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Versions: (draft-alvestrand-rtcweb-gateways) 00 01 02

RTCWeb Working Group                                       H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                         U. Rauschenbach
Expires: July 24, 2016                                    Nokia Networks
                                                        January 21, 2016

                            WebRTC Gateways


   This document describes interoperability considerations for a class
   of WebRTC-compatible endpoints called "WebRTC gateways", which
   interconnect between WebRTC endpoints and devices that are not WebRTC

Requirements Language

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

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on July 24, 2016.

Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of

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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Implications of the gateway environment . . . . . . . . .   3
     1.2.  Signalling model  . . . . . . . . . . . . . . . . . . . .   3
   2.  WebRTC non-browser requirements that can be relaxed . . . . .   4
   3.  Additional WebRTC gateway requirements  . . . . . . . . . . .   4
   4.  Considerations for SDP-using networks . . . . . . . . . . . .   5
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   5
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .   5
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   6
   8.  Change history  . . . . . . . . . . . . . . . . . . . . . . .   6
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   7
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .   7
     9.2.  Informative References  . . . . . . . . . . . . . . . . .   7
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   7

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 middle boxes 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.  To a WebRTC endpoint, the
   gateway appears as a WebRTC-compatible endpoint.

   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 appears as 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.

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   NOTE IN DRAFT: There is still not a WG consensus called on whether
   this document is Informational or standards-track.  If it becomes
   informational, the use of RFC 2119 language is used to call attention
   to features where non-conformance will render a gateway unable to
   interoperate with WebRTC-based 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.

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2.  WebRTC non-browser requirements that can be relaxed

   WebRTC gateways are intended to communicate with WebRTC
   endpoints[I-D.ietf-rtcweb-overview].  Some features that typical
   WebRTC endpoints are required to support may be meaningless or
   unneccesary for WebRTC gateways; some such things are noted in this
   section.  This lack of conformance means that a gateway is considered
   a WebRTC-compatible endpoint, not a WebRTC endpoint (unless a
   particular gateway claims to be a WebRTC endpoint, which it is of
   course allowed to do).

   A WebRTC gateway which is expected to be deployed where it can be
   reached with a static IP address (as seen from the client) 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 with a static IP address 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 choose to not 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.

   Note that non-support of BUNDLE means that "bundle-only" tracks are
   not supported.  This means that applications using an RTCBundlePolicy
   other than "max-compat" ([I-D.ietf-rtcweb-jsep] section 4.1.1) can
   only use one track of each media type.

   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)

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4.  Considerations for SDP-using networks

   Some networks that are gatewayed into, such as SIP networks, will
   also use SDP to represent the media configurations.  Gateways will,
   however, need to inspect and probably modify the SDP passed between
   the SDP-using network and the WebRTC endpoints to achieve maximum

   Considerations include:

   o  If a correspondent does not offer the features WebRTC depends on,
      connections will not complete.  The support for dtls-srtp, shown
      by the "fingerprint" attribute, is the most obvious example.  The
      gateway is probably better off either ending such calls early or
      acting as a full B2BUA (as defined in [RFC3261]) with media

   o  If a correspondent makes an offer using features that are not
      required by JSEP, these may not be understood by the WebRTC
      implementation.  The gateway may choose to strip out some such

   o  Certain ancient practices (such as using port 0 to place a media
      section on hold with the intent of resuming it later) are not
      conformant with the SDP offer/answer spec ([RFC3264] section 8.2).
      Since WebRTC implementations are expected to be SDP offer/answer
      conformant, such practices may need to be stripped out by the

   [NOTE IN DRAFT: This section may need expanding.]

5.  IANA Considerations

   This document makes no request of IANA.

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

6.  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.

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   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.

7.  Acknowledgements

   Several comments from Christer Holmberg and Andrew Hutton were

8.  Change history

   Changes from draft-alvestrand-rtcweb-gateways-00

   o  Aligned terminology with draft-rtcweb-overview-12

   o  Rewrote text on signaling to improve clarity

   o  Editorial nits

   Changes from draft-alvestrand-rtcweb-gateways-01

   o  Aligned terminology with draft-rtcweb-overview-13 ("non-browser")

   o  Nits

   Changes from draft-alvestrand-rtcweb-gateways-02

   o  Re-submitted as WG draft

   o  Addressed a comment from Andrew Hutton that deployment in open
      internet is an option, not a fact.

   Changes from draft-ietf-rtcweb-gateways-00

   o  Added note about impllications of non-support of BUNDLE

   o  Added "Considerations for SDP-using networks" section

   Changes from draft-ietf-rtcweb-gateways-01: None, this is a keepalive

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9.  References

9.1.  Normative References

              Uberti, J., Jennings, C., and E. Rescorla, "Javascript
              Session Establishment Protocol", draft-ietf-rtcweb-jsep-09
              (work in progress), March 2015.

              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-13
              (work in progress), November 2014.

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

9.2.  Informative References

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June

Authors' Addresses

   Harald Alvestrand

   Email: harald@alvestrand.no

   Uwe Rauschenbach
   Nokia Networks

   Email: uwe.rauschenbach@nokia.com

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