The requirement for direct transcoding with Session Initiation Protocol
draft-kang-sipping-dtransc-requirement-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Author | Taegyu Kang | ||
| Last updated | 2005-06-08 | ||
| 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This Internet-Draft is no longer active.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 presents a set of Session Initiation Protocol (SIP) call control requirements that support communication with direct transcoding capability. Several solutions are addressed for transcoding service. Direct transcoding requires two kinds of requirements: the service requirement and call control requirement. Service requirement is no constraint of codec adaptation and interoperability of models. Call control requirement is exchange of codec information and minimizing of call set up delay. The model might be applied to general-purpose services satisfying the requirements of multimedia applications without an additional INVITE.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)