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 SIP/IMS Standardization
> IETF Standardization Process
> RFCs related to SIP (4 p.) o
> SIP-SIPPING-SIMPLE... I-Ds (22 p.) o
> Audio-Video Transport RFCs (2 p.)
> 3GPP Specifications (12 p.)
> OMA Specifications related to SIP
> TISPAN NGN Specifications (3 p.) o
> SIP Topics
> IMS Topics
 SIP/IMS Call Flows
> RFC3261's Example
> Basic -- RFC3665
> SIP PSTN -- RFC3666 (3 p.)
> SIP Service Examples (19 p.)
> IMS Signaling Flows (35 p.)
 SIP/IMS Architecture
> SIP Protocol Structure
> Dialogs & Routing
> UMTS Network Evolution
 Security
> PKIX-TLS-SMIME... Standards (20 p.) o
> Cryptography Basics
> ASN.1 for PKI Certificate & CRL Profile
> ASN.1 for CMS
> RFC3280's Certificate Examples (4)
> RFC4134's CMS-S/MIME Examples (14)
> RFC4474's SIP Authentication Service
> SSL/TLS Time-Diagrams
> IPSec Guides
 ABNF Grammars
> ABNF Notation & Rules
> URI Generic Syntax
> ABNF for SIP
> SIP Messages & URIs
> SIP Header Fields
> MIME Media Types
> ABNF for SDP
> ABNF for MSRP
> ABNF for MRCPv2
> ABNF for RTSP 2.0
> Internet Message Format
 DiffServ CoS Simulation
> IPVCoSS Simulator
> IP-VPN Case Study
  o (daily updated)
Real-time Applications & Infrastructure (RAI) area
Top I-D List RFC List Charter  
p. 1 IDs in RFC Ed Queue IDs Processed by IESG IETF: ID Exists Individual: ID Exists
## SIPwg ## SIPPINGwg ## SIMPLEwg ## P2PSIPwg ## BLISSwg ## MMUSICwg ## AVTwg ## MEDIACTRLwg ## IPTELwg
## ENUMwg ## SIGTRANwg ## XCONwg ## GEOPRIVwg ## ECRITwg ## SPEECHSCwg ## SPEERMINTwg ## DRINKSwg
## Miscellaneous

Active Individual Drafts

SIPPING working group

andreasen-sipping-
rfc3603bis-05

ID Exists
Feb 21, 2008
(39 p.)
[pdf(2)] [html]
F. Andreasen
B. McKibben
W. Marshall
Private SIP Proxy-to-Proxy Extensions for Supporting the PacketCable Distributed Call Signaling Architecture
In order to deploy a residential telephone service at very large scale across different domains, it is necessary for trusted elements owned by different service providers to exchange trusted information that conveys customer-specific information and expectations about the parties involved in the call. This document describes private extensions to the Session Initiation Protocol (SIP) [RFC3261] for supporting the exchange of customer information and billing information between trusted entities in the PacketCable Distributed Call Signaling Architecture. These extensions provide mechanisms for access network coordination to prevent theft of service, customer originated trace of harassing calls, support for operator services and emergency services, and support for various other regulatory issues. The use of the extensions is only applicable within closed administrative domains, or among federations of administrative domains with previously agreed-upon policies where coordination of charging and other functions is required.
Up  List Intended Status:Informational
audet-sipping-
feature-ref-00

ID Exists
Feb 17, 2008
(20 p.)
[pdf(2)] [html]
F. Audet
A. Johnston
R. Mahy
C. Jennings
Feature Referral in SIP
Feature referral allows for an application to make a high level request to a User Agent to perform an action or "feature", and let the the User Agent actually execute the feature as it sees fit. Feature referral uses the SIP REFER method with a Refer-To header field containing a URN.
Up  List Intended Status:Best Current Practice
bakker-sipping-
3gpp-ims-xml-
body-handling-00

ID Exists
Feb 12, 2008
(8 p.)
[pdf(2)] [html]
J. Bakker
Specification of 3GPP IM CN Subsystem XML body handling
This document registers new disposition-types for the Content-Disposition header that apply to the application/3gpp-ims+xml body used by 3GPP, since Release 5. The applicability of these content-disposition values are limited to 3GPP IMS. The application/ 3gpp-ims+xml body has the following two distinct uses: (1) for redirecting the emergency session to use a different domain (e.g. using a Circuit Switched call), and (2) for delivering user profile specific information from the SIP registrar to an Application Server.
Up  List Intended Status:Experimental
burger-sipping-
kpml-basic-00

ID Exists
(Recently Expired)

Nov 11, 2007
(14 p.)
[pdf(2)] [html]
E. Burger
Basic Profile of the SIP Event Package for Key Press Stimulus (KPML)
This document describes a profile of the SIP Event Package "kpml" that enables simple monitoring of DTMF signals and uses XML documents referred to as Key Press Markup Language (KPML). This profile assumes devices of lesser capability that may have trouble parsing subscription requests or producing generic XML documents. This is a profile of RFC 4730, KPML.
Up  List Intended Status:Standards Track
chadli-sipping-
overload-sub-not-01

ID Exists
Jul 11, 2008
(12 p.)
[pdf(2)] [html]
Y. Chadli
X. Marjou
An overload control package for SIP
This document specifies an event package for the notification of overload control using the Session Initiation Protocol (SIP) events framework. The overload control package allows an upstream server to retrieve overload control information from a downstream server and to be notified when this information changes. This information is used by the upstream server to adapt its flow toward the downstream server and thus to avoid overloading it.
Up  List Intended Status:Standards Track
channabasappa-
sipping-app-
profile-type-02

ID Exists
Feb 21, 2008
(11 p.)
[pdf(2)] [html]
S.Channabasappa
J. Littlefield
S. Loreto
Extension to the ua-profile Event Package to Support the Application Profile Type
The Framework for Session Initiation Protocol User Agent Profile Delivery specifies an event package (ua-profile) that can be used by user agents to retrieve profile data. The framework also allows for optional notification of changes to the retrieved profiles. Three profile types are specified: local-network, device, and user. This document extends that event package to support an additional profile type, application. This would enable User Agents to retrieve profile data specific to one or more applications.
Up  List Intended Status:Standards Track
choi-sipping-
experiments-spit-01

ID Exists
(Recently Expired)

Nov 16, 2007
(14 p.)
[pdf(2)] [html]
Jaeduck Choi
Souhwan Jung
Yujung Jang
Yoojae Won
Youngduk Cho
Experiments on SPIT in the Commercial VoIP Services
This document shows some experimental results on SPIT on commercial VoIP services, in which a SIP UA has not been secured by SIP security protocol such as TLS. Although many service providers have been applying the HTTP digest scheme to authenticate a SIP UA, they often do not apply SIP signaling protection against potential threats between the SIP UA and the SIP proxy. This cause vulnerabilities to the VoIP services like SPIT. The aim of this memo is to inform the service providers of SPIT threats by showing some experimental results of SPIT on current VoIP networks.
Up  List Intended Status:Informational
dawes-sipping-
debug-event-00

ID Exists
May 3, 2008
(31 p.)
[pdf(2)] [html]
P. Dawes
A SIP Event Package for Debugging
This document defines a Session Initiation Protocol (SIP) event package for debugging. SIP requests and responses for session setup can traverse a number of network entities, which might be geographically spread over a large area. This document provides a means by which logging of requests and responses can be configured on a per-entity basis.
Up  List Intended Status:Informational
dawes-sipping-
debug-id-00

ID Exists
May 3, 2008
(13 p.)
[pdf(2)] [html]
P. Dawes
Private Extension to SIP for Debugging
Networks that use SIP to start and stop sessions between their users will frequently be upgraded with software and hardware changes. Users will similarly frequently change their client software and the way they user the network. In order to provide troubleshooting and regression testing, it is useful to provide debugging as part of the network fabric. This draft describes a SIP private header that triggers logging of SIP signalling and identifies logs at mulitiple SIP entities as belonging to a single end-to-end session.
Up  List Intended Status:Standards Track
drage-sipping-
rfc3455bis-01

ID Exists
Jul 14, 2008
(46 p.)
[pdf(2)] [html]
K. Drage
Private Header (P-Header) Extensions to SIP for the 3rd-Generation Partnership Project (3GPP)
This document describes a set of private Session Initiation Protocol (SIP) headers (P-headers) used by the 3rd-Generation Partnership Project (3GPP), along with their applicability, which is limited to particular environments. The P-headers are for a variety of purposes within the networks that the partners use, including charging and information about the networks a call traverses.
Up  List Intended Status:Informational
froment-sipping-
spit-requirements-03

ID Exists
Jul 14, 2008
(15 p.)
[pdf(2)] [html]
H. Tschofenig
G. Dawirs
T. Froment
D. Wing
H. Schulzrinne
Requirements for Authorization Policies to tackle Spam and Unwanted Communication for Internet Telephony
Spam over Internet Telephony (SPIT) is one of the foreseen future forms of spamming that SIP open-wide networks may have to handle. SPIT also has more impact on users than email spam since it is more intrusive. Email as a store-and-forward communication mechanism allows for several filtering mechanisms to be applied to the full content before being presented to the user. Session Initiation Protocol (SIP) interaction is, in contrast, real-time communication and therefore does not provide much information prior to the transmission of the content, making it both harder to filter and more annoying to users. The responsibility for filtering, blocking calls, or taking any other preventive action can belong to different elements in the call flow and may depend on various factors. This document discusses the requirements to define authorization policies that should allow end users or other parties to setup anti-SPIT policies for triggering these actions. These policies typically match a particular SIP communication pattern based on a number of attributes. The range of attributes includes information provided, for example, by the SIP protocol itself, by the SIP identity mechanism, by information carried within SAML assertions or by reputation systems of social networks.
Up  List Intended Status:Informational
garcia-sipping-
file-desc-pidf-01

ID Exists
(Recently Expired)

Nov 16, 2007
(8 p.)
[pdf(2)] [html]
M. Garcia-Martin
M. Matuszewski
File Descriptions Extension to the Presence Information Data Format (PIDF)
The Presence Information Data Format (PIDF) defines a basic format for representing presence information for a presentity. Presentities publish their presence information, typically towards presence agents. PIDF has been extended to provide rich presence information, including, for example, the location of the presentity, their activities, mood, the capabilities of their user agents, etc. Presentities are willing to provide the description of available files at watcher's disposal. This might be the case for photographs taken with a mobile device, a recorded lecture audio file, etc. This document extends the PIDF to provide the syntax and format for the description of files within the PIDF.
Up  List Intended Status:Standards Track
garcia-sipping-
file-event-
package-01

ID Exists
(Recently Expired)

Nov 16, 2007
(15 p.)
[pdf(2)] [html]
M. Garcia-Martin
M. Matuszewski
A SIP Event Package and Data Format for Describing Files
This document specifies the format and semantics associated to a 'file' event package that allows SIP endpoints to publish the availability of files. A file can be, for example, images, video files, audio files, etc. The event package reuses the eXtended Mark-up Language (XML) 'file-metadata' document to provide file descriptions. This event package also allows SIP endpoints to subscribe to changes in the availability of files, or perform searches for the availability and location of a given file. Support for partial notifications and publications is also accomplished by using XML patch operations.
Up  List Intended Status:Standards Track
garcia-sipping-
file-sharing-
framework-01

ID Exists
(Recently Expired)

Nov 16, 2007
(20 p.)
[pdf(2)] [html]
M. Garcia-Martin
M. Matuszewski
N. Beijar
J. Lehtinen
Sharing Files with SIP
This memo proposes a SIP framework used for advertising and searching for shared files within a given community. The memo defines the signaling that users to announce the availability of files stored in their User Agents (UA). It also provides the signaling for users to perform searches of available files and monitor changes in those files. Additionally, this memo describes the signaling used to access a file. These methods can be used in (but are not limited to) SIP peer-to-peer systems based on centralized, semi-centralized or fully distributed architectures.
Up  List Intended Status:Standards Track
haluska-sipping-
directory-
assistance-05

ID Exists
May 8, 2008
(91 p.)
[pdf(2)] [html]
J. Haluska
R. Berkowitz
P. Roder
W. Downum
R. Ahern
P. Lum Lung
N. S. Costantino
C. Blackwell
J. Mellinger
D. Scott
Considerations for Information Services and Operator Services using SIP
Information Services are services whereby information is provided in response to user requests, and may include involvement of a human or automated agent. A popular existing Information Service is Directory Assistance (DA). Moving ahead, Information Services providers envision exciting multimedia services that support simultaneous voice and data interactions with full operator backup at any time during the call. Information Services providers are planning to migrate to SIP based platforms, which will enable such advanced services, while continuing to support traditional DA services. Operator Services are traditional PSTN services which often involve providing human or automated assistance to a caller, and often require the specialized capabilities traditionally provided by an operator services switch. Market and/or regulatory factors in some jurisdictions dictate that some subset of Operator Services continue to be provided going forward. This document aims to identify how Operator and Information Services can be implemented using existing or currently proposed SIP mechanisms, to identity existing protocol gaps, and to provide a set of Best Current Practices to facilitate interoperability. For Operator Services, the intention is to reproduce the current PSTN behaviour.
Up  List Intended Status:Informational
hilt-sipping-
overload-05

ID Exists
Jul 13, 2008
(21 p.)
[pdf(2)] [html]
V. Hilt
I. Widjaja
H. Schulzrinne
SIP Overload Control
Overload occurs in Session Initiation Protocol (SIP) networks when SIP servers have insufficient resources to handle all SIP messages they receive. Even though the SIP protocol provides a limited overload control mechanism through its 503 (Service Unavailable) response code, SIP servers are still vulnerable to overload. This document defines an overload control mechanism for SIP.
Up  List Intended Status:Standards Track
hilt-sipping-
overload-design-00

ID Exists
Jul 5, 2008
(17 p.)
[pdf(2)] [html]
V. Hilt
Design Considerations for SIP Overload Control
Overload occurs in Session Initiation Protocol (SIP) networks when SIP servers have insufficient resources to handle all SIP messages they receive. Even though the SIP protocol provides a limited overload control mechanism through its 503 (Service Unavailable) response code, SIP servers are still vulnerable to overload. This document discusses models and design considerations for a SIP overload control mechanism.
Up  List Intended Status:Informational
izumikawa-sipping-
sipbicast-01

ID Exists
Feb 25, 2008
(23 p.)
[pdf(2)] [html]
H. Izumikawa
R. Lillie
SIP-based Bicasting for Seamless Handover between Heterogeneous Networks
A vertical session handoff occurs in heterogeneous networks when a session media is moved between access networks within the same device. During session handoff, bi-casting media streams to both access networks of the session's device may be desirable to insure a seamless session handoff. This document describes the general methods and specifies the signaling and media flows for providing this service using SIP.
Up  List Intended Status:Informational
jesske-sipping-
etsi-ngn-reason-03

ID Exists
Feb 19, 2008
(6 p.)
[pdf(2)] [html]
R. Jesske
Use of the Reason header field in SIP responses
This document proposes the use of the Reason header field in SIP responses.
Up  List Intended Status:Informational
johnston-sipping-
cc-uui-04

ID Exists
Jul 10, 2008
(14 p.)
[pdf(2)] [html]
A. Johnston
J. McMillen
Transporting User to User Call Control Information in SIP for ISDN
Several approaches to transporting the ITU-T Q.931 User to User Information Element (UU IE) data in SIP have been implemented. As networks move to SIP it is important that applications requiring this data can continue to function in SIP networks as well as the ability to interwork the information to/from ISDN for end-to-end transparency. This document discusses the approaches and recommends a new header field User-to-User be standardized. Example use cases include an exchange between two User Agents, insertion of UUI by a proxy, modification of UUI by a proxy. An example application is in an Automatic Call Distributor (ACD) in a contact center.
Up  List Intended Status:Standards Track
kaplan-sipping-
dtmf-package-00

ID Exists
(Recently Expired)

Nov 12, 2007
(8 p.)
[pdf(2)] [html]
H. Kaplan
C. Holmberg
DTMF Info-Event Package
This document defines a specific SIP Info-event package, based on the generic framework of draft-kaplan-sip-info-events-00.txt, for the purpose of exchanging DTMF events.
Up  List Intended Status:Standards Track
koren-sipping-
conference-
list-event-03

ID Exists
Jul 2, 2008
(17 p.)
[pdf(2)] [html]
O. Koren
M. Fortinsky
I. Ravid
A Conference List Information Event Package for SIP
This document describes the usage of the Session Initiation Protocol (SIP) for subscriptions and notifications related to conference lists. A new conference list event package is specified. This event package allows a user to subscribe to a single event (the conference-list) and receive notifications that contain the list of conferences to which the user belongs and the status of each conference. The notifications sent from the conference server can contain either the entire list of the user's conferences or a partial list with the updates since the previous notification.
Up  List Intended Status:Standards Track
lamparter-sipping-
session-
duplication-00

ID Exists
(Recently Expired)

Dec 19, 2007
(16 p.)
[pdf(2)] [html]
P. Imai
B. Lamparter
P. Loureiro
Duplicate a SIP session
Session duplication copies media of an ongoing communication session from one device to another. After duplication the two devices have independent control over this media. This is in particular useful for one-way media like broadcast IPTV or Video-on-Demand. This document describes the general methods and specifies the signaling and media flows for providing this service using the Session Initiation Protocol (SIP).
Up  List Intended Status:Standards Track
loreto-sipping-
3892bis-00

ID Exists
Jul 4, 2008
(7 p.)
[pdf(2)] [html]
N. Bishai
S. Loreto
A. Haruna
Updates to Referred-By in SIP
SIP has a mechanism for conveying the asserted identity of the originator of a request by means of the P-Asserted-Identity header field. When exploding a SIP MESSAGE request to a pre-defined group URI and when exploding a SIP INVITE request to an ad-hoc group or to a pre-defined group URI, the Referred-By header field in the resulting exploded requests is set to the P-Asserted-Identity header field or to the From header field. The Referred-By header is only included if the P-Asserted-Identity header field or From header field needs to carry another value, e.g. the URI of a pre-defined group, or a conference focus URI. Since the P-Asserted-Identity header field may carry up to two values, the Referred-By definition needs to be extended to allow up to two values as well.
Up  List Intended Status:Standards Track
loreto-sipping-
context-id-
requirements-00

ID Exists
Jun 16, 2008
(9 p.)
[pdf(2)] [html]
G. Camarillo
S. Loreto
Requirements for correlation of multiple SIP sessions for user to user communications
This document shows the need and list the requirements to logically correlate an existing SIP dialog, or even multi SIP transactions sent outside of a dialog, with a new SIP dialog. This mechanism can be used to implement applications using multiple SIP sessions but also in peer-to-peer call control environments.
Up  List Intended Status:Standards Track
lowekamp-sipping-
ice-for-sip-00

ID Exists
(Recently Expired)

Nov 11, 2007
(11 p.)
[pdf(2)] [html]
B. Lowekamp
D. Bryan
Using ICE to establish SIP Dialogs
This draft explores a way SIP can be extended to allow a new dialog directly between the endpoints to replace an initial dialog that had one or more proxies in the signalling path. This technique relies on ICE to perform hole punching that allows a direct connection to be used in deployments where a sip-outbound proxy or SBC is used to establish SIP connections across NAT or firewall boundaries. It can also be used to replace such a dialog with a secure connection directly between the endpoints. This technique can be applied to traditional proxy-based SIP routing as well as to emerging P2PSIP deployments that lack centrally located proxies.
Up  List Intended Status:Standards Track
moran-sipping-
overload-retry-00

ID Exists
Mar 14, 2008
(13 p.)
[pdf(2)] [html]
T. Moran
Determining When to Discard or Retry a SIP Request During Overload
SIP servers can become overload and unable to handle all SIP messages they receive. Even though the SIP protocol provides a limited overload control mechanism through its 503 (Service Unavailable) response code, SIP servers are still vulnerable to overload. This document proposes an architecture for determining when a client should throttle the sending of SIP requests vs. retrying them.
Up  List Intended Status:-
niccolini-sipping-
siphandover-04

ID Exists
Jul 8, 2008
(12 p.)
[pdf(2)] [html]
S. Niccolini
S. Salsano
H. Izumikawa
R. Lillie
L. Veltri
Y. Kishi
Requirements for vertical handover of multimedia sessions using SIP
A vertical handover occurs in heterogeneous networks when a session media is moved among different access network technologies within the same device. This document analyses the issue of handling the vertical handover using SIP.
Up  List Intended Status:Informational
niccolini-sipping-
spam-feedback-00

ID Exists
Feb 14, 2008
(12 p.)
[pdf(2)] [html]
S. Niccolini
K. Fischer
D. Wing
M. Stiemerling
H. Tschofenig
Spam feedback for SIP
This document gives on overview of possible mechanisms for SIP UAs to feedback spam information to the system (e.g. other SIP entities like upstream SIP proxies) thus they can use this information for handling subsequent calls (e.g. blacklist the caller, input this info to reputation systems, compute spam-specific caller statistics, etc.).
Up  List Intended Status:Standards Track
niccolini-sipping-
spitstop-01

ID Exists
Feb 15, 2008
(18 p.)
[pdf(2)] [html]
S. Niccolini
J. Quittek
J. Seedorf
Signaling TO Prevent SPIT (SPITSTOP) Reference Scenario
This memo discusses the need for standards that support SPam over Internet Telephony (SPIT) prevention applications. After explaining the general need for SPIT prevention applications the memo provides a reference scenario for potential communication between entities that may be involved in SPIT prevention. Within this scenario the need for standardizing communication is analyzed for each pair of communicating entities. The analysis is intended to serve as a starting point for discussing the requirements for signaling standards as well as for architectural considerations, that support SPIT prevention applications.

This memo is not intended to suggest any solution of SPIT signaling issues. Such work, if necessary at all, should be object of separate documents.
Up  List Intended Status:Informational
petrie-sipping-
identity-dataset-02

ID Exists
(Recently Expired)

Nov 17, 2007
(17 p.)
[pdf(2)] [html]
D. Petrie
S.Channabasappa
S. Ganesan
The SIP User Agent Identity Profile Data Set
This document defines the properties and data format for the SIP user agent identity profile data set. The properties in this data set define identities or SIP address of records (AORs) related to incoming or outgoing SIP signaling on a user agent. The identities may be those that are registered via the SIP REGISTER method or identities which are provisioned on the user agent. These identities may be used or referenced when defining identity specific handling related to SIP features on the user agent.
Up  List Intended Status:Standards Track
petrie-sipping-
sip-dataset-03

ID Exists
(Recently Expired)

Nov 17, 2007
(22 p.)
[pdf(2)] [html]
D. Petrie
S.Channabasappa
S. Ganesan
The Core SIP User Agent Protocol Data Set
This document defines the properties and format for the core SIP user agent protocol dataset. The properties defined in this document are expected to be common to most SIP user agents regardless of whether the user agent support audio, video, text or any combination of media. These core SIP properties are considered to be a dataset. Several datasets may be combined into documents or profiles that are provided to SIP user agents so that they can operate with the desired behavior.
Up  List Intended Status:Standards Track
petrie-sipping-
voip-features-
dataset-02

ID Exists
(Recently Expired)

Nov 17, 2007
(13 p.)
[pdf(2)] [html]
D. Petrie
S.Channabasappa
S. Ganesan
The SIP User Agent VoIP Features Data Set
This document defines the properties and format for the SIP user agent VoIP Features data set. The properties defined in this document are expected to be common to most SIP user agents that provide VoIP capabilities. These VoIP Feature properties are considered to be a data set. Several types of datasets may be combined into documents that are provided to SIP user agents so that they can operate with the desired behavior.
Up  List Intended Status:Best Current Practice
rosen-sipping-
cap-02

ID Exists
Jul 12, 2008
(12 p.)
[pdf(2)] [html]
B. Rosen
H. Schulzrinne
H. Tschofenig
SIP Event Package for the Common Alerting Protocol (CAP)
The Common Alerting Protocol (CAP) is an XML document format for exchanging emergency alerts and public warnings. This document allows CAP documents to be distributed via the event notification mechanism available with the Session Initiation Protocol (SIP).
Up  List Intended Status:Standards Track
rosenberg-sipping-
siptrunk-00

ID Exists
Feb 15, 2008
(10 p.)
[pdf(2)] [html]
J. Rosenberg
What is a Session Initiation Protocol (SIP) Trunk Anyway?
The term "Session Initiation Protocol (SIP) Trunk" has become almost commonplace amongst vendors and SIP providers. Even though the notion of a 'trunk' has a well defined meaning in circuit switched systems, it has never been defined for SIP. This document provides a formal definition for a SIP trunk, discusses its scope and applications, and establishes best practices for identification and security of SIP trunks.
Up  List Intended Status:Best Current Practice
salsano-sipping-
siphandover-
solution-02

ID Exists
May 7, 2008
(24 p.)
[pdf(2)] [html]
S. Salsano
S. Niccolini
L. Veltri
A. Polidoro
A solution for vertical handover of multimedia sessions using SIP
This document proposes a solution for handling vertical handovers among different network technologies using SIP, fulfilling a set of requirements discussed in the document "Requirements for vertical handover of multimedia sessions using SIP" (draft-niccolini-sipping-siphandover-03). The solution requires a new header field (named "Handover") and a new parameter in the Via header field (named "MMID").
Up  List Intended Status:Informational
schwartz-sipping-
domain-marking-
requirements-00

ID Exists
(Recently Expired)

Nov 12, 2007
(6 p.)
[pdf(2)] [html]
D. Schwartz
D. Besprosvan
Requirements for domain marking for the purpose of Upstream Traffic Characterization
SIP as defined in RFC 3261 defines a Via header as a construct to be used for upstream response routing and for downstream assistance in loop detection. There is an increasing need on downstream administrative domains (ADs) to gain visibility into all the ADs in its upstream path. The information needed is not IP based as internal architectures at upstream ADs is of no consequence to downstream ADs. Logical domain marking, however, is desperately needed for any traffic analysis to occur at the receiving side. Gathering AD information from Via headers is non obvious and in many instances nearly impossible. This documents identifies the requirements for addressing this issue.
Up  List Intended Status:Informational
schwartz-sipping-
nsr-code-00

ID Exists
Jun 24, 2008
(7 p.)
[pdf(2)] [html]
D. Schwartz
No Service To This Number Reject Code
The Session Initiation Protocol (SIP) allows for users to make calls using telephone numbers embedded in either "sip" [RFC3261] or "tel" [RFC3966] URIs. Either way, the telephone number (TN) itself is not restricted and can represent an entity in the public telephone network, an entity in a private telephone network or an entity on the Internet. This TN resolution ambiguity highlights the difference between the LUF and LRF functions defined in [draft-ietf-speermint-terminology] and underscores the need for more precise SIP error rejection codes. SIP has no way to indicate to the calling UAC that the reason for call rejection is due to the fact that this number does not exist in the requested domain or realm but it does exist in the public telephone network for instance. Such an indication is useful to allow the call to be retried at a different context (i.e. the public PSTN in this case) with possibly better results. This specification defines a new SIP response code for this purpose.
Up  List Intended Status:Standards Track
seedorf-sipping-
spam-score-
semantics-00

ID Exists
Feb 18, 2008
(8 p.)
[pdf(2)] [html]
J. Seedorf
S. Niccolini
H. Schulzrinne
Spam score for SIP: a proposal for semantics
This document reports a proposal for semantics of SIP spam scoring in order to achieve a flexible signalling standardization allowing an incremental adoption of the scoring mechanism. This approach can give early experimental implementers the possibility to start using spam scoring extensions in an explorative fashion without running into interoperability problems.
Up  List Intended Status:Informational
shen-sipping-
load-control-
event-package-00

ID Exists
Jul 7, 2008
(22 p.)
[pdf(2)] [html]
C. Shen
H. Schulzrinne
A. Koike
A SIP Load Control Event Package
This document defines a load control event package for the Session Initiation Protocol (SIP). It allows SIP servers to distribute user load control information to SIP servers. The load control information can throttle outbound calls based on their destination domain, telephone number prefix or for a specific user. The mechanism helps to prevent signaling overload and complements feedback-based SIP overload control efforts.
Up  List Intended Status:Standards Track
sun-sipping-
multiple-reply-01

ID Exists
(Recently Expired)

Oct 12, 2007
(18 p.)
[pdf(2)] [html]
Q. Sun
L. Tian
D. Ren
Multiple Reply to MESSAGE requests in SIP
This document defines a multiple target address extension to the Reply-To header field for the SIP MESSAGE method. The extension includes the use of a pointer to a Uniform Resource Identifier (URI)-list in the Reply-To header field.
Up  List Intended Status:-
tschofenig-sipping-
captcha-01

ID Exists
Feb 25, 2008
(24 p.)
[pdf(2)] [html]
H. Tschofenig
E. Leppanen
S. Niccolini
M. Arumaithurai
Completely Automated Public Turing Test to Tell Computers and Humans Apart (CAPTCHA) based Robot Challenges for SIP
A common approach to deal with unwanted communication attempts is to rely on some form of authorization policies, typically whitelists. In order to populate the entries in such an access control list it is helpful to have a way to challenge the entity willing to engage in a conversation (unless they are already pre-authorized). One reason why this is desired is to deal with robots that are aggressively distributing messages.

This document describes how "Completely Automated Public Turing Test to Tell Computers and Humans Apart" (CAPTCHA) tests, which require human interaction, are applied to SIP.
Up  List Intended Status:Standards Track
tschofenig-sipping-
framework-spit-
reduction-04

ID Exists
Jul 14, 2008
(24 p.)
[pdf(2)] [html]
H. Tschofenig
H. Schulzrinne
D. Wing
J. Rosenberg
D. Schwartz
A Framework to tackle Spam and Unwanted Communication for Internet Telephony
Spam, defined as sending unsolicited messages to someone in bulk, is likely to become a problem on SIP open-wide deployed networks. A number of solutions have been proposed for dealing with Spam for Internet Telephony (SPIT) and unwanted communication, such as content filtering, black lists, white lists, consent-based communication, reputation systems, address obfuscation, limited use addresses, turing tests, computational puzzles, payments at risk, circles of trust, and many others.

This document describes the big picture that illustrates how the different building blocks fit together and can be deployed incrementally.
Up  List Intended Status:Informational
tschofenig-sipping-
spit-policy-03

ID Exists
Jul 12, 2008
(27 p.)
[pdf(2)] [html]
H. Tschofenig
D. Wing
H. Schulzrinne
T. Froment
G. Dawirs
A Document Format for Expressing Authorization Policies to tackle Spam and Unwanted Communication for Internet Telephony
SPAM, defined as sending unsolicited messages to someone in bulk, might be a problem on SIP open-wide deployed networks. The responsibility for filtering or blocking calls can belong to different elements in the call flow and may depend on various factors. This document defines an authorization based policy language that allows end users to upload anti-SPIT policies to intermediaries, such as SIP proxies. These policies mitigate unwanted SIP communications. It extends the Common Policy authorization framework with additional conditions and actions. The new conditions match a particular Session Initiation Protocol (SIP) communication pattern based on a number of attributes. The range of attributes includes information provided, for example, by SIP itself, by the SIP identity mechanism, by information carried within SAML assertions.
Up  List Intended Status:Standards Track
vakil-sipping-
notify-pause-02

ID Exists
Mar 12, 2008
(15 p.)
[pdf(2)] [html]
M. Vakil
An Extension to Session Initiation Protocol (SIP) Events for Pausing and Resuming Notifications
The Session Initiation Protocol (SIP) events framework enables a subscriber to receive asynchronous notification of various events from other SIP user agents. It defines mechanisms to create, refresh, terminate subscriptions. This framework also defines mechanism to fetch (poll) an event state of a resource without creating persistent subscriptions. There is no mechanism to temporarily pause the notifications, while still maintaining a subscription on the server. This lack of functionality sometime results in a lot of superfluous notification traffic, and put unnecessary load on the server. This draft defines an extension to SIP events that allows the subscriber to pause, un-pause notifications, and be able to perform fetch (poll) subscriptions within an established (created) subscription dialog.
Up  List Intended Status:Standards Track
vanelburg-sipping-
private-network-
indication-02

ID Exists
Jul 11, 2008
(18 p.)
[pdf(2)] [html]
J. van Elburg
K. Drage
The SIP P-Private-Network-Indication Private-Header (P-Header)
This document describes why a private network indication is needed. A private network indication allows other nodes in a network to treat private network traffic to a different set of rules then public network traffic. The indication also distinguishes one private network from another private network.
Up  List Intended Status:Informational
wing-sipping-
spam-score-02

ID Exists
Feb 23, 2008
(12 p.)
[pdf(2)] [html]
D. Wing
S. Niccolini
M. Stiemerling
H. Tschofenig
Spam Score for SIP
This document defines a mechanism for SIP proxies to communicate a spam score to downstream SIP proxies and to SIP user agents. This information can then be used as input to other decision making engines, for example, to provide alternate call routing or call handling.
Up  List Intended Status:Experimental
wing-sipping-
srtp-key-03

ID Exists
Feb 24, 2008
(25 p.)
[pdf(2)] [html]
D. Wing
F. Audet
S. Fries
H. Tschofenig
A. Johnston
Secure Media Recording and Transcoding with SIP
Call recording is an important feature in enterprise telephony applications. Some industries such as financial traders have requirements to record all calls in which customers give trading orders. This poses a particular problem for Secure RTP systems as many SRTP key exchange mechanisms do not disclose the SRTP session keys to intermediate SIP proxies. As a result, these key exchange mechanisms cannot be used in environments where call recording is needed.

This document specifies a secure mechanism for a cooperating endpoint to disclose its SRTP master keys to an authorized party to allow secure call recording.
Up  List Intended Status:Standards Track