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Network Working Group                                           B. Aboba
INTERNET-DRAFT                                                M. Thomson
Category: Informational                                            Skype
Expires: December 30, 2013                                  13 June 2013

                  Emergency Services Support in WebRTC
                    draft-aboba-rtcweb-ecrit-01.txt

Abstract

   The Web Real-Time Communication (WebRTC) framework supports
   interactive communication between web-browsers, including support for
   audio, video and text.  This document describes how emergency
   services functionality can be implemented within the WebRTC
   framework, including support for location and call routing as well as
   interoperability with Public Safety Answering Points (PSAPs)
   supporting next generation emergency services.

Legal

   THIS DOCUMENT AND THE INFORMATION CONTAINED THEREIN ARE PROVIDED ON
   AN "AS IS" BASIS AND THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
   REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE
   IETF TRUST, AND THE INTERNET ENGINEERING TASK FORCE, DISCLAIM ALL
   WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY
   WARRANTY THAT THE USE OF THE INFORMATION THEREIN WILL NOT INFRINGE
   ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS
   FOR A PARTICULAR PURPOSE.

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 December 30, 2013.






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Copyright Notice

   Copyright (c) 2013 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
   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 . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1  Terminology . . . . . . . . . . . . . . . . . . . . . . .  3
     1.2  Prior Work  . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Location and Call Routing Requirements . . . . . . . . . . .  4
   3   Media Requirements . . . . . . . . . . . . . . . . . . . . .  7
   4.  Accessibility  . . . . . . . . . . . . . . . . . . . . . . .  8
   5.  Security Considerations  . . . . . . . . . . . . . . . . . .  9
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . .  11
   7.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . .  11
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . .  11
     8.1. Normative References  . . . . . . . . . . . . . . . . . .  11
     8.2  Informative references  . . . . . . . . . . . . . . . . .  11
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . .  15





















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1.  Introduction

   The Web Real-Time Communication (WebRTC) framework supports
   interactive communication between web-browsers, including support for
   audio, video and text.  This document describes how emergency
   services functionality can be implemented within the WebRTC
   framework.  Since signaling is out of scope of the WebRTC standards
   suite as noted in "Overview: Real Time Protocols for Browser-based
   Applications" [I-D.ietf-rtcweb-overview] Section 3, this document
   focuses on other aspects such as location, call routing and media
   support.

   No guidance is provided as to whether a given WebRTC application or
   service will be subject to emergency service obligations.  As noted
   in "Best Current Practice for Communications Services in support of
   Emergency Calling" [RFC6881] Section 4:

      Some jurisdictions have regulations governing which devices need
      to support emergency calling and developers are encouraged to
      ensure that devices they develop meet relevant regulatory
      requirements.  Unfortunately, the natural variation in those
      regulations also makes it impossible to accurately describe the
      cases when developers do or do not have to support emergency
      calling.

   It should also be understood that this document does not advocate use
   of IP-based communication in all situations.  For example, where
   accurate location cannot be obtained, emergency callers could be
   better served by utilizing the telephony capabilities of the
   underlying platform (e.g., a mobile-device) where available, as
   proposed in [WebTel].  This can enable location to be provided in
   situations where it would not otherwise be available, as well as
   permitting an emergency call to be placed even when the device does
   not have access to the Internet.

   The document is laid out as follows: Section 1 provides an
   introduction and reviews prior work.  Section 2 discusses
   requirements relating to location and call routing.  Section 3
   discusses media requirements.  Section 4 discusses accessibility.
   Section 5 discusses security considerations.

1.1.  Terminology

   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 [RFC2119].

   This document uses terms from [RFC3261], [RFC5012] and [RFC6443].



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1.2.  Prior Work

   The IETF ECRIT WG has developed an overview of the emergency calling
   architecture as well as a best current practice document detailing
   implementation requirements.

   "Framework for Emergency Calling using Internet Multimedia" [RFC6443]
   provides an overview of how IETF specifications can be used to
   support emergency calling using multimedia.  At a high level, this
   involves determination of the caller location, conveyance of the
   location within a signaling protocol such as Session Initiation
   Protocol (SIP) [RFC6442], routing of the call using the Location-to-
   Service Translation (LoST) protocol [RFC5222], and exchange of media
   using Real-time Transport Protocol (RTP) [RFC3550].

   "Best Current Practice for Communication Services in support of
   Emergency Calling" [RFC6881] builds on [RFC6443] to describe the
   requirements for end devices ("ED-" requirements), access networks
   ("AN-"), service providers ("SP-"), Public Safety Answering Points
   (PSAPs) and intermediate devices ("INT-") to achieve globally
   interoperable emergency calling on the Internet.

   Both [RFC6443] and [RFC6881] assume the use of SIP as the signaling
   mechanism for emergency calling.  As noted in [RFC6443] Section 1:

      This document discusses the use of the Session Initiation Protocol
      (SIP) [RFC3261] by PSAPs and calling parties.  While other inter-
      domain call signaling protocols may be used for emergency calling,
      SIP is ubiquitous and possesses the proper support of this use
      case.

   Since standardization of signaling is out of scope of the WebRTC
   standards effort, and WebRTC applications can utilize a wide variety
   of signaling mechanisms, the requirements described in [RFC6881] do
   not necessarily apply to WebRTC implementations, applications and
   services.  Therefore in this document, we focus on emergency calling
   requirements that are independent of the signaling mechanism, such as
   those relating to accessibility, location, call routing and media.

2.  Location and Call Routing Requirements

   Determination of caller location as well as call routing is an
   essential aspect of emergency services support.  Relevant
   requirements from [RFC6881] include:

      ED-15/INT-4/AN-4 Devices, intermediate Devices and/or access
      networks SHOULD support a manual method to override the location
      the access network determines.  When the override location is



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      supplied in civic form, it MUST be possible for the resultant
      Presence Information Data Format - Location Object (PIDF-LO)
      received at the PSAP to contain any of the elements specified in
      [RFC4119] and [RFC5139].

      ED-17/INT-9/AN-9 Devices that support endpoint measuring of
      location MUST have at least a coarse location capability
      (typically <1km accuracy) for routing of calls.  The location
      mechanism MAY be a service provided by the access network.

      ED-24 Where the operating system supporting application programs
      which need location for emergency calls does not allow access to
      Layer 2 and Layer 3 functions necessary for a client application
      to use DHCP location options and/or other location technologies
      that are specific to the type of access network, the operating
      system MUST provide a published API conforming to ED-12 through
      ED-23 and ED-25 through ED-32.  It is RECOMMENDED that all
      operating systems provide such an API.

      ED-41/SP-20 Location objects MUST be created with information
      about the method by which the location was determined, such as
      GPS, manually entered, or based on access network topology
      included in a PIDF- LO "method" element.  In addition, the source
      of the location information MUST be included in a PIDF-LO
      "provided-by" element.

      ED-49 Endpoints MUST support one or more mechanisms that allow
      them to determine their public IP address, for example, STUN
      [RFC5389].

      ED-50 Endpoints MUST support LIS discovery as described in
      [RFC5986], and the LoST discovery as described in [RFC5223].

   Since browser applications do not have direct access to operating
   system location APIs, ED-24 is not applicable to WebRTC.

   For reasons that will be described, automatically obtaining location
   suitable for emergency use is challenging for WebRTC applications.
   In order to ensure that location is available when needed, as well as
   to provide resilience against errors in automated location
   determination, WebRTC emergency service applications SHOULD support
   manual override as recommended in ED-15.

   The W3C Geolocation API [GeolocationAPI] was not developed with
   emergency services location in mind, so that requirements ED-17 and
   ED-41 are not well supported.  [GeolocationAPI] does not provide
   information on the source of the location information as required in
   ED-41; attempting to infer the source from the accuracy parameter is



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   NOT RECOMMENDED.  Currently, Location Based Services utilized by
   Geolocation APIs do not warrant their use in emergency services and
   do not consistently provide the accuracy required by emergency
   services applications, so that emergency use of the W3C Geolocation
   API is also NOT RECOMMENDED.

   An alternative is to implement location configuration and call
   routing in Javascript, using an HTTP-based protocol such as HELD
   [RFC5985] and LoST [RFC5222].  While this approach can provide
   location usable in emergency services applications, it is only
   applicable on networks with a Location Information Server (LIS), such
   as enterprise deployments subject to Multi-Line Telephone System
   (MLTS) regulations [StateMLTS].

   In order to utilize location and call routing services, it is first
   necessary to locate the appropriate servers.  Since the discovery
   mechanisms described in [RFC5986] and [RFC5223] are based on use of a
   DHCP option, which cannot be assumed to be accessible in Javascript,
   ED-50 is difficult to support within WebRTC-based emergency services
   applications.

   For LoST discovery, the emergency services application can determine
   the appropriate LoST server(s) on its own.  To avoid potential
   issues, it is best to avoid pre-configuration of particular servers,
   allowing the appropriate server to be determined dynamically.

   LIS discovery requires determination of the domain name that can be
   used for LIS discovery, as noted in [RFC5986] Section 3.4:

      If a Device knows one or more alternative domain names that might
      be used for discovery, it MAY repeat the U-NAPTR process using
      those domain names as input.  For instance, static configuration
      of a Device might be used to provide a Device with a domain name.

   While static configuration of the domain name can be used in
   situations where device mobility is restricted, the appropriate LIS
   depends on the network to which the host is attached, so that this is
   not a general solution.

   "Location Information Server (LIS) Discovery using IP address and
   Reverse DNS" [I.D.ietf-geopriv-res-gw-lis-discovery] specifies a
   means for a device to discover several alternative domain names that
   can be used as input to the Dynamic Delegation Discovery Service
   (DDDS).  Since several of the techniques (such as use of PTR RRs and
   Session Traversal Utilities for NAT (STUN) [RFC5389]) are potentially
   implementable in WebRTC-based emergency services applications this
   approach MAY be used.




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3.  Media Requirements

   Within [RFC6881] media-related requirements are covered in Section
   14.  These include:

      ED-71 Endpoints MUST send and receive media streams on RTP
      [RFC3550].

      ED-72 Normal SIP offer/answer [RFC3264] negotiations MUST be used
      to agree on the media streams to be used.

      ED-73/SP-41 G.711 A law (and mu Law if they are intended be used
      in North America) encoded voice as described in [RFC3551] MUST be
      supported.  If the endpoint cannot support G.711, a transcoder
      MUST be used so that the offer received at the PSAP contains
      G.711.  It is desirable to include wideband codecs such as G.722
      and AMR-WB in the offer.  PSAPs SHOULD support narrowband codecs
      common on endpoints in their area to avoid transcoding.

      ED-74 Silence suppression (Voice Activity Detection methods) MUST
      NOT be used on emergency calls.  PSAP call takers sometimes get
      information on what is happening in the background to determine
      how to process the call.

      ED-77 Endpoints supporting video MUST support H.264 per [RFC6184].

   Requirement ED-71 is satisfied by compliant WebRTC implementations
   since "Web Real-Time Communication (WebRTC): Media Transport and Use
   of RTP" [I-D.ietf-rtcweb-rtp-usage] Section 4.1 requires support for
   RTP [RFC3550].

   Requirement ED-72 is specific to SIP and so does not apply generally
   to WebRTC implementations, applications and services.  However, it is
   believed that the APIs under development within the W3C WebRTC WG can
   support this requirement.

   Requirement ED-74 is satisfied by compliant WebRTC implementations
   since the WebRTC APIs under development within W3C [WEBRTC], support
   silence suppression control via the "constraints" parameter.

   [I.D.ietf-rtcweb-rtp-usage] Section 4.3 does not provide a
   recommendation on a mandatory-to-implement set of codecs.  While
   ED-73 does not require implementation of G.711 if the service
   supports transcoding, G.711 is not difficult to implement and is
   widely supported, with a high level of interoperability.  Therefore
   it is recommended that G.711 be included as a mandatory-to-implement
   audio codec within [I-D.ietf-rtcweb-rtp-usage] Section 4.3.




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   Currently the disposition of ED-77 is unclear.  Discussion of
   mandatory-to-implement video codecs is ongoing within the IETF RTCWEB
   WG, but has not reached a conclusion.  While there is a need to
   support interoperable video within emergency services applications,
   more options may be available within an emergency services context
   than would be the case for general use.  For example, within the
   PSAP, it may be feasible to support multiple video codecs, either by
   installation of browser plugins, or by use of multiple browsers.  In
   some emergency service applications (such as the VRS), codec
   requirements may be specific to the service and may be satisfiable by
   a custom device or browser approved for use with that service, which
   may include the required codecs implemented natively or via plug-ins,
   as the service provider sees fit.

4.  Accessibility

   By lowering the barriers to development of realtime-enabled browser
   applications, as well as by building on accessibility support within
   the browser, WebRTC promises to enable the development of a new
   generation of accessible emergency applications and services.

   In order to support accessibility, it is RECOMMENDED that WebRTC-
   based emergency applications and services conform to the Web Content
   Accessibility Guidelines (WCAG) v2.0 [WCAG].

   In order to support accessibility for individuals with hearing or
   speech disabilities, support for textual communications is important.

   Currently the W3C is developing a proposed charter for the Timed Text
   Working Group [TTWG], which will potentially produce a second edition
   of the timed Text Markup Language (TTML) 1.0 recommendation as well
   as publishing a recommendation for a version 1.1 specification.

   Text-related requirements in [RFC6881] are covered in Section 14,
   including:

      ED-75 Endpoints supporting Instant Messaging (IM) MUST support
      either [RFC3428] and [RFC4975].

      ED-76 Endpoints supporting real-time text MUST use [RFC4103].  The
      expectations for emergency service support for the real-time text
      medium are described in [RFC5194], Section 7.1.

   Since [RFC3428] and [RFC4975] are both based on SIP, ED-75 does not
   apply to all WebRTC-based emergency applications and services.  As
   noted in "Emergency Services Functionality with the Extensible
   Messaging and Presence Protocol (XMPP)" [I-D.tschofenig-ecrit-xmpp-
   es], XMPP [RFC6120] is a potential alternative for emergency services



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   applications looking to support instant messaging [RFC6121] and
   multi-user chat [XEP-045] functionality.

   "RTP Payload for Text Conversation" [RFC4103] is typically
   implemented along with SIP signaling as described in "Framework for
   Real-Time Text over IP Using the Session Initiation Protocol (SIP)"
   [RFC5194].  As a result, ED-76 does not apply to WebRTC
   implementations.

   Alternatives to support of real-time text functionality are
   available, such as "In-Band Real Time Text" [XEP-0301], which
   supports real-time text by addition of child elements within XMPP
   message stanzas.  The use of child elements to encapsulate real-time
   text, as well as transmission of complete lines enables [XEP-0301] to
   provide backward compatibility with existing XMPP instant-messaging
   and Multi-User Chat (MUC) clients, with no changes required to XMPP
   servers.  Since XMPP can be encapsulated within HTTP via mechanisms
   such as BOSH [XEP-0206] or WebSockets [RFC6455], [XEP-0301] can be
   implemented in Javascript.  Experience with Javascript implementation
   using the [Strophe] XMPP library indicates that adequate performance
   is achievable.  In contrast, implementing real-time text as media as
   in [RFC4103] requires native browser support, as well as requiring
   changes to the configuration of intermediaries such as Session Border
   Controllers (SBCs).  Also, [RFC4103] is not backward compatible with
   SIP instant messaging implementations supporting page-mode [RFC3428]
   or session [RFC4975] approaches.

5.  Security Considerations

   Security requirements in [RFC6881] include:

      ED-48/SP-24 TLS [RFC5746] MUST be used to protect location (but
      see Section 9.1).  All implementations MUST support TLS.

      ED-58/SP-30 TLS is the primary mechanism used to secure the
      signaling for emergency calls.  IPsec [RFC4301] MAY be used
      instead of TLS for any hop.  Either TLS or IPSEC MUST be used when
      attempting to signal an emergency call.

      ED-59/SP-31 If TLS session establishment is not available, or
      fails, the call MUST be retried without TLS.

      ED-60/SP-32 [RFC5626] is RECOMMENDED to maintain persistent TLS
      connections between entities when one of the entity is an
      endpoint.  Persistent TLS connection between proxies is
      RECOMMENDED using any suitable mechanism.

      ED-61/AN-28 TLS SHOULD be used when attempting to retrieve



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      location (configuration or dereferencing) with HELD.  The use of
      [RFC5077] is RECOMMENDED to minimize the time to establish TLS
      sessions without keeping server-side state.  IPsec MAY be used
      instead of TLS.

      ED-62/AN-29 When TLS session establishment fails, the location
      retrieval MUST be retried without TLS.

   For WebRTC, HTTPS MUST be used to protect signaling for an emergency
   call, with potential fail-over to HTTP.  HTTPS SHOULD be used to
   protect location retrieval (HELD) and call routing (LoST).

   WebRTC security considerations are discussed in "Security
   Considerations for RTC-Web" [I-D.ietf-rtcweb-security].  The WebRTC
   security architecture, described in "RTCWEB Security Architecture"
   [I-D.ietf-rtcweb-security-arch], requires implementation of Secure
   RTP [RFC3711] as well as DTLS/SRTP [RFC5764].

   While the security features of WebRTC exceed the requirements
   outlined in [RFC6881], support for emergency services within WebRTC
   raises concerns about potential attacks on the emergency services
   infrastructure, given the potential for malicious code to be executed
   within the browser.  One way to lessen the likelihood of attacks by
   untrusted Javascript applications is for PSAPs to put up their own
   sites for emergency calling, protected by HTTPS.

   While ICE [RFC5245] provides demonstration of liveness and consent to
   receive, it is possible for an attacker to overwhelm the PSAP by
   generating a large number of prank calls.  IP relay services are also
   potential targets since these don't require forging of Caller-Id nor
   do they provide audio or video from the attacker.

   Security threats to IP-based emergency services are described in
   "Security Threats and Requirements for Emergency Call Marking and
   Mapping" [RFC5069].  These include attacks on the emergency services
   system, such as attempting to deny system services to all users in a
   given area, to gain fraudulent use of services and to divert
   emergency calls to non-emergency sites.  [RFC5069] also describes
   attacks against individuals, including attempts to prevent an
   individual from receiving aid, or to gain information about an
   emergency.

   "Threat Analysis of the Geopriv Protocol" [RFC3694] describes threats
   against geographic location privacy, including protocol threats,
   threats resulting from the storage of geographic location data, and
   threats posed by the abuse of information.

   Overall, experience indicates a relationship between anonymity and



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   the prevalence of prank calling.  Therefore some protection may be
   provided through authentication of the caller either in the signaling
   or media plane.  It is NOT RECOMMENDED that WebRTC-based emergency
   applications and services support anonymous emergency calling.

6.  IANA Considerations

   This document does not require actions by IANA.

7.  Acknowledgments

   We would like to thank the members of the IETF RTCWEB Working Group
   for discussions related to this topic.

8.  References

8.1.  Normative References

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

[RFC6881]   Rosen, B. and J. Polk, "Best Current Practice for
            Communications Services in Support of Emergency Calling",
            RFC 6881, March 2013.

[WCAG]      Caldwell, B., Cooper, M., Reid, L.G. and G. Vanderheiden,
            "Web Content Accessibility Guidelines (WCAG) 2.0",
            http://www.w3.org/TR/WCAG20/, December 2008.

8.2.  Informative References

[GeolocationAPI]
            Popescu, A., "Geolocation API Specification", W3C,
            http://dev.w3.org/geo/api/spec-source.html

[I.D.ietf-geopriv-res-gw-lis-discovery]
            Thomson, M. and R. Bellis, "Location Information Server
            (LIS) Discovery using IP address and Reverse DNS", draft-
            ietf-geopriv-res-gw-lis-discovery-05 (work in progress),
            April 2013.

[I-D.ietf-rtcweb-overview]
            Alvestrand, H., "Overview: Real Time Protocols for Browser-
            based Applications", draft-ietf-rtcweb-overview-06 (work in
            progress), February 2013.

[I-D.ietf-rtcweb-rtp-usage]
            Perkins, C., Westerlund, M. and J. Ott, "Web Real-Time



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            Communication (WebRTC): Media Transport and Use of RTP",
            draft-ietf-rtcweb-rtp-usage-06 (work in progress), February
            2013.

[I-D.ietf-rtcweb-security]
            Rescorla, E., "Security Considerations for RTC-Web", draft-
            ietf-rtcweb-security-04 (work in progress), January 2013.

[I-D.ietf-rtcweb-security-arch]
            Rescorla, E., "RTCWEB Security Architecture", draft-ietf-
            rtcweb-security-arch-06 (work in progress), July 2013.

[I-D.tschofenig-ecrit-xmpp-es]
            Tschofenig. H., "Emergency Services Functionality with the
            Extensible Messaging and Presence Protocol (XMPP)", draft-
            tschofenig-ecrit-xmpp-es-00 (work in progress), March 2012.

[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 R. Schulzrinne, "An Offer/Answer Model
            with the Session Description Protocol (SDP)", RFC 3264, June
            2002.

[RFC3428]   Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C.
            and D. Gurle, "Session Initiation Protocol (SIP) Extension
            for Instant Messaging", RFC 3428, December 2002.

[RFC3550]   Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, July 2003.

[RFC3551]   Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            July 2003.

[RFC3694]   Danley, M., Mulligan, D., Morris, J. and J. Peterson,
            "Threat Analysis of the Geopriv Protocol", RFC 3694,
            February 2004.

[RFC3711]   Baugher, M., McGrew, D., Naslund, M., Carrara, E. and K.
            Norrman, "The Secure Real-time Transport Protocol (SRTP)",
            RFC 3711, March 2004.

[RFC4103]   Hellstrom, G. and P. Jones, "RTP Payload for Text
            Conversation", RFC 4103, June 2005.




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[RFC4119]   Peterson, J., "A Presence-based GEOPRIV Location Object
            Format", RFC 4119, December 2005.

[RFC4301]   Kent, S. and K. Seo, "Security Architecture for the Internet
            Protocol", RFC 4301, December 2005.

[RFC4975]   Campbell, B., Mahy, R. and C. Jennings, "The Message Session
            Relay Protocol (MSRP)", RFC 4975, September 2007.

[RFC5012]   Schulzrinne, H. and R. Marshall, "Requirements for Emergency
            Context Resolution with Internet Technologies", RFC 5012,
            January 2008.

[RFC5069]   Taylor, T., Tschofenig, H., Schulzrinne, H. and M.
            Shanmugam, "Security Trheats and Requirements for Emergency
            Call Marking and Mapping", RFC 5069, January 2008.

[RFC5077]   Salowey, J., Zhou, H., Eronen, P. and H. Tschofenig,
            "Transport Layer Security (TLS)  Session Resumption without
            Server-Side State", RFC 5077, January 2008.

[RFC5139]   Thomson, M. and J. Winterbottom, "Revised Civic Location
            Format for Presence Information Data Format Location Object
            (PIDF-LO)", RFC 5139, February 2008.

[RFC5194]   van Wijk, A. and G. Gybels, "Framework for Real-Time Text
            over IP Using the Session Initiation Protocol (SIP)", RFC
            5194, June 2008.

[RFC5222]   Hardie, T., Newton, A., Schulzrinne, H. and H. Tschofenig,
            "LoST: A Location-to-Service Translation Protocol", RFC
            5222, August 2008.

[RFC5223]   Schulzrinne, H., Polk, J. and H. Tschofenig, "Discovering
            Location-to-Service Translation (LoST) Servers Using the
            Dynamic Host Configuration Protocol (DHCP)", RFC 5223,
            August 2008.

[RFC5245]   Rosenberg, J., "Interactive Connectivity Establishment
            (ICE): A Protocol for Network Address Translator (NAT)
            Traversal for Offer/Answer Protocols", RFC 5245, April 2010.

[RFC5389]   Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
            Traversal Utilities for NAT", RFC 5389, October 2008.

[RFC5626]   Jennings, C., Mahy, R. and F. Audet, "Managing Client-
            Initiated Connections in the Session Initiation Protocol
            (SIP)", RFC 5626, October 2009.



Aboba & Thomson               Informational                    [Page 13]


INTERNET-DRAFT    Emergency Services Support in WebRTC      13 June 2013


[RFC5746]   Rescorla, E., Ray, M., Dispensa, S. and N. Oskov, "Transport
            Layer Security (TLS) Renegotiation Indication Extension",
            RFC 5746, February 2010.

[RFC5764]   McGrew, D. and E. Rescorla, "Datagram Transport Layer
            Security (DTLS) Extension to Establish Keys for the Secure
            Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

[RFC5985]   Barnes, M., "HTTP Enabled Location Delivery (HELD)", RFC
            5985, September 2010.

[RFC5986]   Thomson, M. and J. Winterbottom, "Discovering the Local
            Location Information Server (LIS)", RFC 5986, September
            2010.

[RFC6120]   Saint-Andre, P., "Extensible Messaging and Presence Protocol
            (XMPP): Core", RFC 6120, March 2011.

[RFC6121]   Saint-Andre, P., "Extensible Messaging and Presence Protocol
            (XMPP): Instant Messaging and Presence", RFC 6121, March
            2011.

[RFC6184]   Wang, Y.-K., Even, R., Kristensen, T. and R. Jesup, "RTP
            Payload Format for H.264 Video", RFC 6184, May 2011.

[RFC6442]   Polk, J., Rosen, B. and J. Peterson, "Location Conveyance
            for the Session Initiation Protocol", RFC 6442, December
            2011.

[RFC6443]   Rosen, B., Schulzrinne. H., Polk, J. and A. Newton,
            "Framework for Emergency Calling Using Internet Multimedia",
            RFC 6443, December 2011.

[RFC6455]   Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC
            6455, December 2011.

[StateMLTS] "State E911 Legislation",
            http://www1.911enable.com/resource-center/state-
            e911-legislation

[Strophe]   "Libraries for XMPP Poets", http://strophe.im

[TTWG]      "Proposed Timed Text Working Group Charter",
            http://www.w3.org/2012/02/timed-text-wg-charter.html

[WEBRTC]    Bergkvist, A., Burnett, D., Jennings, C. and A. Narayanan,
            "WebRTC 1.0: Real-time Communication Between Browsers", W3C
            Editor's Draft (work in progress),



Aboba & Thomson               Informational                    [Page 14]


INTERNET-DRAFT    Emergency Services Support in WebRTC      13 June 2013


            http://dev.w3.org/2011/webrtc/editor/webrtc.html, June 2013.

[WebTel]    WebAPI/WebTelephony,
            https://wiki.mozilla.org/WebAPI/WebTelephony

[XEP-0206]  Paterson, I. and P. Saint-Andre, "XMPP Over BOSH", XEP-0206
            version 1.3, http://xmpp.org/extensions/xep-0206.html, July
            2010.

[XEP-0301]  Rejhon, M., "In-Band Real Time Text", XEP-0301 version 0.2,
            http://xmpp.org/extensions/xep-0301.html, March 2012.

[XEP-045]   Saint-Andre, P., "Multi-User Chat", XEP 0045 version 1.25,
            http://xmpp.org/extensions/xep-0045.html, February 2012.

Authors' Addresses

   Bernard Aboba
   Skype
   Redmond, WA  98052
   US

   EMail:  bernard_aboba@hotmail.com

   Martin Thomson
   Skype
   3210 Porter Drive
   Palo Alto, CA  94304
   US

   Phone: +1 650-353-1925
   Email: martin.thomson@gmail.com



















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