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RECENT IETF DRAFTS
SIP internet drafts statistics
- 153 SIP related internet drafts (IETF).
- 11 new and updated drafts posted in the last 14 days.
| Title | Author | Date |
| VoIP SIP Peering Use Cases | Adam Uzelac, Yiu Lee | 2008-05-08 |
| This document depicts many common VoIP use case for SIP Peering. These use cases are categorized into static and on-demand, and then further sub-categorized into direct and indirect. These use cases are not an exhaustive set, but rather the most common use cases deployed today. This document captures them to provide a reference. | ||
| Requirements for vertical handover of multimedia sessions using SIP | Saverio Niccolini, Stefano Salsano, Luca Veltri | 2008-05-07 |
| This document analyses the issue of handling vertical handovers among different network technologies using SIP. | ||
| A solution for vertical handover of multimedia sessions using SIP | Stefano Salsano, Saverio Niccolini, Luca Veltri, Andrea Polidoro | 2008-05-07 |
| 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"). | ||
| Session Initiation Protocol (SIP) Event Package for the Common Alerting Protocol (CAP) | Brian Rosen, Henning Schulzrinne, Hannes Tschofenig | 2008-05-07 |
| 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). | ||
| Addressing an Amplification Vulnerability in Session Initiation Protocol (SIP) Forking Proxies | Robert Sparks, Scott Lawrence, Alan Hawrylyshen, Byron Campen | 2008-05-06 |
| This document normatively updates RFC 3261, the Session Initiation Protocol (SIP), to address a security vulnerability identified in SIP proxy behavior. This vulnerability enables an attack against SIP networks where a small number of legitimate, even authorized, SIP requests can stimulate massive amounts of proxy-to-proxy traffic. This document strengthens loop-detection requirements on SIP proxies when they fork requests (that is, forward a request to more than one destination). It also corrects and clarifies the description of the loop-detection algorithm such proxies are required to implement. Additionally, this document defines a Max-Breadth mechanism for limiting the number of concurrent branches pursued for any given request. | ||
| A Session Initiation Protocol (SIP) Event Package for Debugging | Peter Dawes | 2008-05-03 |
| 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. | ||
| Private Extension to the Session Initiation Protocol (SIP) for Debugging | Peter Dawes | 2008-05-03 |
| 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. | ||
| Proxy Mutual Authentication in SIP | Steve Dotson, Stuart Hoggan, Sumanth Channabasappa | 2008-05-01 |
| This document defines the Proxy-Authentication-Info header field for the Session Initiation Protocol (SIP). When a UA is required to authenticate to a proxy using digest authentication specified in SIP this header field allows for the UA to authenticate the proxy, enabling mutual authentication. This header field can also provide integrity checks over the bodies. | ||
| A Framework for Session Initiation Protocol (SIP) Session Policies | Volker Hilt, Gonzalo Camarillo, Jonathan Rosenberg | 2008-04-27 |
| Proxy servers play a central role as an intermediary in the Session Initiation Protocol (SIP) as they define and impact policies on call routing, rendezvous, and other call features. This document specifies a framework for SIP session policies that provides a standard mechanism by which a proxy can define or influence policies on sessions, such as the codecs or media types to be used. It defines a model, an overall architecture and new protocol mechanisms for session policies. | ||
| Best Current Practices for NAT Traversal for SIP | Chris Boulton, Jonathan Rosenberg, Gonzalo Camarillo | 2008-04-25 |
| Traversal of the Session Initiation Protocol (SIP) and the sessions it establishes through Network Address Translators (NAT) is a complex problem. Currently there are many deployment scenarios and traversal mechanisms for media traffic. This document aims to provide concrete recommendations and a unified method for NAT traversal as well as documenting corresponding flows. | ||
| SIP (Session Initiation Protocol) Usage of the Offer/Answer Model | Takuya Sawada, Paul Kyzivat | 2008-04-25 |
| The Session Initiation Protocol (SIP) utilizes the offer/answer model to establish and update multimedia sessions using the Session Description Protocol (SDP). The description of the offer/answer model in SIP is dispersed across multiple RFCs. This document summarizes all the current usages of the offer/answer model in SIP communication. | ||
| The Session Initiation Protocol (SIP) P-Served-User Private-Header (P-Header) | Hans Erik van Elburg | 2008-04-21 |
| This document specifies the SIP P-Served-User P-header. This header field addresses an issue that was found in the 3rd-Generation Partnership Project (3GPP) IMS (IP Multimedia Subsystem) between an S-CSCF (Serving Call Session Control Function) and an AS (Application Server) on the ISC (IMS Subsystem Service Control) interface to convey the identity of the served user and the session case that applies to this particular communication session and application invocation. | ||
| Private Extensions to the Session Initiation Protocol (SIP) for Service Interaction Indicator | Yuzhong Shen | 2008-04-18 |
| In SIP-based networks, a SIP session MAY involve several application servers on the originating and terminating side. In a certain case, an application server needs to set some indications in SIP message to indicate service information (what are invoked, what can be allowed and what should blocked). This kind of information will be also required for composition of SIP applications. There is a need to provide indicators for service interaction between SIP application servers or other SIP endpoints. This document describes a mechanism of service interaction indicator for the Session Initiation Protocol (SIP) that enhances service interaction between SIP application servers in a trusted network. | ||
| A Call Control and Multi-party usage framework for the Session Initiation Protocol (SIP) | Rohan Mahy, Robert Sparks, Jonathan Rosenberg, Dan Petrie, Alan Johnston | 2008-04-16 |
| This document defines a framework and requirements for call control and multi-party usage of SIP. To enable discussion of multi-party features and applications we define an abstract call model for describing the media relationships required by many of these. The model and actions described here are specifically chosen to be independent of the SIP signaling and/or mixing approach chosen to actually setup the media relationships. In addition to its dialog manipulation aspect, this framework includes requirements for communicating related information and events such as conference and session state, and session history. This framework also describes other goals that embody the spirit of SIP applications as used on the Internet. | ||
| Certificate Management Service for The Session Initiation Protocol (SIP) | Cullen Jennings, Jason Fischl | 2008-04-05 |
| This draft defines a Credential Service that allows Session Initiation Protocol (SIP) User Agents (UAs) to use a SIP event package to discover the certificates of other users. This mechanism allows user agents that want to contact a given Address-of-Record (AOR) to retrieve that AOR\'s certificate by subscribing to the Credential Service, which returns an authenticated response containing that certificate. The Credential Service also allows users to store and retrieve their own certificates and private keys. | ||
| Framework for real-time text over IP using the Session Initiation Protocol (SIP) | Arnoud Van Wijk, Guido Gybels | 2008-04-04 |
| This document lists the essential requirements for real-time Text- over-IP (ToIP) and defines a framework for implementation of all required functions based on the Session Initiation Protocol (SIP) and the Real-Time Transport Protocol (RTP). This includes interworking between Text-over-IP and existing text telephony on the PSTN and other networks. | ||
| Updates to Asserted Identity in the Session Initiation Protocol (SIP) | John Elwell | 2008-04-04 |
| SIP has a mechanism for conveying the asserted identity of the originator of a request by means of the P-Asserted-Identity header field. This header field is specified for use in requests using a number of SIP methods, in particular the INVITE method. However, RFC 3325 does not specify the insertion of this header field by a trusted UAC, does not specify the use of this header field with the SIP UPDATE, REGISTER, MESSAGE or PUBLISH methods, and is unclear on the use of this header field in responses. This document extends RFC 3325 to cover these situations. This work is being discussed on the sipping@ietf.org mailing list. | ||
| Private Header Extension to SIP for Support of Mobility Operations | Michael Coulas, Apostolis Salkintzis | 2008-04-02 |
| Voice call continuity and media over IP session mobility operations are typically characterized by the movement of a specific segment of a communication session across networks and/or devices. This movement may be implemented through the use of SIP procedures that result in the establishment of a new segment replacing the existing segment. This document proposes a private extension to the Session Initiation Protocol (SIP) that enable User Agents (UA) to indicate that a SIP request is associated with a mobility operation for an existing session. | ||
| Session Initiation Protocol (SIP) Version 4.0: P2P2PSIP | Hadriel Kaplan, Bob Penfield | 2008-04-01 |
| This document defines a new and improved version of SIP, which tastes great and is less filling than the previous SIP. This draft updates all previous and future RFCs related to SIP in SIPPING, SIMPLE, MMUSIC, BEHAVE, and so on. | ||
| Guidelines for Using the Privacy Mechanism for SIP | Mayumi Munakata, Shida Schubert, Takumi Ohba | 2008-03-27 |
| This is an informational document that provides guidelines for using the privacy mechanism for Session Initiation Protocol (SIP), that is specified in RFC 3323 and subsequently extended in RFCs 3325 and 4244. It is intended to clarify the handling of the target SIP headers/parameters and SDP parameters for each of the privacy header values (priv-values). | ||
| Requirements from SIP (Session Initiation Protocol) Session Border Control Deployments | Jani Hautakorpi, Gonzalo Camarillo, Bob Penfield, Alan Hawrylyshen, Medhavi Bhatia | 2008-03-25 |
| This document describes functions implemented in Session Initiation Protocol (SIP) intermediaries known as Session Border Controllers (SBCs). The goal of this document is to describe the commonly provided functions of SBCs. A special focus is given to those practices that are viewed to be in conflict with SIP architectural principles. This document also explores the underlying requirements of network operators that have led to the use of these functions and practices in order to identify protocol requirements and determine whether those requirements are satisfied by existing specifications or additional standards work is required. | ||
| Managing Client Initiated Connections in the Session Initiation Protocol (SIP) | Cullen Jennings, Rohan Mahy | 2008-03-21 |
| The Session Initiation Protocol (SIP) allows proxy servers to initiate TCP connections and send asynchronous UDP datagrams to User Agents in order to deliver requests. However, many practical considerations, such as the existence of firewalls and Network Address Translators (NATs), prevent servers from connecting to User Agents in this way. This specification defines behaviors for User Agents, registrars and proxy servers that allow requests to be delivered on existing connections established by the User Agent. It also defines keep alive behaviors needed to keep NAT bindings open and specifies the usage of multiple connections from the User Agent to its Registrar. | ||
| Session Initiation Protocol (SIP) User Agent Capability Extension to Presence Information Data Format (PIDF) | Mikko Lonnfors, Krisztian Kiss | 2008-03-18 |
| Presence Information Data Format (PIDF) defines a common presence data format for Common Profile for Presence (CPP) compliant Presence protocols. This memo defines an extension to represent SIP User Agent capabilities in the Presence Information Document Format (PIDF) compliant presence documents. | ||
| Determining When to Discard or Retry a SIP Request During Overload | Timothy Moran | 2008-03-14 |
| 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. | ||
| IANA Registration of New Session Initiation Protocol (SIP) Resource-Priority Namespaces | James Polk | 2008-03-13 |
| This document creates additional Session Initiation Protocol (SIP) Resource-Priority namespaces to meet the requirements of the US Defense Information Systems Agency, and places these namespaces in the IANA registry. | ||
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SIP SPOKESPERSON
Anwar Siddiqui is currently a Technical Manager of the Avaya Chief Technology Office (CTO) in Lincroft, New Jersey.
