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