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SIP internet drafts statistics

  • 22 SIP related internet drafts (IETF).
  • 0 new and updated drafts posted in the last 14 days.

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Title Author Date
Session Initiation Protocol (SIP) Cause URI parameter for Service Number translation Marianne Mohali, Mary Barnes 0000-00-00
RFC4458 defines a "cause" URI parameter, which may appear in the Request-URI of a SIP request, that is used to indicate a reason why the request arrived to the User Agent Server (UAS) receiving the message. This specification creates a new predefined value for the "cause" URI parameter to cover service number translation for cases of retargeting due to specific service action leading to the translation of a called service access number. This document updates RFC4458.

Session Initiation Protocol (SIP) Via header field parameter to indicate received realm Christer Holmberg, Yi Jiang 0000-00-00
This specification defines a new Session Initiation Protocol (SIP) Via header field parameter, "received-realm", which allows a SIP entity acting as an entry point to a transit network to indicate from which adjacent upstream network a SIP request is received, using a network realm value associated with the adjacent network.

IANA Registration of New Session Initiation Protocol (SIP) Resource- Priority Namespace for Mission Critical Push To Talk service Christer Holmberg, Joergen Axell 0000-00-00
This document creates an additional Session Initiation Protocol (SIP) Resource-Priority namespace to meet the requirements of the 3GPP defined Mission Critical Push To Talk, and places this namespace in the IANA registry.

Requirements for Marking SIP Messages to be Logged Peter Dawes, Chidambaram Arunachalam 0000-00-00
SIP networks use signaling monitoring tools to debug customer reported problems 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 signaling will take between clients, and therefore impractical to monitor SIP signaling end-to-end. This draft describes requirements for adding an indicator to the SIP protocol data unit (PDU, or a SIP message) that marks the PDU as a candidate for logging. Such marking will typically be applied as part of network testing controlled by the network operator and not used in regular client signaling. 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, Suhas Nandakumar 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, Thomas Stach, 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).

A Session Initiation Protocol (SIP) INFO package for Private Wire Finlay Fraser, Chris Boulton 0000-00-00
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 Session Initiation Protocol (SIP) OAuth Rifaat Shekh-Yusef, Victor Pascual, Christer Holmberg 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.

SIP Authentication using the EC-SRP5 Protocol Fuwen liu, Minpeng Qi, Min Zuo 0000-00-00
This document specifies how 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.

P-Served-User Header Field Parameter for Originating CDIV session case in Session Initiation Protocol (SIP) Marianne Mohali 0000-00-00
This specification defines a new Session Initiation Protocol (SIP) P- Served-User header field parameter, "orig-cdiv-param", which defines the session case used by a proxy when handling an originating session after Call Diversion (CDIV) services has been invoked for the served user. The P-Served-User header field is defined in RFC5502. The P- Served-User header field conveys the identity of the served user and the session case that applies to this particular communication session and application invocation. This document updates RFC5502 in order to add the originating after CDIV session case.

Clarifications for when to use the name-addr production in SIP messages Robert Sparks 0000-00-00
RFC3261 constrained several SIP header fields whose grammar contains the "name-addr / addr-spec" alternative to use name-addr when certain characters appear. Unfortunately it expressed the constraints with prose copied into each header field definition, and at least one header field was missed. Further, the constraint has not been copied into documents defining extension headers whose grammar contains the alternative. This document updates RFC3261 to state the constraint generically, and clarifies that the constraint applies to all SIP header fields where there is a choice between using name-addr or addr-spec. It also updates those extension SIP header fields that use the alternative to clarify that the constraint applies (RFCs 3325, 3515, 3892, 4508, 5002, 5318, 5360, and 5502).

Contacting Session Initiation Protocol (SIP) Servers in a Dual-Stack IP Network Dale Worley 0000-00-00
In a dual-stack (IPv4 and IPv6) environment, the procedures of RFC 3263 by which a Session Initiation Protocol (SIP) client contacts a server may not suffice to provide a good user experience. This document describes "Happy Eyeballs" modifications -- modifications of the procedures of RFC 3263, as well as additional client procedures -- which improve the SIP user experience in many circumstances.

Content ID header field in Session Initiation Protocol (SIP) Christer Holmberg, Ivo Sedlacek 0000-00-00
This document specifies the Content-ID header field for usage in the Session Initiation Protocol (SIP).

Setting up a SIP (Session Initiation Protocol) connection in a dual stack network using connection oriented transports Olle Johansson, Gonzalo Salgueiro, Dale Worley 0000-00-00
The Session Initiation Protocol (SIP) supports multiple transports running both over IPv4 and IPv6 protocols. In more and more cases, a SIP user agent (UA) is connected to multiple network interfaces. In these cases setting up a connection from a dual stack client to a dual stack server may suffer from the issues described in RFC 6555 [RFC6555] - Happy Eyeballs - significant delays in the process of setting up a working flow to a server. This negatively affects user experience. This document builds on RFC 6555 and explains how a RFC3261 [RFC3261] compliant SIP implementation can quickly set up working flows to a given hostname (located by using DNS NAPTR and SRV lookups) in a dual stack network using connection oriented transport protocols. A solution for connectionless transport protocols is discussed in a separate document.

SIP Call-Info Parameters for Labeling Calls Henning Schulzrinne 0000-00-00
Called parties often wish to decide whether to accept, reject or redirect calls based on the likely nature of the call. For example, they may want to reject unwanted telemarketing or fraudulent calls, but accept emergency alerts from numbers not in their address book. This document describes SIP Call-Info parameters and a feature tag that allow originating, intermediate and terminating SIP entities to label calls as to their type, spam probability and references to additional information.

A SIP Response Code for Unwanted Calls Henning Schulzrinne 0000-00-00
This document defines the 666 (Unwanted) SIP response code, allowing called parties to indicate that the call was unwanted. The terminating SIP entity may use this information to adjust future call handling behavior for this called party or more broadly.

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.

Best Practices for Securing RTP Media Signaled with SIP Jon Peterson, Eric Rescorla, Richard Barnes, Russ Housley 0000-00-00
Although the Session Initiation Protocol (SIP) includes a suite of security services that has been expanded by numerous specifications over the years, there is no single place that explains how to use SIP to establish confidential media sessions. Additionally, existing mechanisms have some feature gaps that need to be identified and resolved in order for them to address the pervasive monitoring threat model. This specification describes best practices for negotiating confidential media with SIP, including both comprehensive protection solutions which bind the media to SIP-layer identities as well as opportunistic security solutions.

Session Initiation Protocol (SIP) Recording Call Flows Ram R, Parthasarathi Ravindran, Paul Kyzivat 0000-00-00
Session recording is a critical requirement in many communications environments, such as call centers and financial trading organizations. In some of these environments, all calls must be recorded for regulatory, compliance, and consumer protection reasons. The recording of a session is typically performed by sending a copy of a media stream to a recording device. This document lists call flows with metadata snapshots sent from a Session Recording Client(SRC) to a Session Recording Server(SRS).

Authenticated Identity Management in the Session Initiation Protocol (SIP) Jon Peterson, Cullen Jennings, Eric Rescorla, Chris Wendt 0000-00-00
The baseline security mechanisms in the Session Initiation Protocol (SIP) are inadequate for cryptographically assuring the identity of the end users that originate SIP requests, especially in an interdomain context. This document defines a mechanism for securely identifying originators of SIP requests. It does so by defining a SIP header field for conveying a signature used for validating the identity, and for conveying a reference to the credentials of the signer.

Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence Peter Saint-Andre 0000-00-00
This document defines a bidirectional protocol mapping for the exchange of presence information between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP). This document obsoletes RFC 7248.

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