ECRITInternet Engineering Task Force (IETF) H. SchulzrinneInternet-DraftRequest for Comments: 7090 Columbia UniversityIntended status:Category: Standards Track H. TschofenigExpires: April 17, 2014 Nokia Solutions and NetworksISSN: 2070-1721 C. Holmberg Ericsson M. PatelInterDigital Communications October 14, 2013Huawei Technologies (UK) Co., Ltd. April 2014 Public Safety Answering Point (PSAP) Callbackdraft-ietf-ecrit-psap-callback-13.txtAbstract After an emergency call is completed(either(terminated either prematurelyterminatedby the emergency caller or normally by the calltaker) it is possible thattaker), the call takerfeelsmay feel the need for further communication. For example, the call may have been dropped by accident without the call taker having sufficient information about the currentsituationstate ofa wounded person.an accident victim. A call taker may trigger a callbacktowardsto the emergency caller using the contact information provided with the initial emergency call. This callback could, under certain circumstances, be treated like any other callandand, as aconsequenceconsequence, it may get blocked by authorization policies or may get forwarded to an answering machine. The IETF emergency services architecture specification already offers a solution approach for allowingPSAPPublic Safety Answering Point (PSAP) callbacks to bypass authorization policies in order to reach the caller without unnecessary delays. Unfortunately, the specified mechanism only supports limited scenarios. This document discusses shortcomings of the current mechanisms and illustrates additional scenarios wherebetter-than- normalbetter-than-normal call treatment behavior would be desirable.AWe describe a solution based on a new header fieldvalue, called "psap-callback",value for the SIP Priority headerfield is specifiedfield, called "psap-callback", toaccomplish themark PSAPcallback marking.callbacks. Status of This Memo ThisInternet-Draftissubmitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documentsan Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF).Note that other groups may also distribute working documents as Internet-Drafts. The listIt represents the consensus ofcurrent Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents validthe IETF community. It has received public review and has been approved fora maximumpublication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 5741. Information about the current status ofsix monthsthis document, any errata, and how to provide feedback on it may beupdated, replaced, or obsoleted by other documentsobtained atany 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 April 17, 2014.http://www.rfc-editor.org/info/rfc7090. Copyright Notice Copyright (c)20132014 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....................................................4 2. Terminology. . . . . . . . . . . . . . . . . . . . . . . . . 4.....................................................5 3. Callback Scenarios. . . . . . . . . . . . . . . . . . . . . 4..............................................5 3.1. Routing Asymmetry. . . . . . . . . . . . . . . . . . . . 5..........................................6 3.2. Multi-Stage Routing. . . . . . . . . . . . . . . . . . . 5........................................7 3.3. Call Forwarding. . . . . . . . . . . . . . . . . . . . . 6............................................8 3.4.Network-basedNetwork-Based Service URN Resolution. . . . . . . . . . 8......................10 3.5. PSTN Interworking. . . . . . . . . . . . . . . . . . . . 9.........................................11 4. SIP PSAP Callback Indicator. . . . . . . . . . . . . . . . . 10....................................12 4.1. General. . . . . . . . . . . . . . . . . . . . . . . . . 10...................................................12 4.2. Usage. . . . . . . . . . . . . . . . . . . . . . . . . . 10.....................................................12 4.3. Syntax. . . . . . . . . . . . . . . . . . . . . . . . . 10....................................................12 4.3.1. General. . . . . . . . . . . . . . . . . . . . . . . 10............................................12 4.3.2. ABNF. . . . . . . . . . . . . . . . . . . . . . . . 10...............................................12 5. Security Considerations. . . . . . . . . . . . . . . . . . . 10........................................12 5.1. Security Threat. . . . . . . . . . . . . . . . . . . . . 10...........................................12 5.2. Security Requirements. . . . . . . . . . . . . . . . . . 11.....................................13 5.3. Security Solution. . . . . . . . . . . . . . . . . . . . 11.........................................13 6. IANA Considerations. . . . . . . . . . . . . . . . . . . . . 13............................................15 7. Acknowledgements. . . . . . . . . . . . . . . . . . . . . . 13...............................................16 8. References. . . . . . . . . . . . . . . . . . . . . . . . . 14.....................................................16 8.1. Normative References. . . . . . . . . . . . . . . . . . 14......................................16 8.2. Informative References. . . . . . . . . . . . . . . . . 14....................................16 1. Introduction Summoning police, the firedepartmentdepartment, or an ambulance in emergencies is one of the fundamental andmost-valuedmost valuable functions of the telephone. As telephone functionality moves from circuit-switched telephony to Internet telephony, its users rightfully expect that this core functionality will continue to work at least as well as it has for the legacy technology. New devices and services are being made available that could be used to make a request forhelp, whichhelp and that are not traditionaltelephones, and userstelephones. Users are increasingly expecting them to be used to place emergency calls. An overview of the protocol interactions for emergency calling using the IETF emergency services architectureareis described in[RFC6443][RFC6443], and [RFC6881] specifies the technical details. As part of the emergency call setupprocedureprocedure, two important identifiers are conveyed to the PSAP call taker's user agent, namely theAddress-Of-Recordaddress-of- record (AOR),and,and if available, the Globally Routable User Agent (UA) URIs(GRUU).(GRUUs). RFC 3261 [RFC3261] defines the AOR as:"AnAn address-of-record (AOR) is a SIP or SIPS URI that points to a domain with a location service that can map the URI to another URI where the user might be available. Typically, the location service is populated through registrations. An AOR is frequently thought of as the "public address" of theuser."user. In SIPsystemssystems, a single user can have a number of user agents (handsets, softphones, voicemail accounts, etc.)whichthat are all referenced by the same AOR. There are a number of cases in which it is desirable to have an identifierwhichthat addresses a single user agent rather than the group of user agents indicated by an AOR. The GRUU is such a unique user-agent identifier,whichand it isstillalso globally routable.RFC 5627[RFC5627] specifies how to obtain and use GRUUs. [RFC6881] also makes use of the GRUU for emergency calls. Regulatory requirements demand that the emergency call setup procedure itself provides enough information to allow the call taker to initiate a callback to the emergency caller. This is desirable in those cases where the callgotis dropped prematurely or when further communicationneed arises.needs arise. The AOR and the GRUU serve this purpose. The communication attempt by the PSAP call taker back to the emergency caller is called'PSAP callback'.a "PSAP callback". A PSAP callback may, however, be blocked byuser configureduser-configured authorization policies or may be forwarded to an answering machine since SIP entities (SIP proxies as well as the SIP user equipment itself) cannot differentiate the PSAP callback from any other SIP call. "Call barring", "do not disturb", or "calldiversion"(akadiversion" (also called call forwarding) are features that prevent delivery of a call. It is important to note that these features may be implemented by SIP intermediaries as well as by the user agent. Among the emergency servicescommunitycommunity, there isthea desire tooffertreat PSAP callbacksa treatmentin such a way that the chancesare increased that it reachesof reaching the emergencycaller.caller are increased. At the sametime a designtime, any solution mustdeal withminimize thenegative side-effects of allowing certainchance that other callstobypass call forwarding or other authorization policies. Ideally, the PSAP callback has to relate to an earlier emergency call that was made "not too long ago". An exact time interval is difficult to define in a global IETF standard due to the variety of national regulatoryrequirementsrequirements, but [RFC6881] suggests 30 minutes.To neverthelessNevertheless, to meet the needs from the emergency servicescommunitycommunity, a basic mechanism for preferential treatment of PSAP callbacks was defined in Section 13 of [RFC6443]. The specification says:"AA UA may be able to determine a PSAP callback by examining the domain of incoming calls after placing an emergency call and comparing that to the domain of the answering PSAP from the emergency call. Any call from the same domain and directed to the supplied Contact header or AOR after an emergency call should be accepted as a callback from the PSAP if it occurs within a reasonable time after an emergency call wasplaced."placed. This approach mimics a statefulpacket filteringpacket-filtering firewall and is indeed helpful in a number of cases. It is also relatively simple to implement even though it requires call state to be maintained by the user agent as well as by SIP intermediaries. Unfortunately, the solution does not work in all deployment scenarios. In Section 3 we describe cases where the currently standardized approach is insufficient. 2. TerminologyEmergency services relatedEmergency-services-related terminology is borrowed from [RFC5012]. This includes terminology like emergency caller, user equipment, call taker, Emergency Service Routing Proxy (ESRP), and Public Safety Answering Point (PSAP). 3. Callback Scenarios This section illustrates a number of scenarios where the currently specified solution, asspecifieddescribed in [RFC6881], for preferential treatment of callbacks fails. As explained in Section11, a SIP entity examines an incoming PSAP callback by comparing the domain of the PSAP with the destination domain of the outbound emergency call placed earlier. 3.1. Routing Asymmetry In some deploymentenvironmentsenvironments, it is common to have incoming and outgoing SIP messaging routed through different SIP entities. Figure 1 shows this graphically whereby aVoIPVoice over IP (VoIP) provider uses different SIP proxies for inbound and for outbound call handling. Unless the two devices are synchronized, the callbackhittingreaching the inbound proxy would get treated like any other call since the emergency call established state information at the outbound proxy only. ,-------. ,' `. ,-------. / Emergency \ ,' `. | Services | / VoIP \ I | Network | | Provider | n | | | | t | | | | e | | | +-------+ | r | | +--+---|Inbound|<--+-----m | | | | |Proxy | | e | +------+ | | | +-------+ | d | |PSAP | | | | | i | +--+---+ | +----+ | | | a-+ | | | | UA |<---+ | | t | | | | | |----+ | | e | | | | +----+ | | | | | | | | | | P | | | | | | | r | | | | | | +--------+ | o | | | | +--+-->|Outbound|--+---->v | | +--+---+ | | |Proxy | | i | | +-+ESRP | | | +--------+ | d | | | +------+ | | | e||| | | | | | r|+-++----+-+ | \ / | | `. ,' \ / '-------' `. ,' '-------' Figure 1: Example for RoutingAsymmetry.Asymmetry 3.2. Multi-Stage Routing Consider thefollowingemergency call routing scenario shown in Figure 2 where routing towards the PSAP occurs in several stages. In thisscenarioscenario, we consider a SIP UA that uses the Location-to-Service TranslationProtocol(LoST) Protocol [RFC5222] to learn thenext hopnext-hop destination, namely esrp@example.net, to get the call closer to the PSAP. This call is then sent to the proxy of the user's VoIP provider (example.org). The user's VoIP provider receives the emergency call and creates a state based on the destination domain, namely example.net. It then routesitthe call to the indicated ESRP. When the ESRP receivesitthe call, it needs to decide what the next hop is to get to the final PSAP. In ourexampleexample, the next hop is the PSAP with the URI psap@example.com. When a callback is sent from psap@example.com towards the emergencycallercaller, the call will get normal treatment by the proxy of the VoIP provider since the domain of the PSAP does not match the stored state information. ,-----------. +----+ ,' `. | UA |--- esrp@example.net / Emergency \ +----+ \ | Services | \ ,-------. | Network | ,' `. | | / VoIP \ | +------+ | ( Provider ) | | PSAP | | \ example.org / | +--+---+ | `. ,' | | | '---+---' | | | | | psap@example.com | esrp@example.net | | | | | | | | | | | | | +--+---+ | +------------+-----+ ESRP | | | +------+ | | | \ / `. ,' '----------' Figure 2: Example for Multi-StageRouting.Routing 3.3. Call Forwarding Imagine the following case where an emergency call enters an emergency network (state.example) via anESRPESRP, but then it gets forwarded to a different emergency services network (in ourexampleexample, to example.net,example.orgexample.org, or example.com). The same considerations apply when the police, fireandand, ambulance networks are part of the state.examplesub-domainssubdomains (e.g., police.state.example). Similar to the previousscenario the problem here is withscenario, the wrong state information is beingestablishedset up during the emergency call setup procedure. A callback would originate in the example.net,example.orgexample.org, or example.com domains whereas the emergency caller's SIP UA or the VoIP outbound proxy has stored state.example. ,-------. ,' `. / Emergency \ | Services | | Network | |(state.example)| | | | | | +------+ | | |PSAP +--+ | | +--+---+ | | | | | | | | | | | | | | | | | | | | | | | +--+---+ | | ------------------+---+ESRP | | | esrp-a@state.org | +------+ | | | | | | Call Fwd | | | +-+-+---+ | \ | | | / `. | | | ,' '-|-|-|-' ,-------. Police | | | Fire ,' `. +------------+ | +----+ / Emergency \ ,-------. | | | | Services | ,' `. | | | | Network | / Emergency \ | Ambulance | | (Fire) | | Services | | | | | | | Network | | +----+ | | +------+ | | (Police) | | ,-------. | +----+---+PSAP | | | | | ,' `. | | +------+ | | +------+ | | / Emergency \ | | | | |PSAP +----+--+ | Services | | | example.com , | +------+ | | Network | | `~~~~~~~~~~~~~~~ | | | (Ambulance) | | | example.net , | | | `~~~~~~~~~~~~~~~ | +------+ | | | |PSAP +----+ + | +------+ | | | | example.org , `~~~~~~~~~~~~~~~ Figure 3: Example for CallForwarding.Forwarding 3.4.Network-basedNetwork-Based Service URN Resolution The IETF emergency services architecture also considers cases where the resolution from the Service URN to the PSAP URI does not only happen at the SIP UA itself but at intermediate SIP entities, such as the user's VoIP provider. Figure 4 shows this message exchange of the outgoing emergency call and the incoming PSAP graphically. While the state information stored at the VoIP provider iscorrectcorrect, the state allocated at the SIP UA is not. ,-------. ,' `. / Emergency \ | Services | | Network | | example.com | | | | +------+ |InviteINVITE to police@example.com | |PSAP +<---+------------------------+ | | +----+--------------------+ ^ | +------+|Invite|INVITE from | | | ,police@example.com | | `~~~~~~~~~~~~~~~ | | v | +--------+ Query with location +--+---+-+ | | + urn:service:sos | VoIP | | LoST |<-----------------------|Service | | Server | police@example.com |Provider| | |----------------------->| | +--------+ +--------+ | ^Invite|INVITE| |InviteINVITE from| | to police@example.com| | urn:service:sos V | +-------+ | SIP | | UA | | Alice | +-------+ Figure 4: Example forNetwork-basedNetwork-Based Service URNResolution.Resolution 3.5. PSTN Interworking In case an emergency call enters thePSTN,Public Switched Telephone Network (PSTN), as shown in Figure 5, there is no guarantee that the callbacksome timesometime later leaves the same PSTN/VoIP gateway or that the sameend pointendpoint identifier is used in the forward as well as in the backward direction making it difficult to reliably detect PSAP callbacks. +-----------+ | PSTN |-------------+ | Calltaker | | | Bob |<--------+ | +-----------+ | v ------------------- //// \\\\ +------------+ | | |PSTN / VoIP | | PSTN |---->|Gateway | \\\\ //// | | ------------------- +----+-------+ ^ | | | +-------------+ | +--------+ | | | |VoIP | | PSTN / VoIP | +->|Service | | Gateway | |Provider| ||<------Invite----||<------INVITE----| Y | +-------------+ +--------+ | ^ | |Invite InviteINVITE INVITE | | V | +-------+ | SIP | | UA | | Alice | +-------+ Figure 5: Example for PSTNInterworking.Interworking Note: This scenario is considered outside the scope of this document. The specified solution does not support this use case. 4. SIP PSAP Callback Indicator 4.1. General This section defines a new header field value, called "psap- callback", for the SIP Priority header field defined in [RFC3261]. The value is used to inform SIP entities that the request is associated with a PSAP callback SIP session. 4.2. Usage SIP entities that receive the header field value within an initial request for a SIP session can, depending on local policies, apply PSAPcallback specificcallback-specific procedures for the session or request. The PSAPcallback specificcallback-specific procedures may be applied by SIP-based network entities and by the callee. The specificproceduresactions taken when receivingsucha call marked as a PSAP callback marked call, such as bypassing services and barring procedures, are outside the scope of this document. 4.3. Syntax 4.3.1. General This section defines the ABNF [RFC5234] for the new SIP Priority header field value "psap-callback". 4.3.2. ABNF priority-value /= "psap-callback" Figure 6: ABNF 5. Security Considerations 5.1. Security Threat The PSAP callback functionality described in this document allows marked calls to bypassblacklists,blacklists and ignorecall forwardingcall-forwarding procedures and other similar features used to raise the attention of emergency callers when attempting to contact them. In the case where the SIP Priority header value,'psap-callback',"psap-callback", is supported by the SIP UA, it would overrideuser interfaceuser-interface configurations, such as vibrate-only mode, to alert the caller of the incoming call. 5.2. Security Requirements The security threat discussed in Section 5.1 leads to the requirement to ensure that the mechanisms described in this documentcan notcannot be used for malicious purposes, including telemarketing. Furthermore, if the newly defined extension is not recognized, not verified adequately, or not obeyed by SIP intermediaries or SIPendpointsendpoints, then it must not lead to a failure of the call handling procedure. Such a call must be treated like a call that does not have any marking attached. The indicator described in Section 4 can be inserted by any SIP entity, including attackers. So it is critical that the indicator only lead to preferential call treatment in cases where the recipient has some trust in the caller, as described in the next section. 5.3. Security Solution The approach for dealing withimplementingthe implementation of the security requirements described in Section 5.2 can be differentiated between the behavior applied by the UA and by SIP proxies. A UA that has made an emergency call MUST keep state information so that it can recognize andacceptedaccept a callback from the PSAP if it occurs within a reasonable time after an emergency call was placed, as described in Section 13 of [RFC6443]. Only a timer started at the time when the original emergency call has ended is required; information about the calling party identity is not needed since the callback may use a different calling party identity, as described in Section 3. Since these SIP UA considerations are described already in [RFC6443] as well as in [RFC6881] the rest of this section focuses on the behavior of SIP proxies. Figure 7 shows the architecture that utilizes the identity of the PSAP to decide whether a preferential treatment of callbacks should be provided. To make this policy decision, the identity of the PSAP (i.e., calling party identity) is compared with a PSAPs white list. +----------+ | List of |+ | valid || | PSAPs || +----------+| +----------+ * * white list * V Incoming +----------+ Normal SIP Msg | SIP |+ Treatment -------------->| Entity ||======================> + Identity | ||(if not in white list) Info +----------+| +----------+ || || || Preferential || Treatment ++========================> (if successfully verified) Figure 7:Identity-basedIdentity-Based Authorization The identity assurance in SIP can come in different forms, namely via the SIP Identity [RFC4474] or the P-Asserted-Identity [RFC3325] mechanisms. The former technique relies on a cryptographic assurance and the latter on a chain of trust.AlsoAlso, the usage ofTLSTransport Layer Security (TLS) between neighboring SIP entities may provide useful identity information. At the time ofwritingwriting, these identity technologies are being revised in the Secure Telephone Identity Revisited (stir) working group [STIR] to offer better support for legacy technologies interworking and SIP intermediaries that modify the content of various SIP headers and the body. Once the work on these specifications has beencompletedcompleted, they will offer a stronger calling party identity mechanism that limits or prevents identity spoofing. An important aspect from a security point of view is the relationship between the emergency services network (containing the PSAPs) and the VoIPprovider (assumingprovider, assuming that the emergency call travels via the VoIP provider and not directly between the SIP UA and thePSAP).PSAP. The establishment of a white list with PSAP identities may be operationally complex and dependent on the relationship between the emergency services operator and the VoIP provider.WhenIf there is a relationship between the VoIP provider and the PSAP operator, forexampleexample, when they are both operating in the same geographical region, then populating the white list is fairly simple and consequently the identification of a PSAP callback is less problematic compared to the case where the two entities have never interacted with each other before. In the end, the VoIP provider has to verify whether the marked callback message indeed came from a legitimate source. VoIP providers MUST only give PSAP callbacks preferential treatment when the calling party identity of the PSAP was successfully matched against entries in the white list. If it cannot be verified (because there was nomatch),thenmatch), then the VoIP provider MUST remove the PSAP callback marking. Thereby, the callbackis degeneratedreverts to a normal call. As a second step, SIP UAs MUST maintain a timer that is started with the original emergency call and this timer expires within a reasonable amount of time, such as 30 minutes per [RFC6881]. Such a timer also ensures that VoIP providers cannot misuse the PSAP callback mechanism, forexampleexample, to ensure that their support callsreachesreach their customers. Finally, a PSAP callback MUST use the same media as the original emergency call. For example, when an initial emergency call established a real-time text communicationsessionsession, then the PSAP callback must also attempt to establish a real-time communication interaction. The reason for this istwo-fold.twofold. First, the person seekingforhelp may have disabilities that prevent them from using certain media and hence using the same media for the callback avoids unpleasant surprises and delays. Second, the emergency caller may have intentionally chosen a certain media and does not prefer to communicate in a different way. For example, it would be unfortunate if a hostage tries to seekforhelp using instant messaging to avoid any noise when subsequently thering-toneringtone triggered by a PSAP callback using a voice call gets the attention of the hostage-taker.UserUser- interface designs need to cater to such situations. 6. IANA Considerations This document adds the "psap-callback" value to the SIPPriority header IANA"Priority Header Field Values" registry allocated by [RFC6878]. The semantic of the newly defined "psap-callback" value is defined in Section 4. 7. Acknowledgements We would like to thank the following persons for their feedback:Paul Kyzivat, Martin Thomson, Robert Sparks, Keith Drage, Cullen Jennings Brian Rosen, Martin Dolly,Bernard Aboba, Andrew Allen,Atle Monrad,John-Luc Bakker,John Elwell, Geoff Thompson, Dan Romascanu, James Polk, John Medland, Hadriel Kaplan,Kenneth Carlberg, Martin Dolly, Keith Drage, Timothy Dwight, John Elwell, JanetGunnGunn, Cullen Jennings, Hadriel Kaplan, Paul Kyzivat, John Medland, Atle Monrad, James Polk, Dan Romascanu, Brian Rosen, Robert Sparks, Geoff Thompson, and Martin Thomson. We would also like to thank the ECRIT working group chairs, Marc Linsner and Roger Marshall, for their support. Roger Marshall was the document shepherd for this document. Vijay Gurbani provided the general area review. During IESGreviewreview, the document received good feedback from Barry Leiba, Spencer Dawkins, Richard Barnes, Joel Jaeggli, Stephen Farrell, and Benoit Claise. 8. References 8.1. Normative 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. [RFC5234] Crocker, D., Ed., and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, January 2008. [RFC5627] Rosenberg, J., "Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP)", RFC 5627, October 2009. [RFC6878] Roach, A., "IANA Registry for the Session Initiation Protocol (SIP) "Priority" Header Field", RFC 6878, March 2013. 8.2. Informative References [RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", RFC 3325, November 2002. [RFC4474] Peterson, J. and C. Jennings, "Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)", RFC 4474, August 2006. [RFC5012] Schulzrinne, H. and R. Marshall, "Requirements for Emergency Context Resolution with Internet Technologies", RFC 5012, January 2008. [RFC5222] Hardie, T., Newton, A., Schulzrinne, H., and H. Tschofenig, "LoST: A Location-to-Service Translation Protocol", RFC 5222, August 2008. [RFC6443] Rosen, B., Schulzrinne, H., Polk, J., and A. Newton, "Framework for Emergency Calling Using Internet Multimedia", RFC 6443, December 2011. [RFC6881] Rosen, B. and J. Polk, "Best Current Practice for Communications Services in Support of Emergency Calling", BCP 181, RFC 6881, March 2013. [STIR] IETF, "Secure Telephone Identity Revisited (stir) Working Group",URL:http://datatracker.ietf.org/wg/stir/charter/,OctOctober 2013. Authors' Addresses Henning Schulzrinne Columbia University Department of Computer Science 450 Computer Science Building New York, NY 10027 US Phone: +1 212 939 7004 EMail: hgs+ecrit@cs.columbia.edu URI: http://www.cs.columbia.edu Hannes TschofenigNokia Solutions and Networks Linnoitustie 6 Espoo 02600 Finland Phone: +358 (50) 4871445EMail: Hannes.Tschofenig@gmx.net URI: http://www.tschofenig.priv.at Christer Holmberg Ericsson Hirsalantie 11 Jorvas 02420 Finland EMail: christer.holmberg@ericsson.com Milan PatelInterDigital CommunicationsHuawei Technologies (UK) Co., Ltd. 300 South Oak Way, Green Park Reading, Berkshire RG2 6UF U.K. EMail:Milan.Patel@interdigital.comMilan.Patel@huawei.com