RTCWeb Working Group                                       H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                         U. Rauschenbach
Expires: April 25, 2015                                   Nokia Networks
                                                        October 22, 2014

                            WebRTC Gateways


   This document specifies conformance requirements for a class of
   WebRTC-compatible endpoints called "WebRTC gateways", which
   interconnect between WebRTC endpoints and devices that are not
   WebRTC endpoints.

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

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Table of Contents

   1.  Introduction
     1.1.  Implications of the gateway environment
     1.2.  Signalling model
   2.  WebRTC device requirements that can be relaxed
   3.  Additional WebRTC gateway requirements
   4.  IANA Considerations
   5.  Security Considerations
   6.  Acknowledgements
   7.  Change history
   8.  Normative References
   Authors' Addresses

1.  Introduction

   The WebRTC model described in [I-D.ietf-rtcweb-overview] is focused
   on direct browser to browser communication as its primary use case.
   Nevertheless, it is clearly interesting to have WebRTC endpoints
   connect to other types of devices, including but not limited to SIP
   phones, legacy phones, CLUE-based teleconferencing systems, XMPP-
   based conferencing systems, and entirely proprietary devices or

   WebRTC gateways are a specific type of WebRTC-compatible end points
   which enable the exchange of media streams between WebRTC endpoints
   on one side, and the other types of devices mentioned above on the
   other side.

   This document describes the requirements that need to be placed on
   such gateways, both the requirements on WebRTC endpoints that can be
   relaxed and the additional requirements that need to be applied.

   A WebRTC gateway is a WebRTC-compatible endpoint, and will thus not
   be conformant with all requirements for a WebRTC endpoint (it does
   not do everything a WebRTC endpoint does), but is able to
   interoperate with WebRTC endpoints.

1.1.  Implications of the gateway environment

   A gateway will be limited in the functionality it can offer by the
   system or class of devices it is gatewaying to.  For instance, a
   gateway into the telephone system will not be able to relay data or
   video, no matter how much it is required.  Therefore, a number of
   functions that are mandatory to support in WebRTC endpoints are not
   mandatory on gateways; the requirement on the gateway is that it is
   able to negotiate those features away correctly.

1.2.  Signalling model

   The WebRTC model is that signalling is outside the scope of the
   specification.  This document does not change that.

   Nevertheless, any practical gateway needs to deal with signalling.
   For that, this document assumes that the overall system consists of
   an application running in the WebRTC browser, possibly one or more
   signalling relays that mediate signalling and thereby enable
   communication between the application and the gateway, and the actual
   gateway that is responsible for handling the media flows.

   The application, the signalling relays (if any) and the gateway
   together need to be able to:

   o  adhere to the offer/answer semantics

   o  deal with the description of configuration coming from the
      browser; this is specified in SDP format in the WebRTC browser API

   o  generate the information that is needed by the browser to set up
      the session, and express that information in the form of SDP.

   The shorthand notation "The gateway MUST/SHOULD/MAY support <SDP
   function xxx>" used below means that an application running in the
   Web browser, the signalling relays, and the gateway together
   MUST/SHOULD/MAY support this functionality; it is not a requirement
   that this happens at the media gateway itself.

2.  WebRTC device requirements that can be relaxed

   WebRTC gateways are intended to communicate with WebRTC endpoints.
   WebRTC gateways are no User Agents.  They are therefore expected to
   conform to the requirements for WebRTC devices in [I-D.ietf-rtcweb-
   overview], with the exceptions defined in this section.

   Since a gateway is expected to be deployed where it can be reached
   with a static IP address (as seen from the client), a WebRTC gateway
   does not need to support full ICE; it therefore MAY implement ICE-
   Lite only.

   ICE-Lite implementations do not send consent checks, so a gateway MAY
   choose not to send consent checks too, but MUST respond to consent
   checks it receives.

   A gateway is expected to not need to hide its location, so it does
   not need to support functionality for operating only via a TURN
   server; instead it MAY choose to produce Host ICE candidates only.

   If a gateway serves as a media relay into another RTP domain, it MAY
   choose to support only features available in that network.  This
   means that it MAY not (need to) support Bundle and any of the RTP/
   RTCP extensions related to it, RTCP-Mux, or Trickle Ice. However, the
   gateway MUST support DTLS-SRTP, since this is required for
   interworking with WebRTC endpoints.

   If a gateway serves as a media relay into a network or to devices not
   implementing the WebRTC Datachannel, it MAY choose to not support the

3.  Additional WebRTC gateway requirements

   (nothing yet)

4.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an

5.  Security Considerations

   A WebRTC gateway may operate in two security modes: Security-context
   termination and security-context relaying.

   Relaying is only possible where signed and encrypted content can be
   passed through unchanged, and where keys can be exchanged directly
   between the endpoints.

   When the gateway terminates the security context, it means that the
   WebRTC user has to place trust in the gateway to perform all
   verification of identity and protection of content in the realm on
   the other side of the gateway; there is no way the end-user can
   detect a man-in-the-middle attack, an identity spoofing attack or a
   recording done at the gateway.  For many scenarios, this is not going
   to be seen as a problem, but needs to be considered when one decides
   to use a gatewayed service.

6.  Acknowledgements

   Several comments from Christer Holmberg were included.

7.  Change history

   Changes from draft-holmberg-mmusic-sdp-multiplex-negotiation-00

   o  Aligned terminology with draft-rtcweb-overview-12

   o  Rewrote text on signaling to improve clarity

   o  Editorial nits

8.  Normative References

              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-12
              (work in progress), October 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

Authors' Addresses

   Harald Alvestrand

   Email: harald@alvestrand.no

   Uwe Rauschenbach
   Nokia Networks

   Email: uwe.rauschenbach@nsn.com