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IETF Drafts

SIP Call-Info Parameters for Labeling Calls

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.

Best Practices for Securing RTP Media Signaled with SIP

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.

Happy Earballs: Success with Dual-Stack SIP

TBD: 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 network interfaces with multiple address families. In these cases sending a message from a dual stack client to a dual stack server may suffer from the issues described in [RFC6555] (“Happy Eyeballs”): the UA attempts to send the message using IPv6, but IPv6 connectivity is not working to the server. This can cause significant delays in the process of sending the message to the server. This negatively affects the user’s experience. TBD: This document builds on [RFC6555] by modifying the procedures specified in [RFC3263] and related specifications to require that a client ensure that communication targets are accessible before sending messages to them, to allow a client to contact targets out of the order required by other specifications, and to require a client to properly distribute the message load among targets over time.

A P-Served-User Header Field Parameter for Originating CDIV session case in Session Initiation Protocol (SIP)

This specification defines a new parameter of the P-Served-User header field in the Session Initiation Protocol (SIP). This new “orig-cdiv” parameter 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 to convey the identity of the served user and the session case that applies to this particular communication session and application invocation. This document updates RFC5502 to add the “originating after CDIV” session case and to provide more guidance for using the P-Served-User header field in IP networks that were missing in RFC5502.

Authenticated Identity Management in the Session Initiation Protocol (SIP)

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.

Considerations for Information Services and Operator Services Using SIP

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.

Location Source Parameter for the SIP Geolocation Header Field

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.

ISUP Cause Location Parameter for the SIP Reason Header Field

The SIP Reason header field is defined for carrying ISUP cause values as well as SIP response codes. Some services in SIP networks may need to know the ISUP location where the call was released in the PSTN network to correctly interpret the reason of release.

Location Parameter for the SIP Reason Header Field

The SIP Reason header field is defined for Q.850 cause values. Some services in SIP networks may need to know the location where the call was released in the PSTN network to correctly interpret the reason of release.

Third-Party Authentication for Session Initiation Protocol (SIP)

This document defines an authentication mechanism for SIP, that is based on the OAuth 2.0 and OpenID Connect Core 1.0 specifications, to enable the delegation of the user authentication to a dedicated third-party IdP entity that is separate from the SIP network elements that provide the SIP service.