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RECENT IETF DRAFTS
SIP internet drafts statistics
- 93 SIP related internet drafts (IETF).
- 7 new and updated drafts posted in the last 14 days.
| Title | Author | Date |
| SIP Forum - Message Body in SIP REGISTER Message | Vidyut Gupta | 2010-09-02 |
| In SIP, Register message normally don\'t carry message body, but a lot of things can be simplified by putting the one. User can configure their call preferences at any moment by just sending a Register message. In present scenario when users are becoming more and more demanding it\'s perfectly logical to allow them to configure their call preferences dynamically. The present draft present two such call preferences requested by user - call barring and advertising unavailability. It can, however, be extended for other scenarios as well. | ||
| Requirements for SIP-based Media Recording (SIPREC) | Ken Rehor, Rajnish Jain, Leon Portman, Andrew Hutton | 2010-09-01 |
| Session recording is a critical requirement in many business communications environments such as call centers and financial trading floors. In some of these environments, all calls must be recorded for regulatory and compliance reasons. In others, calls may be recorded for quality control or business analytics. Recording is typically done by sending a copy of the session media to the recording devices. This document specifies requirements for extensions to SIP that will manage delivery of RTP media from an end- point that originates media (or that has access to it) to a recording device. This is being referred to as SIP-based Media Recording. | ||
| Session Initiation Protocol (SIP) Resource availability Event package | Parthasarathi R, Sheshadri Shalya | 2010-08-30 |
| Collecting Resource availability information in Session Initiation Protocol (SIP) devices in real time helps in better managebility of resources in the SIP network. Resource availability monitoring in SIP devices helps administrator to take informed decision and corrective measures to tackle under or over resource utilization in the network. Resource availability information can also be used for SIP overload control in SIP based Media servers by passing resource demand vector between devices. This document defines resource availability XML document and the mechanisms that can be used to exchange the document between SIP entities. | ||
| Migrating SIP to IPv6 Media Without Connectivity Checks | Dan Wing, Andrew Yourtchenko | 2010-08-27 |
| During the migration from IPv4 to IPv6, it is anticipated that an IPv6 path might be broken for a variety of reasons, causing endpoints to not receive RTP data. Connectivity checks would detect and avoid the user noticing such a problem, but there is industry reluctance to implement connectivity checks. This document describes a mechanism allowing dual-stack SIP endpoints to attempt communications over IPv6 and fall back to IPv4 if the IPv6 path is not working. The mechanism does not require connectivity checks. | ||
| Session Initiation Protocol (SIP) Overload Control | Vijay Gurbani, Volker Hilt, Henning Schulzrinne | 2010-08-23 |
| 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. | ||
| Session Initiation Protocol (SIP) Recording Metadata | Ram R, Parthasarathi R, Paul Kyzivat | 2010-08-23 |
| 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). | ||
| Requirements and Use Cases for Combining SIP Based Real-time Media Sessions With XMPP Based Instant Messaging and Presence Service. | Simo Veikkolainen, Markus Isomaki | 2010-08-22 |
| This memo defines use cases and requirements for combining Session Initiation Protocol (SIP) based real-time media sessions with Extensible Messaging and Presence Protocol (XMPP) based instant messaging and presence services in a seamless manner. | ||
| A Session Initiation Protocol (SIP) INFO Package for Dual-Tone Multi-Frequency (DTMF) Events | Hadriel Kaplan | 2010-08-20 |
| The SIP INFO request method now supports explicit indication and exchange of specified application information. Such usages are documented as an "INFO Package", following the requirements outlined in [info-packages]. This document specifies one such SIP INFO Package, for the purpose of indicating DTMF signals. | ||
| A Common Log Format for SIP using IPFIX Files | Saverio Niccolini, Benoit Claise, Brian Trammell, Hadriel Kaplan | 2010-08-17 |
| This draft defines a log file format conforming to the information model defined in the SIP-CLF problem statement based upon the IPFIX File Format. It details the creation and interpretation of these files, and provides examples based on common SIP situations. | ||
| SIP-Specific Event Notification | Adam Roach | 2010-08-17 |
| This document describes an extension to the Session Initiation Protocol (SIP). 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. | ||
| Design Considerations for Session Initiation Protocol (SIP) Overload Control | Volker Hilt, Eric Noel, Charles Shen, Ahmed Abdelal | 2010-08-12 |
| 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 discusses models and design considerations for a SIP overload control mechanism. | ||
| transit ioi extension for the P-Charging-Vector header in SIP (Session Initiation Protocol) | Roland Jesske | 2010-08-09 |
| This document adds the term transit-ioi to the P-Charging-Vector Header to mark the transit network in interconnection cases. | ||
| Session Initiation Protocol (SIP) Header Parameter for Debugging | Peter Dawes | 2010-08-06 |
| 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 use the network. In order to allow troubleshooting and regression testing, it is useful to provide debugging as part of the network fabric. This draft describes an event package that provides debugging configuration to SIP entities and 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. | ||
| Basic Telephony SIP End-to-End Performance Metrics | Daryl Malas, Al Morton | 2010-07-30 |
| This document defines a set of metrics and their usage to evaluate the performance of end-to-end Session Initiation Protocol (SIP) for telephony services in both production and testing environments. The purpose of this document is to combine a standard set of common metrics, allowing interoperable performance measurements, easing the comparison of industry implementations. | ||
| A Session Initiation Protocol (SIP) Extension for the Identification of Services | Keith Drage | 2010-07-28 |
| This document describes private extensions to the Session Initiation Protocol (SIP) that enable a network of trusted SIP servers to assert the service of authenticated users. The use of these extensions is only applicable inside an administrative domain with previously agreed-upon policies for generation, transport and usage of such information. This document does NOT offer a general service identification model suitable for use between different trust domains, or use in the Internet at large. The document also defines a URN to identify both services and UA applications. This URN can be used within the SIP header fields defined in this document to identify services, and also within the framework defined for caller preferences and callee capabilities to identify usage of both services and applications between end UAs. | ||
| SOS Uniform Resource Identifier (URI) Parameter for Marking of Session Initiation Protocol (SIP) Requests related to Emergency Services | Milan Patel | 2010-07-27 |
| This document defines a new Session Initiation Protocol (SIP) Uniform Resource Identifier (URI) parameter intended for marking SIP registration requests related to emergency services. The URI parameter is extensible to allow future values to be defined if required by other use cases that require specific SIP registrations to be distinctly identified. The usage of this new URI parameter complements the usage of the Service Uniform Resource Name (URN) and is not intended to replace it. | ||
| Requirements for multiple address of record (AOR) reachability information in the Session Initiation Protocol (SIP) | John Elwell, Hadriel Kaplan | 2010-07-26 |
| This document states requirements for a standardized SIP registration mechanism for multiple addresses of record, the mechanism being suitable for deployment by SIP service providers on a large scale in support of small to medium sized Private Branch Exchanges (PBXs). The requirements are for a solution that can, as a minimum, support AORs based on E.164 numbers. | ||
| Routing of Service Numbers with-in SIP (Session Initiation Protocol) networks | Roland Jesske | 2010-07-26 |
| The combination of "rn" and "npdi" parameters which are normally used for number portability (NP) can also solve numbering and routing problems. Database dips to obtain routing numbers are not only needed for NP, but also for the routing of service numbers and short code numbers in the PSTN and also in SIP networks. This document defines the use of the tel URI parameters defined for NP ("rn" and "npdi") to route service numbers and short code numbers. | ||
| Re-INVITE and Target-refresh Request Handling in the Session Initiation Protocol (SIP) | Gonzalo Camarillo, Christer Holmberg, Gao yang | 2010-07-26 |
| In this document, we clarify the handling of re-INVITEs in SIP. We clarify in which situations a UAS (User Agent Server) should generate a success response and in which situations a UAS should generate an error response to a re-INVITE. Additionally, we clarify issues related to target-refresh requests. | ||
| Shared Appearances of a Session Initiation Protocol (SIP) Address of Record (AOR) | Alan Johnston, Mohsen Soroushnejad, Venkatesh Venkataramanan | 2010-07-12 |
| 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. | ||
| Methodology for Benchmarking SIP Networking Devices | Scott Poretsky, Vijay Gurbani, Carol Davids | 2010-07-12 |
| 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 | Scott Poretsky, Vijay Gurbani, Carol Davids | 2010-07-12 |
| 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. | ||
| A Session Identifier for the Session Initiation Protocol (SIP) | Hadriel Kaplan | 2010-07-12 |
| There is a need for having a globally unique session identifier for the same SIP session, which can be consistently maintained across Proxies, B2BUAs and other SIP middle-boxes, for the purpose of Troubleshooting. This draft proposes a new SIP header to carry such a value: Session-ID. | ||
| Routing SIP Requests with ENUM | Hadriel Kaplan, Colin Pons | 2010-07-12 |
| A common ENUM use-case is for hop-by-hop or domain-by-domain "routing" of SIP requests, using private DNS trees and servers. This document describes this use-case and the need for a source- based query/answer mechanism for such. | ||
| A SIP Usage for RELOAD | Cullen Jennings, Bruce Lowekamp, Eric Rescorla, Salman Baset, Henning Schulzrinne | 2010-07-12 |
| This document defines a SIP Usage for REsource LOcation And Discovery (RELOAD), The SIP Usage provides the functionality of a SIP proxy or registrar in a fully-distributed system. The SIP Usage provides lookup service for AoRs stored in the overlay. The SIP Usage also defines GRUUs that allow the registrations to map an AoR to a specific node reachable through the overlay. The AppAttach method is used to establish a direct connection between nodes through which SIP messages are exchanged. | ||
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