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RECENT IETF DRAFTS

SIP internet drafts statistics

  • 36 SIP related internet drafts (IETF).
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Title Author Date
Terminology for Benchmarking Session Initiation Protocol (SIP) Devices: Basic session setup and registration Carol Davids, Vijay Gurbani, Scott Poretsky 0000-00-00
This document provides a terminology for benchmarking the Session Initiation Protocol (SIP) performance of devices. Methodology related to benchmarking SIP devices is described in the companion methodology document. Using these two documents, benchmarks can be obtained and compared for different types of devices such as SIP Proxy Servers, Registrars and Session Border Controllers. The term "performance" in this context means the capacity of the device-under- test (DUT) to process SIP messages. Media streams are used only to study how they impact the signaling behavior. The intent of the two documents is to provide a normalized set of tests that will enable an objective comparison of the capacity of SIP devices. Test setup parameters and a methodology is necessary because SIP allows a wide range of configuration and operational conditions that can influence performance benchmark measurements. A standard terminology and methodology will ensure that benchmarks have consistent definition and were obtained following the same procedures.

Methodology for Benchmarking Session Initiation Protocol (SIP) Devices: Basic session setup and registration Carol Davids, Vijay Gurbani, Scott Poretsky 0000-00-00
This document provides a methodology for benchmarking the Session Initiation Protocol (SIP) performance of devices. Terminology related to benchmarking SIP devices is described in the companion terminology document. Using these two documents, benchmarks can be obtained and compared for different types of devices such as SIP Proxy Servers, Registrars and Session Border Controllers. The term "performance" in this context means the capacity of the device-under- test (DUT) to process SIP messages. Media streams are used only to study how they impact the signaling behavior. The intent of the two documents is to provide a normalized set of tests that will enable an objective comparison of the capacity of SIP devices. Test setup parameters and a methodology are necessary because SIP allows a wide range of configuration and operational conditions that can influence performance benchmark measurements.

Requirements for Marking SIP Messages to be Logged Peter Dawes 0000-00-00
SIP networks use signalling monitoring tools to diagnose user reported problem and for regression testing if network or client software is upgraded. As networks grow and become interconnected, including connection via transit networks, it becomes impractical to predict the path that SIP signalling will take between clients, and therefore impractical to monitor SIP signalling end-to-end. This draft describes requirements for adding an indicator to the SIP protocol which can be used to mark signalling as of interest to logging. Such marking will typically be applied as part of network testing controlled by the network operator and not used in regular client signalling. However, such marking can be carried end-to-end including the SIP terminals, even if a session originates and terminates in different networks.

Marking SIP Messages to be Logged Peter Dawes 0000-00-00
SIP networks use signalling monitoring tools to diagnose user reported problems and for regression testing if network or user agent software is upgraded. As networks grow and become interconnected, including connection via transit networks, it becomes impractical to predict the path that SIP signalling will take between user agents, and therefore impractical to monitor SIP signalling end-to-end. This document describes an indicator for the SIP protocol which can be used to mark signalling as of interest to logging. Such marking will typically be applied as part of network testing controlled by the network operator and not used in regular user agent signalling. However, such marking can be carried end-to-end including the SIP user agents, even if a session originates and terminates in different networks.

Using Interactive Connectivity Establishment (ICE) with Session Description Protocol (SDP) offer/answer and Session Initiation Protocol (SIP) Marc Petit-Huguenin, Ari Keranen 0000-00-00
This document describes how Interactive Connectivity Establishment (ICE) is used with Session Description Protocol (SDP) offer/answer and Session Initiation Protocol (SIP).

A Session Initiation Protocol (SIP) usage for Trickle ICE Emil Ivov, Enrico Marocco, Christer Holmberg 0000-00-00
The Interactive Connectivity Establishment (ICE) protocol describes a Network Address Translator (NAT) traversal mechanism for UDP-based multimedia sessions established with the offer/answer model. The ICE extension for Incremental Provisioning of Candidates (Trickle ICE) defines a mechanism that allows ICE agents to shorten session establishment delays by making the candidate gathering and connectivity checking phases of ICE non-blocking and by executing them in parallel. This document defines usage semantics for Trickle ICE with the Session Initiation Protocol (SIP).

P-Charge-Info - A Private Header (P-Header) Extension to the Session Initiation Protocol (SIP) Dan York, Tolga Asveren 0000-00-00
This text documents \'P-Charge-Info\', an existing 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. NOTE: This document has been in development since 2008 under the name draft-york-sipping-p-charge-info. This -04 document is identical to draft-york-sipping-p-charge-info-15 except for edits to the text to indicate this is now for the DISPATCH working group as the SIPPING working group no longer exists.

Session Initiation Protocol (SIP) Instance ID usage by the Open Mobile Alliance Push-to-talk over Cellular Andrew Allen, Alan Soloway 0000-00-00
This document describes how the SIP Instance ID as defined in RFC 5626 [1] is used by the Open Mobile Alliance (OMA), for Push-to-talk over Cellular (PoC) and Push-to-Communicate for Public Safety (PCPS) and addresses security concerns with use of the SIP instance ID in non-register requests and responses. This document updates RFC 7255 [2] to allow the use of the International Mobile Equipment Identity (IMEI) as an instance ID in the Contact header field of non-register requests and responses by the OMA PoC and PCPS enablers for the purposes described in this document.

Locating Session Initiation Protocol (SIP) Servers in a Dual-Stack IP Network oej, Gonzalo Salgueiro 0000-00-00
RFC 3263 defines how a Session Initiation Protocol (SIP) implementation, given a SIP Uniform Resource Identifier (URI), should locate the next hop SIP server using Domain Name System (DNS) procedures. The specification repeatedly states that the implementation should look up IPv4 or IPv6 addresses. This is not a suitable solution and one that can cause severely degraded user experience dual-stack clients, as detailed in the Happy Eyeballs specification. This document specifies amended procedures for dual- stack SIP implementations so that they look up both IPv4 and IPv6 addresses. This way, the SIP implementation can find the preferred network path and protocol with an improved chance of successfully reaching the desired service. This document also clarifies DNS SRV usage for single-stack clients.

PCP for SIP Deployments in Managed Networks Mohamed Boucadair, Parthasarathi Ravindran 0000-00-00
This document discusses how PCP (Port Control Protocol) can be used in SIP deployments in managed networks. This document applies for both IPv4 and IPv6.

Mapping and interworking of Diversion information Between Diversion and History-Info Headers in the Session Initiation Protocol (SIP) Marianne Mohali 0000-00-00
Although the SIP History-Info header field is the solution adopted in IETF, the non-standard Diversion header field is nevertheless already implemented and used for conveying call diversion related information in the Session Initiation Protocol (SIP) signaling. On one hand, the non-standard Diversion header field is described, as Historic, in [RFC5806]. On the other hand, the History-Info header field is described in [RFC7044] that obsoletes the original[RFC4244] describing the History-Info header field. [RFC7044] defines the SIP header field, History-Info, for capturing the history information in requests and new 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. [RFC7044] also defines a value for the Privacy header field that directs the anonymization of values in the History-Info header field. Since the Diversion header field is used in existing network implementations for the transport of call diversion information, its interworking with the SIP History-Info standardized solution is needed. This document describes a recommended interworking guideline between the Diversion header field and the History-Info header field to handle call diversion information. In addition, an interworking policy is proposed to manage the headers\' coexistence. This work is intended to enable the migration from non-standard implementations and deployments toward IETF specification-based implementations and deployments. This document obsoletes [RFC6044]that describes the interworking between the Diversion header field [RFC5806] and the obsoleted History-Info header field as defined on [RFC4244].

Overview for MSRP Recording based on SIPREC Michael Yan, Paul Kyzivat 0000-00-00
SIPREC is capable of recording interactive text media that is transmitted via RTP. However that format is not commonly used for message or chat scenarios. There is also a need for recording text media carried via MSRP. One case of note is exchange of text between hearing-impaired users and emergence service bureaus. Also, recording support is needed for MSRP used in chat conferences and multimedia conferences. This document describes how to achieve MSRP channel recording within the mechanism of SIP Recording (SIPREC).

3rd-Generation Partnership Project (3GPP) SIP URI Inter Operator Traffic Leg parameter Christer Holmberg, Jan Holm, Roland Jesske, Martin Dolly 0000-00-00
In 3rd-Generation Partnership Project (3GPP) networks, the signalling path between a calling user and a called user can be partioned into segments, referred to as traffic legs. Each traffic leg may span networks belonging to different operators, and will have its own characteristics that can be different from other traffic legs in the same call. A traffic leg might be associated with multiple SIP dialogs, e.g. in case a B2BUA which modifies the SIP dialog identifier is located within the traffic leg. This document defines a new SIP URI parameter, \'iotl\'. The parameter can be used in a SIP URI to indicate that the entity associated with the address, or an entity responsible for the host part of the address, represents the end of a specific traffic leg (or multiple traffic legs). The SIP URI \'iotl\' parameter defined in this document has known uses in 3GPP networks. Usage in other networks is also possible.

The Session Initiation Protocol (SIP) OAuth Rifaat Shekh-Yusef, Victor Pascual 0000-00-00
This document defines an authorization framework for SIP that is based on the OAuth 2.0 framework, and adds a simple identity layer on top of that, based on the OpenID Connect Core 1.0, to enable Clients to verify the identity of the End-User based on the authentication performed by an Authorization Server, as well as to obtain basic profile information about the End-User.

DTLS-SRTP Handling in Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) Ram R, Tirumaleswar Reddy, Gonzalo Salgueiro, Victor Pascual, Parthasarathi Ravindran 0000-00-00
Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) often function on the media plane, rather than just on the signaling path. This document describes the behavior B2BUAs should follow when acting on the media plane that use Secure Real-time Transport (SRTP) security context setup with Datagram Transport Layer Security (DTLS) protocol.

Interoperability Impacts of IPv6 Interworking with Existing IPv4 SIP Implementations Carl Klatsky, oej, Rifaat Shekh-Yusef, Andrew Hutton, Gonzalo Salgueiro 0000-00-00
This document captures potential impacts to IPv4 SIP implementations when interworking with IPv6 SIP implementations. Although some amount of interworking translation will occur at the network and application layers, an IPv4 SIP application may still encounter a SIP message with some IPv6 values in it, resulting in unforeseen error conditions. Such potential scenarios will be identified in this document so that SIP application developers can define solutions to handle these cases. Note, this document is not intended to be an exhaustive list, rather to provide an overview of some of the more commonly encountered potential scenarios.

TLS sessions in SIP using DNS-based Authentication of Named Entities (DANE) TLSA records oej 0000-00-00
Use of TLS in the SIP protocol is defined in multiple documents, starting with RFC 3261. The actual verification that happens when setting up a SIP TLS connection to a SIP server based on a SIP URI is described in detail in RFC 5922 - SIP Domain Certificates. In this document, an alternative method is defined, using DNS-Based Authentication of Named Entities (DANE). By looking up TLSA DNS records and using DNSsec protection of the required queries, including lookups for NAPTR and SRV records, a SIP Client can verify the identity of the TLS SIP server in a different way, matching on the SRV host name in the X.509 PKIX certificate instead of the SIP domain. This provides more scalability in hosting solutions and make it easier to use standard CA certificates (if needed at all).

Session Initiation Protocol (SIP) Cause URI parameter for Service Number translation Marianne Mohali 0000-00-00
[RFC4458] defines a "cause" URI parameter as having predefined values for Redirecting reasons as a mapping from ITU-T Q.732.2-5 Redirecting Reasons. The "cause" URI parameter is to be used in SIP or SIPs URI. In particular, it may appear in the History-Info header defined in [RFC7044] that MUST be added in retargeted requests. This specification creates a new predefined value for cases when the retargeting is caused by a specific service action leading to a called number translation. This document updates [RFC4458].

Session Initiation Protocol (SIP) Cause URI parameter for Service Number translation Marianne Mohali 0000-00-00
[RFC4458] defines a "cause" URI parameter as having predefined values for Redirecting reasons as a mapping from ITU-T Q.732.2-5 Redirecting Reasons. The "cause" URI parameter is to be used in SIP or SIPs URI. In particular, it may appear in the History-Info header field defined in [RFC7044] that must be added in retargeted requests. This specification creates a new predefined value for cases when the retargeting is caused by a specific service action leading to a called service access number translation. This document updates [RFC4458].

Telecommunications Relay Service Purpose for the Call-Info Header Field in the Session Initiation Protocol (SIP) Paul Kyzivat 0000-00-00
This document defines and registers a value of "original-identity" for the "purpose" header field parameter of the Call-Info header field in the Session Initiation Protocol (SIP).

Remote Call Control and Call Pick-up in SIP Anton Tveretin 0000-00-00
This memo defines a mechanism by which a SIP user agent could inspect calls at another user agent, and control a call, including picking up for itself.

SIP Authentication using the EC-SRP5 Protocol Fuwen liu, Minpeng Qi, Min Zuo 0000-00-00
This document specifies the way that the elliptic curve secure remote protocol (EC-SRP) is applied to SIP authentication. SIP Client and server perform mutual authenticate by using the modern \'zero knowledge\' method without disclosing the password in the process. It has low computation complexity and low bandwidth consumption due to the use of elliptical curve cryptography. This makes it more suitable for resource-constrained environments,e.g. wireless network. The security of the scheme is based on the computational intractability of the elliptic curve discrete logarithm problem. It is resilient to various kinds of attacks, including off-line dictionary attacks.

A SIP Usage for RELOAD Cullen Jennings, Bruce Lowekamp, Eric Rescorla, Salman Baset, Henning Schulzrinne, Thomas Schmidt 0000-00-00
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 and includes a lookup service for Address of Records (AORs) stored in the overlay. It also defines Globally Routable User Agent Uris (GRUUs) that allow the registrations to map an AOR to a specific node reachable through the overlay. After such initial contact of a peer, the AppAttach method is used to establish a direct connection between nodes through which SIP messages are exchanged.

Considerations for Information Services and Operator Services Using SIP John Haluska, Richard Ahern, Marty Cruze, Chris Blackwell 0000-00-00
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 describe how current operator services can continue to be provided to PSTN based subscribers via a SIP based operator services architecture. It also looks at how current operator services might be provided to SIP based subscribers via such an architecture, but does not consider the larger question of the need for or usefulness or suitability of each of these services for SIP based subscribers. This document addresses the needs of current Operator and Information Services providers; as such, the intended audience includes vendors of equipment and services to such providers.

Locating Session Initiation Protocol (SIP) Servers in a Dual-Stack IP Network oej, Gonzalo Salgueiro, Vijay Gurbani 0000-00-00
RFC 3263 defines how a Session Initiation Protocol (SIP) implementation, given a SIP Uniform Resource Identifier (URI), should locate the next hop SIP server using Domain Name System (DNS) procedures. As SIP networks increasingly transition from IPv4-only to dual-stack, a quality user experience must be ensured for dual- stack SIP implementations. This document supplements the DNS procedures described in RFC 3263 for dual-stack SIP implementations and ensures that they properly align to the optimizations detailed by Happy Eyeballs.

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