrack. IETF, May 2000. [12] RFC 3264: An Offer/Answer Model with the Session Description Protocol (SDP) by J. Rosenberg and H. Schulzrinne. IETF, June 2002. [13] RFC 3555: S. MIME Type Registration of RTP Payload Formats by S. Casner and P. Hoschka, IETF, July 2003. [15] RFC 3665: Session Initiation Protocol (SIP) Basic Call Flow Examples by A. Johnston et al., IETF, December 2003. [14] RFC 3388: Grouping of Media Lines in the Session Description Protocol (SDP)" by G. Camarillo et al. IETF, December 2002. [16] RFC 3666: Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows by A. Johnston, IETF December 2003. [17] RFC 3725: Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP) by J. Rosenberg et al. IETF, April 2004. [18] RFC 3351: User Requirements for the Session Initiation Protocol (SIP) in Support of Deaf, Hard of Hearing and Speech-impaired Individuals. IETF, August 2002. [19] RFC 2327: SDP: Session Description Protocol by M. Handley and V. Jacobson. IETF, April 1998. Sinnreich et al. Expires February 2005 [Page 31]
SIP Telephony Device Requirements June 2004 [20] RFC 3550: RTP: A Transport Protocol for Real-Time Applications by H. Schulzrinne et al. IETF, July 2003. [21] RFC 2611: RTP Control Protocol Extended Reports (RTCP XR) by T. Friedman et al. IETF, November 2003. [22] RFC 2597: "Assured Forwarding PHB Group" by Heinanen, J. et al. IETF, June 1999. [23] RFC 2205: Resource ReSerVation Protocol (RSVP)- Version 1 Functional Specification by R. Braden et al. IETF, September 1997. [24] RFC 3551: RTP Profile for Audio and Video Conferences with Minimal Control. 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, draft- ietf-avt-ilbc-codec-04.txt, IETF, November 2003. [27] A. Duric: RTP Payload Format for iLBC Speech, draft-ietf-avt- rtp-ilbc-04.txt, IETF, November 2003. [28] G. Herlein et al.: RTP Payload Format for the Speex Codec, draft-herlein-avt-rtp-speex-00.txt, IETF, March 2003. [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 [32] RFC 3489: STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) by J. Rosenberg et al. 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] RFC 3066: Tags for the Identification of Languages by H. Alvestrand. IETF, January 2001. [35] RFC 3087: Control of Service Context using SIP Request-URI by B. Campbell and R. Sparks. IETF, April 2001. Sinnreich et al. Expires February 2005 [Page 32]
SIP Telephony Device Requirements June 2004 [36] RFC 2246: The TLS protocol Version 1.0 by T. Dierks. IETF, January 1999. [37] RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks C. Jennings et al., IETF, November 2002. [38] RFC 3323: A Privacy Mechanism for the Session Initiation Protocol (SIP), by J. Peterson, IETF, Nov. 2002. [39] RFC 3647: Internet X.509 Public Key Infrastructure, Certificate Policy and Certification Practices Framework by S. Chokhani et al., IETF, Nov. 2003 [40] RFC 2633: S/MIME Version 3 Message Specification" by B. Ramsdell, IETF, June 1999. [41] RFC 3711: The Secure Real-time Transport Protocol (SRTP) by M. Baugher et al., IETF March 2004. 8.2 Informative References [42] Mahy, R. et al: "A Call Control and Multi-party usage framework for the Session Initiation Protocol (SIP)", draft-ietf-sipping-cc- framework-04. October 2003. [43] R. Mahy: A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP), draft-ietf- sipping-mwi-04.txt, IETF, December 2003, work in progress. [44] H. Schulzrinne: The tel URI for Telephone Numbers, draft-ietf- iptel-rfc2806bis-07, IETF April 2004, work in progress. [45] S. Olson and O. Levin: Extended-REFER framework and other REFER extensions,draft-olson-sipping-refer-extensions-01,IETF February 2004, work in progress. [46] A. Johnston: SIP Service Examples, draft-ietf-sipping-service- examples-06, IETF February 2004. Work in progress. [47] RFC 3666: Public Switched Telephone Network (PSTN) Call Flows by A. Johsnton et al. IETF, December 2003. [48] A. Johnston and O. Levin: Session Initiation Protocol Call Control - Conferencing for User Agents, IETF, February 2004, work in progress. Sinnreich et al. Expires February 2005 [Page 33]
SIP Telephony Device Requirements June 2004 [49] R. Even and N. Ismail: Conferencing Scenarios draft-ietf-xcon- conference-scenarios-00.txt, IETF, November 2003, work in progress. [50] Rosenberg, J.,"SIP Extensions for Presence", draft-ietf-simple- presence-10 (work in progress), January 2003. [51] H. Schulzrinne et al.: RPID: Rich Presence Extensions to the Presence Information Data Format (PIDF), draft-ietf-simple-rpid-03,IETF, September 2003. [52] J. Rosenberg et al: Caller Preferences for the Session Initiation Protocol (SIP), draft-ietf-sip-callerprefs, IETF, October 2003, work in progress. [53] H. Schulzrinne and B. Rosen: Emergency Services for Internet Telephony Systems, draft-schulzrinne-sipping-emergency-arch, IETF, February 2004. Work in prgress. [54] H. Schulzrinne: Emergency Services URI for the Session Initiation Protocol, draft-ietf-sipping-sos-00. IETF, February 2004. Work in progress. [55] H. Schulzrinne and J. Polk: Communications Resource Priority for the Session Initiation Protocol, IETF, draft-ietf-sip-resource- priority-03, March 2004. [56] G. Hellström et al.: Framework of requirements for real-time text conversation using SIP. IETF, February 2004. [57] A. Johnston: RTCP Summary Report Delivery to Third Parties, draft-johnston-sipping-rtcp-summary-02, IETF, February 2004. Work in progress. [58] C. Jennings: NAT Classification Results using STUN, draft- jennings-midcom-stun-results-00, IETF, February 2004. Work in progress. [59] R. Mahy: A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP), draft-ietf- sipping-mwi-04.txt, IETF, December 2003. [60] D. Petrie: A Framework for SIP User Agent Profile Delivery, draft-ietf-sipping-config-framework-03.txt, IETF, May 2004. [61] C. Jennings: SIP Tutorial: SIP Security presented at the VON Spring 2004 conference, March 29, 2004, Santa Clara, CA. Sinnreich et al. Expires February 2005 [Page 34]
SIP Telephony Device Requirements June 2004 [62] J. Rosenberg et al.: Traversal Using Relay NAT (TURN), IETF, Feb. 2004, work in progress. [63] J. Rosenberg: Interactive Connectivity Establishment (ICE): A Methodology for Network Address Translator (NAT) Traversal for Multimedia Session Establishment Protocols, IETF, February 2004, work in progress. [64] C. Jennings: Example call flows using SIP security mechanisms, draft-jennings-sip-sec-flows-01, IETF, February 2004, work in progress. [65] J. Rosenberg et al.: Caller Preferences for the Session Initiation Protocol (SIP), draft-ietf-sip-callerprefs-10, IETF October 2003. Work in progress. [66] J. Peterson: SIP Authenticated Identity Body (AIB) Format, draft-ietf-sip-authid-body, IETF, May 2003. [67] J. Peterson: S/MIME AES Requirements for SIP draft-ietf-sip- smime-aes, IETF, June 2003, work in progress. [68] J. Peterson and C. Jennings: Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP), draft- ietf-sip-identity, May 2004, work in progress. [69] J. Peterson and C. Jennings: Certificate Management Services for SIP, draft-jennings-sipping-certs, May 2004. [70] RFC 2793bis: "RTP Payload for Text Conversation" by H. Hellstrom and P. Jones. Internet Draft, work in progress. IETF, June 2004. 9. Acknowledgments 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. 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. 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 Internet Draft: Henning Schulzrinne, Jörgen Björkner, Jay Batson, Eric Tremblay, Gunnar Sinnreich et al. Expires February 2005 [Page 35]
SIP Telephony Device Requirements June 2004 Hellström, 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 parts of this Internet Draft. 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. As mentioned, their contributions are now the focus of separate documents. 10. Author's Addresses Henry Sinnreich MCI 400 International Parkway Richardson, TX 75081, USA Email: henry.sinnreich@mci.com Phone : +1-972-729-4983 Steven Lass MCI 400 International Parkway Richardson, TX 75081, USA Email: steven.lass@mci.com Phone: +1 972 729 4469 Christian Stredicke snom technology AG Pascalstrasse 10e 10587 Berlin, Germany Email: cs@snom.de Phone: +49(30)39833-0 11. Intellectual Property and Validity Statement 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 Sinnreich et al. Expires February 2005 [Page 36]
SIP Telephony Device Requirements June 2004 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. Disclaimer of Validity 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. Copyright Statement Copyright (C) The Internet Society (2004). 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. Acknowledgment Funding for the RFC Editor function is currently provided by the Internet Society. Sinnreich et al. Expires February 2005 [Page 37]
