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IETF Drafts

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.

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.

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.

Clarifications for when to use the name-addr production in SIP messages

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).

A SIP Response Code for Unwanted Calls

This document defines the 666 (Unwanted) SIP response code, allowing called parties to indicate that the call or message was unwanted. SIP entities may use this information to adjust how future calls from this calling party are handled for the called party or more broadly.

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.

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.

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.

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.

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