Login Form

Lost Password?

No account yet? Register

RECENT IETF DRAFTS

SIP internet drafts statistics

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

Read more ...

Title Author Date
Shared Appearances of a Session Initiation Protocol (SIP) Address of Record (AOR) Alan Johnston, Mohsen Soroushnejad, Venkatesh Venkataramanan 0000-00-00
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. This specification updates RFC3261 and RFC4235.

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.

A Mechanism for Transporting User to User Call Control Information in SIP Alan Johnston, James Rafferty 0000-00-00
There is a class of applications which benefit from using SIP to exchange User to User Information (UUI) data during session establishment. This information, known as call control UUI data, is a small piece of data inserted by an application initiating the session, and utilized by an application accepting the session. The syntax and semantics for the UUI data used by a specific application are defined by a UUI package. This UUI data is opaque to SIP and its function is unrelated to any basic SIP function. This document defines a new SIP header field, User-to-User, to transport UUI data, along with an extension mechanism.

Interworking ISDN Call Control User Information with SIP Keith Drage, Alan Johnston 0000-00-00
The motivation and use cases for interworking and transporting ITU-T DSS1 User to User information element data in SIP are described in RFC 6567. As networks move to SIP, it is important that applications requiring this data can continue to function in SIP networks as well as the ability to interwork with this ISDN service for end-to-end transparency. This document defines a usage (a new package) of the User-to-User header field to enable interworking with this ISDN service. This document covers the interworking with both public ISDN and private ISDN capabilities, so the potential interworking with QSIG will also be addressed. The package is identified by a new value "isdn-uui" of the "purpose" header field parameter.

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).

EDNS Option Code for SIP and PSTN Source Reference Info Hadriel Kaplan, Robert Walter, Pierce Gorman, Manjul Maharishi 0000-00-00
This document requests an IANA allocation for an EDNS0 Option-Code, per [RFC2671], for a UTF-8 encoded string field containing a URI for private use. The intended use of this field is for providing SIP and PSTN-type source information for ENUM-resolution DNS queries, in private DNS server environments such as Private ENUM.

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.

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).

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 0000-00-00
This document discusses how PCP (Port Control Protocol) can be used in SIP deployments in managed networks.

Internet Assigned Numbers Authority (IANA) Registration of the Session Initiation Protocol (SIP) Feature-Capability indicators Andrew Allen 0000-00-00
This document registers with IANA the SIP Feature-Capability indicators in the "SIP Feature-Capability Indicator Registration Tree" of the IANA "Proxy-Feature Feature-Capability Indicator Trees" registry for use with the SIP Feature-Caps header field.

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. The directionality in traffic legs relates to a SIP request creating a dialogue and stand-alone SIP request. 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\', which 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 \'iotl\' parameter is defined in order to fulfil requirements from the 3GPP.

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 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 Protocol (SRTP) security context setup with Datagram Transport Layer Security (DTLS) protocol.

Session Traversal Utilities for NAT (STUN) Message Handling for Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) Ram R, Tirumaleswar Reddy, Gonzalo Salgueiro 0000-00-00
Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) are often designed to be on the media path, rather than just intercepting signaling. This means that B2BUAs often act on the media path leading to separate media legs that the B2BUA correlates and bridges together. When acting on the media path, B2BUAs are likely to receive Session Traversal Utilities for NAT (STUN) packets as part of Interactive Connectivity Establishment (ICE) processing. It is critical that B2BUAs handle these STUN messages properly. This document defines behavior for a B2BUA performing ICE processing.

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.

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].

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.

<< Start < Prev 1 2 Next > End >>
Display # Results 1 - 25 of 38