Requirements for Session Initiation Protocol (SIP)-based Emergency Calls
draft-schulzrinne-sipping-emergency-req-01
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Author | Henning Schulzrinne | ||
| Last updated | 2004-10-25 | ||
| RFC stream | (None) | ||
| Intended RFC status | (None) | ||
| Formats | |||
| Stream | Stream state | (No stream defined) | |
| Consensus boilerplate | Unknown | ||
| RFC Editor Note | (None) | ||
| IESG | IESG state | Expired | |
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
Abstract
This document enumerates requirements for emergency calls in VoIP and general Internet multimedia systems. We divide the requirements into 'trunk replacement' and 'end-to-end'. Trunking solutions only exchange the emergency call center's circuit-switched access by an IP-based system. The requirements for end-to-end IP-based emergency calling address functional and security issues for determining the correct emergency address, for identifying the appropriate emergency call center and for identifying the caller and its location. While we focus on systems that employ the Session Initiation Protocol (SIP), many of the requirements may also apply to other environments, such as those using H.248/Megaco, MGCP or H.323.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)