SIPPING WG H. Sinnreich/pulver.com, editor
Internet Draft S. Lass/MCI
C. Stredicke/snom
June 12, 2005
SIP Telephony Device Requirements and Configuration
draft-sinnreich-sipdev-req-07.txt
Status of this Memo
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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.....................5
2.4. Support for SIP Services...................................8
2.5. Basic Telephony and Presence Information Support...........8
2.6. Emergency and Resource Priority Support....................9
2.7. Multi-Line Requirements...................................10
2.8. User Mobility.............................................10
2.9. Interactive Text Support..................................11
2.10. Other Related Protocols...................................12
2.11. SIP Device Security Requirements..........................12
2.12. Quality of Service........................................13
2.13. Media Requirements........................................13
2.14. Voice Codecs..............................................13
2.15. Telephony Sound Requirements..............................14
2.16. International Requirements................................15
2.17. Support for Related Applications..........................15
2.18. Web Based Feature Management..............................15
2.19. Firewall and NAT Traversal................................15
2.20. Device Interfaces.........................................16
3. Glossary and Usage for the Configuration Settings..............17
3.1. Device ID.................................................17
3.2. Signaling Port............................................18
3.3. RTP Port Range............................................18
3.4. Quality of Service........................................18
3.5. Default Call Handling.....................................18
3.5.1. Outbound Proxy.......................................18
3.5.2. Default Outbound Proxy...............................19
3.5.3. SIP Session Timer....................................19
3.6. Telephone Dialing Functions...............................19
3.6.1. Phone Number Representations.........................19
3.6.2. Digit Maps and/or the Dial/OK Key....................19
3.6.3. Default Digit Map....................................20
3.7. SIP Timer Settings........................................20
3.8. Audio Codecs..............................................21
3.9. DTMF Method...............................................21
3.10. Local and Regional Parameters.............................21
3.11. Time Server...............................................22
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3.12. Language..................................................22
3.13. Inbound Authentication....................................22
3.14. Voice Message Settings....................................23
3.15. Phonebook and Call History................................23
3.16. User Related Settings and Mobility........................23
3.17. AOR Related Settings......................................24
3.18. Maximum Connections.......................................24
3.19. Automatic Configuration and Upgrade.......................25
3.20. Security Configurations...................................25
4. Security Considerations........................................25
4.1. Threats and Problem Statement.............................25
4.2. SIP Telephony Device Security.............................26
4.3. Privacy...................................................27
4.4. Support for NAT and Firewall Traversal....................28
5. IANA Considerations............................................28
6. Acknowledgments................................................29
7. Changes from Previous Versions.................................29
8. References.....................................................31
9. Author's Addresses.............................................36
10. Copyright Notice...............................................36
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.
SIP devices MAY support various other media besides voice, such as
text, video, games and other Internet applications; however the
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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 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.
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.
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.
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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.
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.
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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. 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@ietf.org;user=phone
sips:+12125551234@ietf.org;user=phone
sip:5551234@ietf.org
sips:5551234@ietf.org
tel:5551234;phone-context=l1.ietf.org
sip:5551234;phone-context=l1.ietf.org@ietf.org;user=phone
sips:5551234;phone-context=l1.ietf.org@ietf.org;user=phone
sip:5551234;phone-context=l1.ietf.org@ietf.org;user=dialstring
sips:5551234;phone-context=l1.ietf.org@ietf.org;user=dialstring
tel:5551234;phone-context=+1212
sip:5551234;phone-context=+1212@ietf.org;user=phone
sips:5551234;phone-context=+1212@ietf.org;user=phone
sip:5551234;phone-context=+1212@ietf.org;user=dialstring
sips:5551234;phone-context=+1212@ietf.org;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@ietf.org
sips:5551234@ietf.org
sip:5551234;phone-context=l1.ietf.org@ietf.org;user=dialstring
sips:5551234;phone-context=l1.ietf.org@ietf.org;user=dialstring
sip:5551234;phone-context=+1212@ietf.org;user=dialstring
sips:5551234;phone-context=+1212@ietf.org;user=dialstring
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.
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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].
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"
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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 [47].
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.
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SIP telephony devices can also support for example 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.
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 solution is for
further study at this time. There is also work on the requirements
for location conveyance in the SIPPING WG, see [77].
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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.
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.
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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 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 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.
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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 FXO port to support FXS
devices. If an RJ-11 (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-58: SIP telephony devices MAY provide a 2.5 mm audio port, in
portable SIP devices, such as PDA"s 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.
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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.
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.
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 [24] lists 17 registered MIME subtypes for
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audio codecs. This memo however requires the support of a minimal
number of codecs used in wireline VoIP, and also codecs found in
mobile phones.
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 3G wireless mobile phones. 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 is
needed (for dial-up Internet access) or G.722 for high
quality audio conferences.
Req-75: SIP telephony devices MAY support G.729 and its
annexes.
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 burden of the additional 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 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.
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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.
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. 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) are given in [76].
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 have two types of interface
capabilities, for both phone numbers and URIs, both
accessible to the end user.
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.
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.
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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.
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"
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3.2. Signaling Port
The port that MUST be used for a specific transport protocol for SIP
MUST 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"
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 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
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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 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
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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
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.
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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 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"
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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 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.
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3.14. Voice Message Settings
Various voice message settings require the use of URI'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 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.
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
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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 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].
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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 settings 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].
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
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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:
o Peers can communicate across domains without any pre-arranged
trust relationship,
o There may be many intermediaries in the signaling path,
o Multiple endpoints can be involved in such telephony operations as
forwarding, forking, transfer or conferencing,
o There are seemingly conflicting service requirements when
supporting anonymity, legal intercept, call trace and privacy,
o 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.
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
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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 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.
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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 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.
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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.
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.
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, Gunnar Hellstrom, 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-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.
o Edited and added example for a dial plan in Req-17,
o 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.
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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
o Added edits on text over IP has suggested by Gunnar Hellstrom and
Jon Peterson.
Changes from draft-sinnreich-sipdev-req-04
o 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].
o Removed the reference to RFC 791, since that is implied by
referencing the DiffServ code points in RFC 2597 [22].
o Reviewed and tightened the language based on comments by John
Elwell.
Changes from draft-sinnreich-sipdev-req-03
o Version 03 of the memo is focused more narrowly on SIP telephony
device requirements and configuration only.
o Automatic configuration over the network has been ommitted since
it is addressed separately in [60].
o The section with the example with XML based configuration data
has been omitted, since such data formats are different topic
altogether.
o 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
o Re-edited the section on Interactive text support for hearing or
speech disabled users.
o Shortened the sections on phonebook, call history and line
related settings.
o Deleted the section on ringer behavior.
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o 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] R. Droms:: "Dynamic Host Configuration Protocol", RFC 2131.
IETF, March 1997.
[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.
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[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.
[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 2611. 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-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.
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[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] 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.
[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 Message Specification" RFC 2633.
IETF, June 1999.
[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.
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[44] J. Peterson: "Telephone Number Mapping (ENUM) Service
Registration for Presence Services". RFC 3953. IETF, January 2005.
[45] S. Olson and O. Levin: "REFER extensions",draft-olson-
sipping-refer-extensions-02,IETF July 2004.
[46] A. Johnston: "SIP Service Examples", draft-ietf-
sipping-service-examples-07, IETF July 2004. 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-02.txt, IETF, June 2004.
[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-04,
IETF, October 2004.
[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. 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-05, October 2004.
[56] G. Hellstrom and P. Jones: "RTP Payload for Text
Conversation", draft-ietf-avt-rfc2793bis-09.txt, IETF, August 2004,
work in progress.
[57] A. Johnston: "A Performance Report Event Package For SIP",
draft-johnston-sipping-rtcp-summary-04, IETF, October 2004. Work in
progress.
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[58] C. Jennings: "NAT Classification Results using STUN",
draft-jennings-midcom-stun-results-02, IETF, October 2004. 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-05.txt, IETF, October 2004.
[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-06.txt, IETF, October. 2004, 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-01, IETF, February 2004.
[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 AES Requirements for SIP" draft-
ietf-sip-smime-aes, IETF, June 2003.
[68] J. Peterson and C. Jennings: "Enhancements for Authenticated
Identity Management in the Session Initiation Protocol (SIP)",
draft-ietf-sip-identity, May 2004.
[69] J. Peterson and C. Jennings: "Certificate Management Services
for SIP", draft-sipping-certs, October 2004.
[70] G. Hellstrom and P. Jones: "RTP Payload for Text
Conversation", RFC 2793bis. Internet Draft. Work in progress.
draft-ietf-avt-rfc2793bis-09.txt, IETF, August 2004.
[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.
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[73] F. Audet and C. Jennings: "NAT Behavioral Requirements for
Unicast UDP". IETF, January 2005, work in progress.
[74] R. Mahy: "Connection Reuse in the Session Initiation Protocol
(SIP)". IETF, October 2004. Work in progress.
[75] C. Boulton et al: "Guidelines for implementers using
connection-oriented transports in the Session Initiation Protocol
(SIP)". IETF, February 2005. Work in progress.
[76] J. Rosenberg: "Interactive Connectivity Establishment (ICE): A
Methodology for Network Address Translator (NAT) Traversal for
Multimedia Session Establishment Protocols". Internet Draft, IETF,
October 2004. Work in progress.
[77] J. Polk: "Requirements for Session Initiation Protocol Location
Conveyance". 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
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
Sinnreich, et al Expires December 11, 2005 [Page 36]
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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
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Sinnreich, et al Expires December 11, 2005 [Page 37]
