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RECENT IETF DRAFTS

SIP internet drafts statistics

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

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Title Author Date
End-to-End Session Identification in IP-Based Multimedia Communication Networks Paul Jones, James Polk, Gonzalo Salgueiro, Chris Pearce 0000-00-00
This document describes an end-to-end Session Identifier for use in IP-based multimedia communication systems that enables endpoints, intermediate devices, and management systems to identify a session end-to-end, associate multiple endpoints with a given multipoint conference, track communication sessions when they are redirected, and associate one or more media flows with a given communication session. This document also describes a backwards compatibility mechanism for an existing (RFC 7329) session identifier implementation that is sufficiently different from the procedures defined in this document.

Requirements for Marking SIP Messages to be Logged Peter Dawes, Chidambaram Arunachalam 0000-00-00
SIP networks use signalling 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 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 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 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, Thomas, 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 -05 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.

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.

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).

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.

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 Request-URI of a SIP request. This specification creates a new predefined value for the "cause" URI parameter for cases when the retargeting is due to specific service action leading to a called service access number translation. This document updates [RFC4458].

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.

\'Urgent\' Call Indicator in the Session Initiation Protocol (SIP) Ananda Somadder, Yogendra Agarwal 0000-00-00
This document proposes an extension to the Session Initiation Protocol (SIP). This extension provides the ability for calling SIP User to specify the call urgency while originating a call and for called SIP user agent to identify that call is an \'Urgent\' call.

Improving the Organizational Flexibility of the SIP Change Process. Ben Campbell, Alissa Cooper, Barry Leiba 0000-00-00
RFC 5727 defines several processes for the Real-time Applications and Infrastructure (RAI) area. These processes include the evolution of the Session Initiation Protocol (SIP) and related protocols, as well as the operation of the DISPATCH and SIPCORE working groups. This document updates RFC 5727 to allow flexibility for the area and working group structure, while preserving the SIP change processes. It also generalizes the DISPATCH working group processes so that they can be easily adopted by other working groups.

Location Source Parameter for the SIP Geolocation Header Field James Winterbottom, Laura Liess, Bruno Chatras, Andrew Hutton 0000-00-00
There are some circumstances where a geolocation header field may contain more than one location value. Knowing the identity of the node adding the location value allows the recipient more freedom in selecting the value to look at first rather than relying solely on the order of the location values.

The SIP P-User-Location-Info Private-Header (P-Header) for the 3GPP IP Multimedia (IM) Core Network (CN) Subsystem david.niks@vodafone.com, Andre RameilGreen 0000-00-00
This document specifies the SIP P-User-Location-Info P-header. This header field addresses an issue that was identified when non-3GPP access and non-trusted networks are integrated to IMS (IP Multimedia Subsystem) networks. This header field conveys the trusted network determined location information of a served user when the user is registered for IMS services via non-3GPP access networks.

A Session Initiation Protocol (SIP) Feature Tag for Back-to-Back User Agents (B2BUAs) Ram R, Tirumaleswar Reddy, Gonzalo Salgueiro 0000-00-00
The User Agent capabilities specification allows Session Initiation Protocol (SIP) User Agents to convey their capabilities and characteristics to other User Agents and to the registrar for its domain. This information is conveyed as parameters of the Contact header field. Amongst those capabilities are the type of User Agent that is available at a SIP Uniform Resource Identifier (URI). This document extends the User Agent capabilities specification to allow indication of Back-to-Back User Agent (B2BUA) types.

Concepts and Terminology for Peer to Peer SIP David Bryan, Philip Matthews, Eunsoo Shim, Dean Willis, Spencer Dawkins 0000-00-00
This document defines concepts and terminology for the use of the Session Initiation Protocol in a peer-to-peer environment where the traditional proxy-registrar and message routing functions are replaced by a distributed mechanism. These mechanisms may be implemented using a distributed hash table or other distributed data mechanism with similar external properties. This document includes a high-level view of the functional relationships between the network elements defined herein, a conceptual model of operations, and an outline of the related problems addressed by the P2PSIP working group and the RELOAD protocol and SIP usage ([RFC6940], [I-D.ietf-p2psip-sip]) defined by the working group.

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.

Session Initiation Protocol (SIP) Recording Metadata 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. 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) 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. 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 lists call flows that has snapshot of metadata sent from SRC to SRS. This is purely an informational document that is written to support the model defined in the metadata draft.

Authenticated Identity Management in the Session Initiation Protocol (SIP) Jon Peterson, Cullen Jennings, Eric Rescorla 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 new SIP header fields 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): Groupchat Peter Saint-Andre, Saúl Corretgé, Salvatore Loreto 0000-00-00
This document defines a bidirectional protocol mapping for the exchange of instant messages in the context of a multiparty chat session among users of the Session Initiation Protocol (SIP) and users of the Extensible Messaging and Presence Protocol (XMPP). Specifically, this document defines a mapping between the SIP-based Message Session Relay Protocol (MSRP) and the XMPP Multi-User Chat (MUC) extension.

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