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RECENT IETF DRAFTS
SIP internet drafts statistics
- 45 SIP related internet drafts (IETF).
- 1 new and updated drafts posted in the last 14 days.
| Title | Author | Date |
| Resolution of The SPF and Sender ID Experiments | Murray Kucherawy | 2012-05-11 |
| In 2006 the IETF published a suite of protocol documents comprising SPF and Sender ID, two proposed email authentication protocols. Both of these protocols enable one to publish via the Domain Name System a policy declaring which mail servers were authorized to send email on behalf of the domain name being queried. There was concern that the two would conflict in some significant operational situations, interfering with message delivery. The IESG required the publication of all of these documents (RFC4405, RFC4406, RFC4407, and RFC4408) with Experimental status, and requested that the community observe deployment and operation of the protocols over a period of two years from the date of publication to determine a reasonable path forward. After six years, sufficient experience and evidence have been collected that the experiments thus created can be considered concluded. This document presents those findings. | ||
| The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP) | Inaki Castillo, Jose Millan, Victor Pascual | 2012-05-05 |
| The WebSocket protocol enables two-way realtime communication between clients and servers. This document specifies a new WebSocket sub- protocol as a reliable transport mechanism between SIP (Session Initiation Protocol) entities and enables usage of the SIP protocol in new scenarios. | ||
| Shared Appearances of a Session Initiation Protocol (SIP) Address of Record (AOR) | Alan Johnston, Mohsen Soroushnejad, Venkatesh Venkataramanan | 2012-05-04 |
| This document describes the requirements and implementation of a group telephony feature commonly known as Bridged Line Appearance (BLA) or Multiple Line Appearance (MLA), or Shared Call/Line Appearance (SCA). When implemented using the Session Initiation Protocol (SIP), it is referred to as shared appearances of an Address of Record (AOR) since SIP does not have the concept of lines. This feature is commonly offered in IP Centrex services and IP-PBX offerings and is likely to be implemented on SIP IP telephones and SIP feature servers used in a business environment. This feature allows several user agents (UAs) to share a common AOR, learn about calls placed and received by other UAs in the group, and pick up or join calls within the group. This document discusses use cases, lists requirements and defines extensions to implement this feature. | ||
| A Mechanism for Transporting User to User Call Control Information in SIP | Alan Johnston, James Rafferty | 2012-05-04 |
| There is a class of applications which benefit from using SIP to exchange User to User Information (UUI) data during session establishment. This information, known as call control UUI data, is a small piece of data inserted by an application initiating the session, and utilized by an application accepting the session. The rules which apply for a specific application are defined by a UUI package. This UUI data is opaque to SIP and its function is unrelated to any basic SIP function. This document defines a new SIP header field, User-to-User, to transport UUI data, along with an extension mechanism. | ||
| Interworking ISDN Call Control User Information with SIP | Keith Drage, Alan Johnston | 2012-05-04 |
| The motivation and use cases for interworking and transporting ITU-T DSS1 User-user information element data in SIP are described in the "Problem Statement and Requirements for Transporting User to User Call Control Information in SIP" document. 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 with this ISDN service for end-to- end transparency. This document defines a usage (a new package) of the User-to-User header field to enable interworking with this ISDN service. This document covers the interworking with both public ISDN and private ISDN capabilities, so the potential interworking with QSIG will also be addressed. The package is identified by a new value "isdn-uui" of the "purpose" header field parameter. | ||
| SIP-Specific Event Notification | Adam Roach | 2012-04-30 |
| This document describes an extension to the Session Initiation Protocol (SIP) defined by RFC 3261. The purpose of this extension is to provide an extensible framework by which SIP nodes can request notification from remote nodes indicating that certain events have occurred. Note that the event notification mechanisms defined herein are NOT intended to be a general-purpose infrastructure for all classes of event subscription and notification. This document represents a backwards-compatible improvement on the original mechanism described by RFC 3265, taking into account several years of implementation experience. Accordingly, this document obsoletes RFC 3265. This document also updates RFC 4660 slightly to accommodate some small changes to the mechanism that were discussed in that document. | ||
| An Extension to the Session Initiation Protocol (SIP) for Request History Information | Mary Barnes, Francois Audet, Shida Schubert, Hans van Elburg, Christer Holmberg | 2012-04-30 |
| This document defines a standard mechanism for capturing the history information associated with a Session Initiation Protocol (SIP) request. This capability enables many enhanced services by providing the information as to how and why a SIP request arrives at a specific application or user. This document defines an optional SIP header field, History-Info, for capturing the history information in requests. The document also defines SIP header field parameters for the History-Info and Contact header fields to tag the method by which the target of a request is determined. In addition, this specification defines a value for the Privacy header field that directs the anonymization of values in the History-Info header field This document obsoletes RFC 4244. | ||
| SIMPLE made Simple: An Overview of the IETF Specifications for Instant Messaging and Presence using the Session Initiation Protocol (SIP) | Jonathan Rosenberg | 2012-04-18 |
| The IETF has produced many specifications related to Presence and Instant Messaging with the Session Initiation Protocol (SIP). Collectively, these specifications are known as SIMPLE - SIP for Instant Messaging and Presence Leveraging Extensions. This document serves as a guide to the SIMPLE suite of specifications. It breaks them up into categories and explains what each is for and how they relate to each other. | ||
| A Session Initiation Protocol (SIP) INFO package for Private Wire | Richard Beauchamp, Finlay Fraser, Chris Boulton | 2012-04-17 |
| Application level data exchanged using the SIP INFO method are supported and documented in specifications known as \'INFO Packages\'. This document defines functionality associated with Session Initiation Protocol (SIP) Private Wire functionality and creates an \'INFO Package\' for carrying such application level data. | ||
| The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP) | Inaki Castillo, Jose Millan, Victor Pascual | 2012-04-16 |
| The WebSocket protocol enables two-way realtime communication between clients and servers. This document specifies a new WebSocket sub- protocol as a reliable transport mechanism between SIP (Session Initiation Protocol) entities and enables usage of the SIP protocol in new scenarios. | ||
| Session Initiation Protocol (SIP) Extension for logging and debugging | Adarsh Kaithal | 2012-04-08 |
| The current mechanisms to debug issues in SIP network are not very efficient. It requires to enable debugging logs across different devices, recreate the problem and then collect the logs. The idea is to provide a solution to automatically enable relevant logs (SIP messages and any other debugging logs meaningful to SIP devices), and also to indicate where the logs are to be collected or stored. The enabling of logs will happen at all the SIP devices(upstream or downstream). This will help to get the logs from all the SIP devices in a Common logging format (CLF). The solution extends SIP to provide the infrastructure to enable logging for upstream and downstream devices with each server deciding how much troubleshooting information it wants to log - with freedom to simply ignore requests if required. This document specifies a new header called "Log-Me" Header in all the SIP messages. | ||
| Alert-Info URNs for the Session Initiation Protocol (SIP) | Laura Liess, Roland Jesske, Alan Johnston, Dale Worley, Paul Kyzivat | 2012-04-05 |
| The Session Initiation Protocol (SIP) supports the capability to provide a reference to a specific rendering to be used by the UA when the user is alerted. This is done using the Alert-Info header field. However, the reference addresses only network resources with specific rendering properties. There is currently no support for predefined standard identifiers for describing the semantics of the alerting situation or the characteristics of the alerting signal, without being tied to a particular rendering. To overcome this limitation and support new applications, a new family of URNs for use in SIP Alert-Info header fields is defined in this specification. This document normatively updates [RFC3261], the Session Initiation Protocol (SIP). It changes the usage of the SIP Alert-Info header field defined in the [RFC3261] by additionally allowing its use in all provisional responses to INVITE (except the 100 response). | ||
| P-Charge-Info - A Private Header (P-Header) Extension to the Session Initiation Protocol (SIP) | Dan York, Tolga Asveren | 2012-04-03 |
| This document describes \'P-Charge-Info\', a private Session Initiation Protocol (SIP) header (P-header) used to convey billing information about the party to be charged. This P-Header is currently in production usage by a number of equipment vendors and carriers and this document is submitted to request the registration of this header with IANA. This P-Header may also be used in some situations to carry the ISUP Charge Number parameter for PSTN interconnection. | ||
| Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP) | Keith Drage, Christer Holmberg, Roland Jesske | 2012-04-02 |
| This document describes a set of private Session Initiation Protocol (SIP) header fields (P-headers) used by the 3rd-Generation Partnership Project (3GPP), along with their applicability, which is limited to particular environments. The P-header fields are for a variety of purposes within the networks that the partners use, including charging and information about the networks a call traverses. | ||
| The References Header for SIP | Dale Worley | 2012-04-02 |
| This document defines a SIP extension header, References, to be used within SIP messages to signify that the message (and the dialog containing it) is related to one or more other dialogs. It is expected to be used largely for diagnostic purposes. | ||
| The Continue Header Field for the Session Initiation Protocol (SIP) | Amardeep Sinha, Subhrajyoti De, Sunil Sinha | 2012-04-02 |
| Before placing a call, it is quite often useful for the Caller to know whether a Callee is in favorable state to receive a call or not. This document defines an optional tag "continue" and a header "Continue" to address the purpose. The "Continue" header field is to confirm the session continuity with the Callee from the Caller after an option for session continuity is placed by the Callee based on the unfavorable state of the Callee. This functionality is needed to resolve the unwillingness of the Callee to receive any call. An option is given to the Callee by the Service Provider or by the Handset Manufacturer or by the Carrier to establish this requirement. | ||
| A Session Initiation Protocol (SIP) Event Package for Communication Diversion Information in support of the Communication Diversion (CDIV) Notification (CDIVN) CDIV service | John-Luc Bakker, Ranjit Avasarala | 2012-03-27 |
| 3GPP and TISPAN are defining the protocol specification for the Communication Diversion (CDIV) service using IP Multimedia (IM) Core Network (CN) subsystem supplementary service. As part of CDIV, a SIP based Event package framework is used for notifying users about diversions (re-directions or forwarding) of their incoming communication sessions. This document proposes a new SIP event package for allowing users to subscribe to and receive such notifications. Users can further define filters to control the rate and content of such notifications. The proposed event package is applicable to the CDIV supplementary service in IMS and may not be applicable to the general internet. . | ||
| Session Initiation Protocol (SIP) History-Info Header Call Flow Examples | Mary Barnes, Francois Audet, Shida Schubert, Hans van Elburg, Christer Holmberg | 2012-03-27 |
| This document describes use cases and documents call flows which require the History-Info header field to capture the Request-URIs as a Session Initiation Protocol (SIP) Request is retargeted. The use cases are described along with the corresponding call flow diagrams and messaging details. | ||
| Methodology for Benchmarking SIP Networking Devices | Carol Davids, Vijay Gurbani, Scott Poretsky | 2012-03-12 |
| This document describes the methodology for benchmarking Session Initiation Protocol (SIP) performance as described in SIP benchmarking terminology document. The methodology and terminology are to be used for benchmarking signaling plane performance with varying signaling and media load. Both scale and establishment rate are measured by signaling plane performance. The SIP Devices to be benchmarked may be a single device under test (DUT) or a system under test (SUT). Benchmarks can be obtained and compared for different types of devices such as SIP Proxy Server, SBC, and server paired with a media relay or Firewall/NAT device. | ||
| Terminology for Benchmarking Session Initiation Protocol (SIP) Networking Devices | Carol Davids, Vijay Gurbani, Scott Poretsky | 2012-03-12 |
| This document provides a terminology for benchmarking SIP performance in networking devices. Terms are included for test components, test setup parameters, and performance benchmark metrics for black-box benchmarking of SIP networking devices. The performance benchmark metrics are obtained for the SIP control plane and media plane. The terms are intended for use in a companion methodology document for complete performance characterization of a device in a variety of conditions making it possible to compare performance of different devices. It is critical to provide test setup parameters and a methodology document for SIP performance benchmarking because SIP allows a wide range of configuration and operational conditions that can influence performance benchmark measurements. It is necessary to have terminology and methodology standards to ensure that reported benchmarks have consistent definition and were obtained following the same procedures. Benchmarks can be applied to compare performance of a variety of SIP networking devices. | ||
| Format for the Session Initiation Protocol (SIP) Common Log Format (CLF) | Gonzalo Salgueiro, Vijay Gurbani, Adam Roach | 2012-03-12 |
| The SIPCLF Workgroup has defined a common log format framework for Session Initiation Protocol (SIP) servers. This common log format mimics the successful event logging format found in well-known web servers like Apache and web proxies like Squid. This document proposes an indexed text encoding format for the SIP Common Log Format (CLF) that retains the key advantages of a text-based format, while significantly increasing processing performance over a purely text-based implementation. This file format adheres to the SIP CLF data model and provides an effective encoding scheme for all mandatory and optional fields that appear in a SIP CLF record. | ||
| Session Initiation Protocol (SIP) Recording Metadata | Ram R, Parthasarathi Ravindran, Paul Kyzivat | 2012-03-12 |
| Session recording is a critical requirement in many communications environments such as call centers and financial trading. In some of these environments, all calls must be recorded for regulatory, compliance, and consumer protection reasons. Recording of a session is typically performed by sending a copy of a media stream to a recording device. This document describes the metadata model as viewed by Session Recording Server(SRS) and the Recording metadata format. | ||
| Session Initiation Protocol (SIP) Overload Control | Vijay Gurbani, Volker Hilt, Henning Schulzrinne | 2012-03-12 |
| 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 the behaviour of SIP servers involved in overload control, and in addition, it specifies a loss-based overload scheme for SIP. | ||
| The Common Log Format (CLF) for the Session Initiation Protocol (SIP): Framework and Data Model | Vijay Gurbani, Eric Burger, Tricha Anjali, Humberto Abdelnur, Olivier Festor | 2012-03-09 |
| Well-known web servers such as Apache and web proxies like Squid support event logging using a common log format. The logs produced using these de-facto standard formats are invaluable to system administrators for trouble-shooting a server and tool writers to craft tools that mine the log files and produce reports and trends. Furthermore, these log files can also be used to train anomaly detection systems and feed events into a security event management system. The Session Initiation Protocol (SIP) does not have a common log format, and as a result, each server supports a distinct log format that makes it unnecessarily complex to produce tools to do trend analysis and security detection. We propose a common log file format for SIP servers that can be used uniformly by user agents, proxies, registrars, redirect servers as well as back-to-back user agents. | ||
| Session Initiation Protocol (SIP) Rate Control | Eric Noel, Philip Williams | 2012-03-08 |
| The prevalent use of Session Initiation Protocol (SIP) [RFC3261] in Next Generation Networks necessitates that SIP networks provide adequate control mechanisms to maintain transaction throughput by preventing congestion collapse during traffic overloads. Already [draft-ietf-soc-overload-control-07] proposes a loss-based solution to remedy known vulnerabilities of the [RFC3261] SIP 503 (service unavailable) overload control mechanism. This document proposes a rate-based control solution to complement the loss-based control defined in [draft-ietf-soc-overload-control-07]. | ||
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