PSTN interop

This memo describes use cases, requirements, and protocol extensions for using the Session Description Protocol (SDP) offer/answer model for establishing audio and video media streams over circuit-switched bearers in the Public Switched Telephone Network (PSTN).

This document lists the essential requirements for real-time Text-over-IP (ToIP) and defines a framework for implementation of all required functions based on the Session Initiation Protocol (SIP) and the Real-Time Transport Protocol (RTP). This includes interworking between Text-over-IP and existing text telephony on the Public Switched Telephone Network (PSTN) and other networks. This memo provides information for the Internet community.

This document describes a private Session Initiation Protocol (SIP) header field (P-header) to be used by the European Telecommunications Standards Institute (ETSI) Telecommunications and Internet-converged Services and Protocols for Advanced Networks (TISPAN) for the purpose of authorizing early media flows in Third Generation Partnership Project (3GPP) IP Multimedia Subsystems (IMS). This header field is useful in any SIP network that is interconnected with other SIP networks and needs to control the flow of media in the early dialog state. This memo provides information for the Internet community.

This document describes a standardized mechanism to convey trunk group parameters in sip and tel Uniform Resource Identifiers (URIs). An extension to the tel URI is defined for this purpose. [STANDARDS-TRACK]

This document registers the Enumservice type "pstn" and subtype "tel" using the URI scheme 'tel', as well as the subtype "sip" using the URI scheme 'sip' as per the IANA registration process defined in the ENUM specification, RFC 3761. This Enumservice is used to facilitate the routing of telephone calls in those countries where number portability exists. [STANDARDS-TRACK]

Without a tel URI parameter to carry an encoding type of Integrated Services Digital Network (ISDN) subaddress, interworking between ISDN User Part (ISUP) network and a Session Initiation Protocol (SIP) network is impossible in some cases. To solve this problem, this document specifies a new optional tel URI parameter to carry the encoding type of ISDN subaddress. This memo provides information for the Internet community.

Public Switched Telephone Network (PSTN) services such as 800-number routing (freephone), time-and-day routing, credit-card calling, and virtual private network (mapping a private network number into a public number) are realized by the Intelligent Network (IN). This document addresses means to support existing IN services from Session Initiation Protocol (SIP) endpoints for an IP-host-to-phone call. The call request is originated on a SIP endpoint, but the services to the call are provided by the data and procedures resident in the PSTN/IN. To provide IN services in a transparent manner to SIP endpoints, this document describes the mechanism for interworking SIP and Intelligent Network Application Part (INAP). This memo provides information for the Internet community.

This document contains best current practice examples of Session Initiation Protocol (SIP) call flows showing interworking with the Public Switched Telephone Network (PSTN). Elements in these call flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways. Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP. PSTN telephony protocols are illustrated using ISDN (Integrated Services Digital Network), ISUP (ISDN User Part), and FGB (Feature Group B) circuit associated signaling. PSTN calls are illustrated using global telephone numbers from the PSTN and private extensions served on by a PBX (Private Branch Exchange). Call flow diagrams and message details are shown.

The Layer 2 Tunneling Protocol (L2TP) defines a mechanism for tunneling Point-to-Point Protocol (PPP) sessions. It is common for these PPP sessions to be established using modems connected over the public switched telephone network. One of the standards governing modem operation defines procedures that enable a client modem to put the call on hold and later, re-establish the modem link with minimal delay and without having to redial. While the modem call is on hold, the client phone line can be used to place or receive other calls. The L2TP base protocol does not provide any means to signal these events from the L2TP Access Controller (LAC), where the modem is physically connected, to the L2TP Network Server (LNS), where the PPP session is handled. This document describes a method to let the LNS know when a client modem connected to a LAC has placed the call on hold. [STANDARDS-TRACK]

This document describes MIME types for application/ISUP and application/QSIG objects for use in SIP applications, according to the rules defined in RFC 2048. These types can be used to identify ISUP and QSIG objects within a SIP message such as INVITE or INFO, as might be implemented when using SIP in an environment where part of the call involves interworking to the PSTN. [STANDARDS-TRACK]

This memo describes a simple method of encoding Global Switched Telephone Network (GSTN) addresses (commonly called "telephone numbers") in the local-part of Internet email addresses, along with an extension mechanism to allow encoding of additional standard attributes needed for email gateways to GSTN-based services. [STANDARDS-TRACK]

This RFC presents a true telephone service, called INETPhone, which supports voice communication through the Internet. This memo provides information for the Internet community. It does not specify an Internet standard of any kind.