Internet DRAFT - draft-ietf-rtcweb-gateways
draft-ietf-rtcweb-gateways
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
draft-ietf-rtcweb-gateways-02
Abstract
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
endpoints.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
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
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Internet-Drafts are draft documents valid for a maximum of six months
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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
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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
systems.
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
Datachannel.
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
interoperability.
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
gatewaying.
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
features.
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
gateway
[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
RFC.
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
included.
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
update.
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9. References
9.1. Normative References
[I-D.ietf-rtcweb-jsep]
Uberti, J., Jennings, C., and E. Rescorla, "Javascript
Session Establishment Protocol", draft-ietf-rtcweb-jsep-09
(work in progress), March 2015.
[I-D.ietf-rtcweb-overview]
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
2002.
Authors' Addresses
Harald Alvestrand
Google
Email: harald@alvestrand.no
Uwe Rauschenbach
Nokia Networks
Email: uwe.rauschenbach@nokia.com
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