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Versions: 00 draft-kinamdar-dispatch-sip-auto-peer

Network Working Group                                         K. Inamdar
Internet-Draft                                              S. Narayanan
Intended status: Standards Track                             C. Jennings
Expires: March 13, 2020                                    Cisco Systems
                                                      September 10, 2019


                    Automatic Peering for SIP Trunks
                draft-kinamdar-dispatch-sip-audo-peer-00

Abstract

   This draft specifies a configuration workflow to enable enterprise
   Session Initiation Protocol (SIP) networks to solicit the capability
   set of a SIP service provider network.  The capability set can
   subsequently be used to configure features and services on the
   enterprise edge element, such as a Session Border Controller (SBC),
   to ensure smooth peering between enterprise and service provider
   networks.

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 https://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 March 13, 2020.

Copyright Notice

   Copyright (c) 2019 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
   (https://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



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   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
   2.  Conventions and Terminology . . . . . . . . . . . . . . . . .   3
   3.  Reference Architecture  . . . . . . . . . . . . . . . . . . .   3
   4.  Configuration Workflow  . . . . . . . . . . . . . . . . . . .   5
   5.  Overview of Operations  . . . . . . . . . . . . . . . . . . .   6
   6.  HTTP Transport  . . . . . . . . . . . . . . . . . . . . . . .   7
     6.1.  HTTP Methods  . . . . . . . . . . . . . . . . . . . . . .   7
     6.2.  Integrity and Confidentiality . . . . . . . . . . . . . .   7
     6.3.  Authenticated Client Identity . . . . . . . . . . . . . .   7
     6.4.  Encoding the Request  . . . . . . . . . . . . . . . . . .   8
     6.5.  Generating the Response . . . . . . . . . . . . . . . . .   8
     6.6.  Identifying the Request Target  . . . . . . . . . . . . .   9
   7.  State Deltas  . . . . . . . . . . . . . . . . . . . . . . . .  12
   8.  Encoding the Service Provider Capability Set  . . . . . . . .  12
   9.  Data Model for Capability Set . . . . . . . . . . . . . . . .  12
     9.1.  Tree Diagram  . . . . . . . . . . . . . . . . . . . . . .  12
     9.2.  YANG Model  . . . . . . . . . . . . . . . . . . . . . . .  14
     9.3.  Node Definitions  . . . . . . . . . . . . . . . . . . . .  19
     9.4.  Extending the Capability Set  . . . . . . . . . . . . . .  25
   10. Example Capability Set Document Encoding  . . . . . . . . . .  26
     10.1.  JSON Capability Set Document . . . . . . . . . . . . . .  26
     10.2.  XML Capability Set Document  . . . . . . . . . . . . . .  28
   11. Example Exchange  . . . . . . . . . . . . . . . . . . . . . .  29
   12. Security Considerations . . . . . . . . . . . . . . . . . . .  32
   13. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  33
   14. References  . . . . . . . . . . . . . . . . . . . . . . . . .  33
     14.1.  Normative References . . . . . . . . . . . . . . . . . .  33
     14.2.  Informative References . . . . . . . . . . . . . . . . .  33
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  35

1.  Introduction

   The deployment of a Session Initiation Protocol (SIP)-based
   infrastructure in enterprise and service provider communication
   networks is increasing at a rapid pace.  Consequently, direct IP
   peering between enterprise and service provider networks is quickly
   replacing traditional methods of interconnection between enterprise
   and service provider networks.

   Currently published standards provide a strong foundation over which
   direct IP peering can be realized.  However, given the sheer number
   of these standards, it is often not clear which behavioral subsets,



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   extensions to baseline protocols and operating principles ought to be
   implemented by service provider and enterprise networks to ensure
   successful peering.

   The SIP Connect technical recommendations aim to solve this problem
   by providing a master reference that promotes seamless peering
   between enterprise and service provider SIP networks.  However,
   despite the extensive set of implementation rules and operating
   guidelines, interoperability issues between service provider and
   enterprise networks persist.  This is in large part because service
   providers and equipment manufacturers aren't required to enforce the
   guidelines of the technical specifications and have a fair degree of
   freedom to deviate from them.  Consequently, enterprise
   administrators usually undertake a fairly rigorous regimen of
   testing, analysis and troubleshooting to arrive at a configuration
   block that ensures seamless service provider peering.

   Another set of interoperability problems arise when enterprise
   administrators are required to translate a set of technical
   recommendations from service providers to configuration blocks across
   one or more devices in the enterprise, which is usually an error
   prone exercise.  Additionally, such technical recommendations might
   not be nuanced enough to intuitively allow the generation of specific
   configuration blocks.

   This draft introduces a mechanism using which an enterprise network
   can solicit a detailed capability set from a SIP service provider;
   the detailed capability set can subsequently be used by automaton or
   an administrator to generate configuration blocks across one or more
   devices within the enterprise to ensure successful service provider
   peering.

2.  Conventions and Terminology

   The The keywords "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]

3.  Reference Architecture

   Figure 1 illustrates a reference architecture that may be deployed to
   support the mechanism described in this document.  The enterprise
   network consists of a SIP-PBX, media endpoints and a Session Border
   Controller.  It may also include additional components such as
   application servers for voicemail, recording, fax etc.  At a high
   level, the service provider consists of a SIP signaling entity (SP-
   SSE), a media entity and a HTTP(S) server.




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    +-----------------------------------------------------+
    | +---------------+         +-----------------------+ |
    | |               |         |                       | |
    | | +----------+  |         |   +-------+           | |
    | | |   Cap    |  | HTTPS   |   |       |           | |
    | | |  Server  |<-|---------|-->|       |           | |
    | | |          |  |         |   |       |   +-----+ | |
    | | +----------+  |         |   |       |   | SIP | | |
    | |               |         |   |       |<->| PBX | | |
    | |               |         |   |       |   +-----+ | |
    | | +----------+  |         |   |  SBC  |           | |
    | | |          |  |   SIP   |   |       |           | |
    | | | SP - SSE |<-|---------|-->|       |   +-----+ | |
    | | |          |  |         |   |       |<->| M.E.| | |
    | | +----------+  |         |   |       |   |     | | |
    | |               |         |   |       |   +-----+ | |
    | | +----------+  |  SRTP   |   |       |           | |
    | | |  Media   |<-|---------|-->+-------+           | |
    | | +----------+  |         |                       | |
    | +---------------+         +-----------------------+ |    |                                                     |
    +-----------------------------------------------------+

    Figure 1: Reference Architecture


   This draft makes use of the following terminology:

   o  Enterprise Network: A communications network infrastructure
      deployed by an enterprise which interconnects with the service
      provider network over SIP.  The enterprise network could include
      devices such as application servers, endpoints, call agents and
      edge devices among others.

   o  Edge Device: A device that is the last hop in the enterprise
      network and that is the transit point for traffic entering and
      leaving the enterprise.  An edge device is typically a back-to-
      back user agent (B2BUA) such as a Session Border Controller (SBC).

   o  Service Provider Network: A communications network infrastructure
      deployed by service providers.  In the context of this draft, the
      service provider network is accessible over SIP for the
      establishment, modification and termination of calls and
      accessible over HTTP(S) for the transfer of the capability set
      document.  The service provider network is also referred to as a
      SIP Service Provider (SSP) or Internet Telephony Service Provider
      (ITSP) network.  These networks typically interconnect with other
      service provider networks over SIP or ISDN.




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   o  Call Control: Call Control within a telephony networks refers to
      software that is responsible for delivering its core
      functionality.  Call control not only provides the basic
      functionality of setting up, sustaining and terminating calls, but
      also provides the necessary control and logic required for
      additional services within the telephony network.

   o  Capability Server: A server hosted in the service provider
      network, such that this server is the target for capability set
      document requests from the enterprise network.

   o  Capability Set: This specification uses the term capability set
      (or capability set document) to refer collectively to a set of
      characteristics within the service provider network, which when
      communicated to the enterprise network allows it to obtain
      sufficient information required to peer successfully with the
      service provider network.

4.  Configuration Workflow

   A workflow that facilitates an enterprise network to solicit the
   capability set of a SIP service provider ought to take into account
   the following considerations:

   o  The configuration workflow must be based on a protocol or a set of
      protocols commonly used between enterprise and service provider
      telephony networks.

   o  The configuration workflow must be flexible enough to allow the
      service provider network to dynamically offload different
      capability sets to different enterprise networks based on the
      identity of the enterprise network.

   o  Capability set documents obtained as a result of the configuration
      workflow must be conducive to easy parsing by automaton.
      Subsequently, automaton may be used for generation of appropriate
      configuration blocks.

   Taking the above considerations into account, this document proposes
   a Hypertext Transfer Protocol (HTTP)-based workflow using which the
   enterprise network can solicit and ultimately obtain the service
   provider capability set.  The enterprise network creates a well
   formed HTTPS GET request to solicit the service provider capability
   set.  Subsequently, the HTTPS response from the SIP service provider
   includes the capability set.  The capability set is encoded in either
   XML or JSON, thus ensuring that the response can be easily parsed by
   automaton.




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   There are alternative mechanisms using which the SIP service provider
   can offload its capability set.  For example, the Session Initiation
   Protocol (SIP) can be extended to define a new event package
   [RFC6665], such that the enterprise network can establish a SIP
   subscription with the service provider for its capability set; the
   SIP service provider can subsequently use the SIP NOTIFY request to
   communicate its capability set or any state deltas to its baseline
   capability set.  This mechanism is likely to result in a significant
   amount of operational complexity.  For example, not only would this
   workflow require enterprise and service provider equipment
   manufacturers to upgrade their software stacks, but it would also
   create a significant amount of ambiguity in terms of which device in
   the service provider network handles subscriptions and generates
   notifications.  Yet another example of an alternative mechanism would
   be for service providers and enterprise equipment manufacturers to
   agree on YANG models [RFC6020] that enable configuration to be pushed
   over NETCONF [RFC6241] to enterprise networks from a centralized
   source hosted in service provider networks.  The presence of
   proprietary software logic for call and media handling in enterprise
   devices would preclude the generation of a "one-size-fits-all" YANG
   model.  Additionally, service provider networks pushing configuration
   to enterprises devices might lead to the loss of implementation
   autonomy on the part of the enterprise network.

5.  Overview of Operations

   To solicit the capability set of the SIP service provider, the edge
   element in the enterprise network generates a well-formed HTTPS GET
   request.  There are two reasons why it makes sense for the enterprise
   edge element to generate the HTTPs request:

   1.  Edge elements are devices that normalize any mismatches between
       the enterprise and service provider networks in the media and
       signaling planes.  As a result, when the capability set is
       received from the SIP service provider network, the edge element
       can generate appropriate configuration blocks (possibly across
       multiple devices) to enable smooth IP peering.
   2.  Given that edge elements are configured to "talk" to networks
       external to the enterprise, the complexity in terms of NAT
       traversal and firewall configuration would be minimal.

   The HTTPs GET request is targeted at a capability server that is
   managed by the SIP service provider such that this server processes,
   and on successfully processing the request, includes the capability
   set document in the response.  The capability set document is
   constructed according the guidelines of the YANG model described in
   this draft.  The capability set document included in a successful
   response is formatted in either XML or JSON.  The formatting depends



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   on the value of the "Accept" header field of the HTTP GET request.
   More details about the formatting of the HTTP request and response
   are provided in Section 6

   Figure 1 provides a reference architecture in which this workflow may
   be implemented.  The architecture depicted in Figure 1 consists of an
   enterprise telephony network and a SIP service provider network, such
   that the enterprise network attempts to provision SIP trunking
   services for the first time.  For the sake of simplicity, the
   enterprise and service provider networks are decomposed into their
   core constituent elements.

6.  HTTP Transport

   This section describes the use of HTTP as a transport protocol for
   the peering workflow.  This workflow is based on HTTP version 1.1

6.1.  HTTP Methods

   The workflow defined in this document leverages the HTTP GET method
   and its corresponding response(s) to request for and subsequently
   obtain the service provider capability set document.  The HTTP POST
   method and its corresponding response(s) is also used for client
   authentication.

   To ensure the smooth operation of this workflow, this draft enforces
   certain controls (not to be confused with HTTP controls as defined in
   [RFC7231] on the HTTP client and server.  These controls as they
   relate to formatting, operational guidelines, security concerns and
   more, are detailed in subsequent sections of this draft.

6.2.  Integrity and Confidentiality

   Peering requests and responses are defined over HTTP.  However, due
   to the sensitive nature of information transmitted between client and
   server, it is required to secure HTTP using Transport Layer Security.
   The enterprise edge element and capability server MUST be compliant
   to [RFC7235].  The enterprise edge element and capability server MUST
   support the use of the https uri scheme as defined in [RFC7230].

6.3.  Authenticated Client Identity

   The configuration workflow and corresponding YANG model described in
   this draft allow for smooth IP peering between enterprise and SIP
   service provider networks by encoding the essential session and media
   characteristics.  It is NOT RECOMMENDED to encode information that is
   sensitive in nature.  It is only required for the client (enterprise
   edge element) to authenticate the SIP service provider.  If however,



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   there is a need for the SIP service provider to authenticate the
   enterprise edge element, client authentication mechanisms based on
   OAuth 2.0 token are RECOMMENDED.  The specifics of how the client
   obtains such tokens is outside the scope of this draft.  Section 11
   provides an example of how clients may be authenticated using OAuth
   2.0 tokens.

6.4.  Encoding the Request

   The edge element in the enterprise network generates a HTTPS GET
   request such that the request-target is obtained using the procedure
   outlined in section 6.6 The MIME types for the capability set
   document defined in this draft are "application/peering-info+json"
   and "application/peering-info+xml".  Accordingly, the Accept header
   field value MUST be restricted only to these MIME types.  It is
   possible that the edge element supports responses formatted in both
   JSON and XML.  In such situations, the edge element might generate a
   HTTPS GET request such that the Accept header field includes both
   MIME types along with the corresponding "qvalue" for each MIME type.
   For implementations that require client authentication, the bearer
   access token acquired by the client (see section 6.3) MUST be
   presented to the capability server to the obtain the capability set
   document.  The bearer token is presented in the "Authorization"
   header field of the GET request as specified in Section 2.1 of
   [RFC6750].

   The generated HTTPS GET request MUST NOT use the "Expect" and "Range"
   header fields.  The requests also MUST NOT use any conditional
   request.

6.5.  Generating the Response

   Capability servers include the capability set documents in the body
   of a successful response.  Capability set documents MUST be formatted
   in XML or JSON.  For requests that are incorrectly formatted, the
   capability server SHOULD generate a "400 Bad Request" response.  If
   the client (enterprise edge element) includes any other MIME types in
   Accept header field other than "application/peering-info+json" or
   "application/peering-info+xml", the capability set MUST reject the
   request with a "406 Not Acceptable" response.

   The capability server can respond to client requests with redirect
   responses, specifically, the server can respond with the following
   redirect responses:

   1.  301 Moved Temporarily

   2.  302 Found



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   3.  307 Temporary Redirect

   The server SHOULD include the Location header field in such
   responses.

   For requests that carry an invalid bearer token in the Authorization
   header field, the capability server SHOULD respond with a HTTP 401
   status code.

6.6.  Identifying the Request Target

   HTTPS GET requests from enterprise edge elements MUST carry a valid
   request-target.  The enterprise edge element might obtain the URL of
   the resource hosted on the capability server in one of two ways:

   1.  Manual Configuration
   2.  Using the WebFinger Protocol

   The complete HTTPS URLs to be used when authenticating the enterprise
   edge element (optional) and obtaining the SIP service provider
   capability set can be obtained from the SIP service provider
   beforehand and entered into the edge element manually via some
   interface - for example, a CLI or GUI.

   However, if the resource URL is unknown to the administrator (and by
   extension of that to the edge element), the WebFinger protocol
   [RFC7033] may be leveraged.  From the perspective of this draft,
   three link relation types (rel) are defined, namely:

   1.  Capability Server: The base URL of the capability server hosting
       the capability set document.
   2.  Authorization Endpoint: The URL of the authorization endpoint to
       be used for OAuth 2.0
   3.  Token Endpoint: The URL of the token endpoint to be used for
       OAuth 2.0

   The corresponding link relation type values are as follows:

   1.  Capability Server: "<http://sipserviceprovider/capserver">
   2.  Authorization Endpoint: "<http://sipserviceprovider/auth">
   3.  Token Endpoint: "<http://sipserviceprovider/token">

   If an enterprise edge element attempts to discover the URL of the
   endpoints hosted in the ssp1.example.com domain, it issues the
   following request (line wraps are for display purposes only)









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              GET /.well-known/webfinger?
                resource=http%3A%2F%2Fssp1.example.com
                &rel=http%3A%2f%2fsipserviceprovider%2fcapserver
                &rel=http%3A%2f%2fsipserviceprovider%2auth
                &rel= http%3A%2f%2fsipserviceprovider%2token
                HTTP/1.1
              Host: ssp1.example.com


   The response to the above request might be as follows:

             HTTP/1.1 200 OK
             Access-Control-Allow-Origin: *
             Content-Type: application/jrd+json
             {
               "subject" : "http://ssp1.example.com",
               "links" :
               [
                 {
                   "rel" : "http://sipserviceprovider/capserver",
                   "href" : "https://capserver.ssp1.com"
                 },
                 {
                   "rel" : "http://sipserviceprovider/auth",
                   "href" : "https://ssp1.com/authorize"
                 },
                 {
                   "rel" : "http://sipserviceprovider/token",
                   "href" : "https://ssp1.com/token"
                 }
               ]
             }

   SIP service providers MUST support the "https" URI and "acct" URI
   [link] schemes in WebFinger queries.  The "acct" URI scheme might be
   used in the WebFinger query, if the enterprise adminsitrator is
   provided with a username by the SIP service provider.  The SIP
   service provider might provide a unique username for each SIP trunk
   purchased by the enterprise network or a single username that is
   applicable to all the trunks purchased by the enterprise network.
   This draft does not require SIP service providers or enterprise
   networks to favor one URI scheme over the other; rather, the choice
   of which scheme to use is left to the discretion of the SIP service
   provider.  The security considerations of using the "acct" URI is
   provided in section 5 of [RFC7565].

   The base URL of the capability server returned in the WebFinger
   response SHOULD not contain a path or query component.  Once the base



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   URL is obtained, the enterprise edge element builds on the base URL
   to identify the capability set document on the capability server.
   The general format for identifying resources on a capability server
   is as follows:

                    <scheme> /<baseURL>/<path>?<query>

   scheme: Is always HTTPS in the context of this draft.

   baseURL: The base URL of the capability server discovered as a result
   of the WebFinger query.

   path: The path expression identifying the capability set document on
   the capability server.  The path expression MUST be set to the static
   string of "capdoc".

   query: A query string identifying a specific representation of the
   capability set document.  The format of the query string is in the
   form of a "name=value" pair.  This draft defines the following
   optional query parameter:

   trunkid: A parameter uniquely identifying the SIP trunk for which the
   capability set document is sought.

   The "trunkid" is useful in situations in which an enterprise SIP
   network has multiple SIP trunks with the SIP service provider, such
   that parameters for such trunks vary, perhaps because of the
   geographical distribution of these sites.  The value of the "trunkid"
   parameter is generated by the SIP service provider and communicated
   to the enterprise SIP network by some out-of-band mechanism, for
   example, it may be provided in an email to the enterprise
   administrator after the trunk is purchased.  It is RECOMMENDED that
   SIP service providers generate unique trunk identifiers across
   enterprise networks.

   It is RECOMMENDED that SIP service provider networks support the
   "trunkid" query parameter.  SIP service providers expose varying
   capability sets to different enterprise SIP telephony networks.
   Using the "trunkid" query parameter not only helps the SIP service
   provider capability server to uniquely identify the trunk/enterprise/
   edge element for which the capability set document is being request,
   but also, it is helpful in generating unique URL strings for
   capability set documents.  These unique URL strings are helpful in
   HTTP content caching.







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7.  State Deltas

   Given that the service provider capability set is largely expected to
   remain static, the work needed to implement an asynchronous push
   mechanism to encode minor changes in the capability set document
   (state deltas) is not commensurate with the benefits.  Rather,
   enterprise edge elements can poll capability servers at pre-defined
   intervals to obtain the full capability set document.  It is
   RECOMMENDED that capability servers are polled every 24 hours.

8.  Encoding the Service Provider Capability Set

   In the context of this draft, the capability set of a service
   provider refers collectively to a set of characteristics which when
   communicated to an enterprise network, provides it with sufficient
   information to directly peer with the service provider network.  The
   capability set document is not designed to encode extremely granular
   details of all features, services, and protocol extensions that are
   supported by the service provider network.  For example, it is
   sufficient to encode that the service provider uses T.38 relay for
   faxing, it is not required to know the value of the
   "T38FaxFillBitRemoval" parameter.

   The parameters within the capability set document represent a wide
   array of characteristics, such that these characteristics
   collectively disseminate sufficient information to enable direct IP
   peering between enterprise and service provider networks.  The
   various parameters represented in the capability set are chosen based
   on existing practises and common problem sets typically seen between
   enterprise and service provider SIP networks.

9.  Data Model for Capability Set

   This section defines a YANG module for encoding the service provider
   capability set.  Section 9.1 provides the tree diagram, which is
   followed by a description of the various nodes within the module
   defined in this draft.

9.1.  Tree Diagram

   This section provides a tree diagram [RFC8340] for the "ietf-
   capability-set" module.  The interpretation of the symbols appearing
   in the tree diagram is as follows:

   o  Brackets "[" and "]" enclose list keys.

   o  Abbreviations before data node names: "rw" means configuration
      (read-write), and "ro" means state data (read-only).



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   o  Symbols after data node names: "?" means an optional node, "!"
      means a presence container, and "*" denotes a list and leaf-list.

   o  Parentheses enclose choice and case nodes, and case nodes are also
      marked with a colon (":").

   o  Ellipsis ("...") stands for contents of subtrees that are not
      shown.

   The data model for the peering capability document has the following
   structure:

              module: ietf-sip-auto-peering
                +--rw peering-info
                   +--rw variant           string
                   +--rw transport-info
                   |  +--rw transport?        enumeration
                   |  +--rw registrar*        host-port
                   |  +--rw registrarRealm?   string
                   |  +--rw callControl*      host-port
                   |  +--rw dns*              inet:ip-address
                   |  +--rw outboundProxy?    host-port
                   +--rw call-specs
                   |  +--rw earlyMedia?         boolean
                   |  +--rw signalingForking?   boolean
                   |  +--rw supportedMethods?   string
                   +--rw media
                   |  +--rw mediaTypeAudio
                   |  |  +--rw mediaFormat*   string
                   |  +--rw fax
                   |  |  +--rw protocol*   enumeration
                   |  +--rw rtp
                   |  |  +--rw RTPTrigger?     boolean
                   |  |  +--rw symmetricRTP?   boolean
                   |  +--rw rtcp
                   |     +--rw symmetricRTCP?   boolean
                   |     +--rw RTCPfeedback?    boolean
                   +--rw dtmf
                   |  +--rw payloadNumber?   int8
                   |  +--rw iteration?       boolean
                   +--rw security
                   |  +--rw signaling
                   |  |  +--rw type?      string
                   |  |  +--rw version?   string
                   |  +--rw mediaSecurity
                   |     +--rw keyManagement?   string
                   +--rw extensions?       string




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9.2.  YANG Model

   This section defines the YANG module for the peering capability set
   document.  It imports modules (ietf-yang-types and ietf-inet-types)
   from [RFC6991].

    module ietf-sip-auto-peering {
      namespace "urn:ietf:params:xml:ns:ietf-sip-auto-peering";
      prefix "peering";

      description
      "Data model for transmitting peering parameters from SP to Enterprise";

      revision 2019-05-06 {
        description "Initial revision of peering-response doc.";
      }

      import ietf-inet-types {
        prefix "inet";
      }

      typedef ipv4-address-port {
        type string {
          pattern '(([0-9]|[1-9][0-9]|1[0-9][0-9]|2[0-4][0-9]|25[0-5])\.){3}'
          +  '([0-9]|[1-9][0-9]|1[0-9][0-9]|2[0-4][0-9]|25[0-5])'
          + ':^()([1-9]|[1-5]?[0-9]{2,4}|6[1-4][0-9]{3}|65[1-4][0-9]{2}|655[1-2][0-9]|6553[1-5])$';
        }
        description "The ipv4-address-port type represents an IPv4 address in
        dotted-quad notation followed by a port number.";
      }

      typedef ipv6-address-port {
        type string {
          pattern '((:|[0-9a-fA-F]{0,4}):)([0-9a-fA-F]{0,4}:){0,5}'
          + '((([0-9a-fA-F]{0,4}:)?(:|[0-9a-fA-F]{0,4}))|'
          + '(((25[0-5]|2[0-4][0-9]|[01]?[0-9]?[0-9])\.){3}'
          + '(25[0-5]|2[0-4][0-9]|[01]?[0-9]?[0-9])))'
          + ':^()([1-9]|[1-5]?[0-9]{2,4}|6[1-4][0-9]{3}|65[1-4][0-9]{2}|655[1-2][0-9]|6553[1-5])$';
          pattern
          '(([^:]+:){6}(([^:]+:[^:]+)|(.*\..*)))|'
          + '((([^:]+:)*[^:]+)?::(([^:]+:)*[^:]+)?)'
          + ':^()([1-9]|[1-5]?[0-9]{2,4}|6[1-4][0-9]{3}|65[1-4][0-9]{2}|655[1-2][0-9]|6553[1-5])$';
        }
          description
          "The ipv6-address type represents an IPv6 address in full,
          mixed, shortened, and shortened-mixed notation followed by a port number.";
      }




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      typedef ip-address-port {
        type union {
          type ipv4-address-port;
          type ipv6-address-port;
        }
        description
        "The ip-address-port type represents an IP address:port number
        and is IP version neutral.";
      }

      typedef domain-name-port {
        type string {
          pattern
          '((([a-zA-Z0-9_]([a-zA-Z0-9\-_]){0,61})?[a-zA-Z0-9]\.)*'
          + '([a-zA-Z0-9_]([a-zA-Z0-9\-_]){0,61})?[a-zA-Z0-9]\.?)'
          + '|\.'
          + ':^()([1-9]|[1-5]?[0-9]{2,4}|6[1-4][0-9]{3}|65[1-4][0-9]{2}|655[1-2][0-9]|6553[1-5])$';
          length "1..258";
        }
        description
        "The domain-name-port type represents a DNS domain name followed by a port number.
        The name SHOULD be fully qualified whenever possible.";
      }

      typedef host-port {
        type union {
          type ip-address-port;
          type domain-name-port;
        }
        description
        "The host type represents either an IP address or a DNS
        domain name followed by a port number.";
      }

      container peering-info {
        leaf variant {
          type string;
          mandatory true;
          description "Variant of peering-response document";
        }

        container transport-info {
          leaf transport {
            type enumeration {
              enum "TCP";
              enum "TLS";
              enum "UDP";
              enum "TCP;TLS";



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              enum "TCP;TLS;UDP";
              enum "TCP;UDP";
            }
            description "Transport Protocol(s) used in SIP communication";
          }

          leaf-list registrar {
            type host-port;
            max-elements 3;
            description "List of service provider registrar servers";
          }

          leaf registrarRealm {
            type string;
            description "Realm for REGISTER requests carrying credentials";
          }

          leaf-list callControl {
            type host-port;
            max-elements 3;
            description "List of service provider call control servers";
          }

          leaf-list dns {
            type inet:ip-address;
            max-elements 2;
            description "IP address of the DNS Server(s) hosted by the service provider";
          }

          leaf outboundProxy {
            type host-port;
            description "SIP Outbound Proxy";
          }
        }

        container call-specs {
          leaf earlyMedia {
            type boolean;
            description "Flag indicating whether the service provider is expected
            to deliver early media.";
          }

          leaf signalingForking {
            type boolean;
            description "Flag indicating whether the service provider is capable
            of forking incoming calls ";
          }




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          leaf supportedMethods {
            type string;
            description "Leaf/Leaf List indicating the different SIP methods
            support by the service provider.";
          }
        }

        container media {
          container mediaTypeAudio {
            leaf-list mediaFormat {
              type string;
              description "Leaf List indicating the audio media formats supported.";
            }
          }

          container fax {
            leaf-list protocol {
              type enumeration {
                enum "pass-through";
                enum "t38";
              }
              max-elements 2;
              description "Leaf List indicating the different fax protocols
              supported by the service provider.";
            }
          }

          container rtp {
            leaf RTPTrigger {
              type boolean;
              description "Flag indicating whether the service provider expects to
              receive the first media packet.";
            }

            leaf symmetricRTP {
              type boolean;
              description "Flag indicating whether the service provider expects
              symmetric RTP defined in [@RFC4961]";
            }
          }

          container rtcp {
            leaf symmetricRTCP {
              type boolean;
              description " Flag indicating whether the service provider expects
              symmetric RTP defined in [@RFC4961].";
            }




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            leaf RTCPfeedback {
              type boolean;
              description "Flag Indicating support for RTP profile extension for
              RTCP-based feedback, as defined in [@RFC4585]";
            }
          }
        }

        container dtmf {
          leaf payloadNumber {
            type int8 {
              range "96..127";
            }
            description "Leaf that indicates the payload number(s) supported by
            the service provider for DTMF relay via Named-Telephony-Events";
          }

          leaf iteration {
            type boolean;
            description "Flag identifying whether the service provider supports
            NTE DTMF relay using the procedures of [@RFC2833] or [@RFC4733] .";
          }
        }

        container security {
          container signaling {
            leaf type {
              type string {
                pattern "TLS";
              }
              description "Type of signaling security supported.";
            }

            leaf version {
              type string {
                pattern "([1-9]\.[0-9])(;[1-9]\.[0-9])?|(NULL)";
              }
              description "Indicates TLS version for SIP signaling";
            }
          }

          container mediaSecurity {
            leaf keyManagement {
              type string {
                pattern "(SDES(;DTLS-SRTP,version=[1-9]\.[0-9](,[1-9]\.[0-9])?)?)|(DTLS-SRTP,version=[1-9]\.[0-9](,[1-9]\.[0-9])?)|(NULL)";
              }
              description "Leaf that identifies the key management methods
              supported by the service provider for SRTP.";



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            }
          }
        }

        leaf extensions {
          type string;
          description "Lists the various SIP extensions supported by SP";
        }
      }
    }

9.3.  Node Definitions

   This sub-sections provides the definition and encoding rules of the
   various nodes of the YANG module defined in section 9.2

   o  *capability-set*: This node serves as a container for all the
      other nodes in the YANG module; the capability-set node is akin to
      the root element of an XML schema.

   o  *variant*: This node identifies the version number of the
      capability set document.  This draft defines the parameters for
      variant 1.0; future specifications might define a richer parameter
      set, in which case the variant can be changed to 2.0, 3.0 and so
      on.  Future extensions to the capability set document MUST also
      ensure that the corresponding YANG module is defined.

   o  *transport-info*: The transport-info node is a container that
      encapsulates transport characteristics of SIP sessions between
      enterprise and service provider networks.

   o  *transport*: A leaf node that enumerates the different Transport
      Layer protocols supported by the SIP service provider.  Valid
      transport layer protocols include: UDP, TCP, TLS or a combination
      of them (with the exception of TLS and UDP).

   o  *registrar*: A leaf-list that specifies the transport address of
      one or more registrar servers in the service provider network.
      The transport address of the registrar can be provided using a
      combination of a valid IP address and port number, or a subdomain
      of the SIP service provider network, or the fully qualified domain
      name (FQDN) of the SIP service provider network.  If the transport
      address of a registrar is specified using either a subdomain or a
      fully qualified domain name, the DNS element needs to be populated
      with one or more valid DNS server IP addresses.

   o  *callControl*: A leaf-list that specifies the transport address of
      the call server(s) in the service provider network.  The



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      enterprise network MUST use an applicable transport protocol in
      conjunction with the call control server(s) transport address when
      transmitting call setup requests.  The transport address of a call
      server(s) within the service provider network can be specified
      using a combination of a valid IP address and port number, or a
      subdomain of the SIP service provider network, or a fully
      qualified domain name of the SIP service provider network.  If the
      transport address of a call control server(s) is specified using
      either a subdomain or a fully qualified domain name, the DNS
      element MUST be populated with one or more valid DNS server IP
      addresses.  The transport address specified in this element can
      also serve as the target for non-call requests such as SIP
      OPTIONS.

   o  *dns*: A leaf list that encodes the IP address of one or more DNS
      servers hosted by the SIP service provider.  If the enterprise
      network is unaware of the IP address, port number, and transport
      protocol of servers within the service provider network (for
      example, the registrar and call control server), it MUST use DNS
      NAPTR and SRV.  Alternatively, if the enterprise network has the
      fully qualified domain name of the SIP service provider network,
      it MUST use DNS to resolve the said FQDN to an IP address.  The
      dns element encodes the IP address of one or more DNS servers
      hosted in the service provider network.  If however, either the
      registrar or callControl elements or both are populated with a
      valid IP address and port pair, the dns element MUST be set to the
      quadruple octet of 0.0.0.0.

   o  *outboundProxy*: A leaf list that specifies the transport address
      of one or more outbound proxies.  The transport address can be
      specified by using a combination of an IP address and a port
      number, a subdomain of the SIP service provider network, or a
      fully qualified domain name and port number of the SIP service
      provider network.  If the outbound-proxy sub-element is populated
      with a valid transport address, it represents the default
      destination for all outbound SIP requests and therefore, the
      registrar and callControl elements MUST be populated with the
      quadruple octet of 0.0.0.0.

   o  *call-specs*: A container that encapsulates information about call
      specifications, restrictions and additional handling criteria for
      SIP calls between the enterprise and service provider network.

   o  *earlyMedia*: A leaf that specifies whether the service provider
      network is expected to deliver in-band announcements/tones before
      call connect.  The P-Early-Media header field can be used to
      indicate pre-connect delivery of tones and announcements on a per-
      call basis.  However, given that signalling and media could



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      traverse a large number of intermediaries with varying
      capabilities (in terms of handling of the P-Early-Media header
      field) within the enterprise, such devices can be appropriately
      configured for media cut through if it is known before-hand that
      early media is expected for some or all of the outbound calls.
      This element is a Boolean type, where a value of 1/true signifies
      that the service provider is capable of early media.  A value of
      0/false signifies that the service provider is not expected to
      generate early media.

   o  *signalingForking*: A leaf that specifies whether outbound call
      requests from the enterprise might be forked on the service
      provider network leading to multiple early dialogs.  This
      information would be useful to the enterprise network in
      appropriately handling multiple early dialogs reliably and in
      enforcing local policy.  This element is a Boolen type, where a
      value of 1/true signifies that the service provider network can
      potentially fork outbound call requests from the enterprise.  A
      value of 0/false indicates that the service provider will not fork
      outbound call requests.

   o  *supportedMethods*: A leaf node that specifies the various SIP
      methods supported by the SIP service provider.  The list of
      supported methods help to appropriately configuration various
      devices within the enterprise network.  For example, if the
      service provider enumerates support for the OPTIONS method, the
      enterprise network could periodically send OPTIONS requests as a
      keep-alive mechanism.

   o  *media*: A container that is used to collectively encapsulate the
      characteristics of UDP-based audio streams.  A future
      extension to this draft may extend the media container to describe
      other media types.  The media container is also used to
      encapsulate basic information about Real-Time Transport Protocol
      (RTP) and Real-Time Transport Control Protocol (RTCP) from the
      perspective of the service provider network.

   o  *mediaTypeAudio*: A container for the mediaFormat leaf-list.  This
      container collectively encapsulates the various audio media
      formats supported by the SIP service provider.

   o  *mediaFormat*: A leaf-list encoding the various audio media
      formats supported by the SIP service provider.  The relative
      ordering of different media format leaf nodes from left to right
      indicates preference from the perspective of the service provider.
      Each mediaFormat node begins with the encoding name of the media
      format, which is the same encoding name as used in the "RTP/AVP"
      and "RTP/SAVP" profiles.  The encoding name is followed by



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      required and optional parameters for the given media format as
      specified when the media format is registered [RFC4855].  Given
      that the parameters of media formats can vary from one
      communication session to another, for example, across two separate
      communication sessions, the packetization time (ptime) used for
      the PCMU media format might vary from 10 to 30 ms, the parameters
      included in the format element MUST be the ones that are expected
      to be invariant from the perspective of the service provider.
      Providing information about supported media formats and their
      respective parameters, allows enterprise networks to configure the
      media plane characteristics of various devices such as endpoints
      and middleboxes.  The encoding name, one or more required
      parameters, one or more optional parameters are all separated by a
      semicolon.  The formatting of a given media format parameter, MUST
      follow the formatting rules as specified for that media format.

   o  *fax*: A container that encapsulates the fax protocol(s) supported
      by the SIP service provider.  The fax container encloses a leaf-
      list (named protocol) that enumerates whether the service provider
      supports t38 relay, protocol-based fax passthrough or both.  The
      relative ordering of leaf nodes within the leaf lists indicates
      preference.

   o  *rtp*: A container that encapsulates generic characteristics of
      RTP sessions between the enterprise and service provider network.
      This node is a container for the "RTPTrigger" and "SymmetricRTP"
      leaf nodes.

   o  *RTPTrigger*: A leaf node indicating whether the SIP service
      provider network always expects the enterprise network to send the
      first RTP packet for an established communication session.  This
      information is useful in scenarios such as "hairpinned" calls, in
      which the caller and callee are on the service provider network
      and because of sub-optimal media routing, an enterprise device
      such as an SBC is retained in the media path.  Based on the
      encoding of this node, it is possible to configure enterprise
      devices such as SBCs to start streaming media (possibly filled
      with silence payloads) toward the address:port tuples provided by
      caller and callee.  This node is a Boolean type.  A value of 1/
      true indicates that the service provider expects the enterprise
      network to send the first RTP packet, whereas a value of 0/false
      indicates that the service provider network does not require the
      enterprise network to send the first media packet.  While the
      practise of preserving the enterprise network in a hairpinned call
      flow is fairly common, it is RECOMMENDED that SIP service
      providers avoid this practise.  In the context of a hairpinned
      call, the enterprise device retained in the call flow can easily
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   o  *symmetricRTP*: A leaf node indicating whether the SIP service
      provider expects the enterprise network to use symmetric RTP as
      defined in [RFC4961].  Uncovering this expectation is useful in
      scenarios where "latching" [RFC3762] is implemented in the service
      provider network.  This node is a Boolean type, a value of 1/true
      indicates that the service provider expects the enterprise network
      to use symmetric RTP, whereas a value of 0/false indicates that
      the enterprise network can use asymmetric RTP.

   o  *rtcp*: A container that encapsulates generic characteristics of
      RTCP sessions between the enterprise and service provider network.
      This node is a container for the "RTCPFeedback" and
      "SymmetricRTCP" leaf nodes.

   o  *RTCPFeedback*: A leaf node that indicates whether the SIP service
      provider supports the RTP profile extension for RTCP-based
      feedback [RFC4585].  Media sessions spanning enterprise and
      service provider networks, are rarely made to flow directly
      between the caller and callee, rather, it is often the case that
      media traffic flows through network intermediaries such as SBCs.
      As a result, RTCP traffic from the service provider network is
      intercepted by these intermediaries, which in turn can either pass
      across RTCP traffic unmodified or modify RTCP traffic before it is
      forwarded to the endpoint in the enterprise network.  Modification
      of RTCP traffic would be required, for example, if the
      intermediary has performed media payload transformation operations
      such as transcoding or transrating.  In a similar vein, for the
      RTCP-based feedback mechanism as defined in [RFC4585] to be truly
      effective, intermediaries MUST ensure that feedback messages are
      passed reliably and with the correct formatting to enterprise
      endpoints.  This might require additional configuration and
      considerations that need to be dealt with at the time of
      provisioning the intermediary device.  This node is a Boolean
      type, a value of 1/true indicates that the service provider
      supports the RTP profile extension for RTP-based feedback and a
      value of 0/false indicates that the service provider does not
      support the RTP profile extension for RTP-based feedback.

   o  *symmetricRTCP*: A leaf node indicating whether the SIP service
      provider expects the enterprise network to use symmetric RTCP as
      defined in [RFC4961].  This node is a Boolean type, a value of 1
      indicates that the service provider expects symmetric RTCP
      reports, whereas a value of 0 indicates that the enterprise can
      use asymmetric RTCP.

   o  *dtmf*: A container that describes the various aspects of DTMF
      relay via RTP Named Telephony Events.  The dtmf container allows




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      SIP service providers to specify two facets of DTMF relay via
      Named Telephony Events:

      1.  The payload type number using the payloadNumber leaf node.
      2.  Support for [RFC2833] or [RFC4733] using the iteration leaf node.

      In the context of named telephony events, senders and receivers may
      negotiate asymmetric payload type numbers.  For example, the sender
      might advertise payload type number 97 and the receiver might
      advertise payload type number 101.  In such instances, it is either
      required for middleboxes to interwork payload type numbers or allow
      the endpoints to send and receive asymmetric payload numbers.  The
      behaviour of middleboxes in this context is largely dependent on
      endpoint capabilities or on service provider constraints.  Therefore,
      the payloadNumber leaf node can be used to determine middlebox
      configuration before-hand.

      [RFC4733]iterates over [RFC2833] by introducing certain changes in
      the way NTE events are transmitted.  SIP service providers can
      indicate support for [RFC4733] by setting the iteration flag to 1 or
      indicating support for [RFC2833] by setting the iteration flag to 0.

   o  *security*: A container that encapsulates characteristics about
      encrypting signalling streams between the enterprise and SIP
      service provider networks.

   o  *signaling*: A container that encapsulates the type of security
      protocol for the SIP communication between the enterprise SBC and
      the service provider.

   o  *type*: A leaf node that specifies the protocol used for
      protecting SIP signalling messages between the enterprise and
      service provider network.  The value of the type leaf node is only
      defined for Transport Layer Security (TLS).  Accordingly, if TLS
      is allowed for SIP sessions between the enterprise and service
      provider network, the type leaf node is set to the string "tls".

   o  *version*: A leaf node that specifies the version(s) of TLS
      supported in decimal format.  If multiple versions of TLS are
      supported, they MUST be separated by semi-colons.  If the service
      provide does not support TLS for protecting SIP sessions, the
      signalling element is set to the string "NULL".

   o  *mediaSecurity*: A container that describes the various
      characteristics of securing media streams between enterprise and
      service provider networks.





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   o  *keyManagement*: A leaf node that specifies the key management
      method used by the service provider.  Possible values of this node
      include: "SDES" and "DTLS-SRTP".  A value of "SDES" signifies that
      the SIP service provider uses the methods defined in [RFC4568] for
      the purpose of key management.  A value of "DTLS-SRTP" signifies
      that the SIP service provider uses the methods defined in
      [RFC5764] for the purpose of key management.  If the value of this
      leaf node is set to "DTLS-SRTP", the various versions of DTLS
      supported by the SIP service provider MUST be encoded as per the
      formatting rules of Section 9.2.  If the service provider does not
      support media security, the keyManagement node MUST be set to
      "NULL".

   o  *extensions*: A leaf node that is a semicolon separated list of
      all possible SIP option tags supported by the service provider
      network.  These extensions MUST be referenced using name
      registered under IANA.  If the service provider network does not
      support any extensions to baseline SIP, the extensions node MUST
      be set to "NULL".

9.4.  Extending the Capability Set

   There are situations in which equipment manufactures or service
   providers would benefit from extending the YANG module defined in
   this draft.  For example, service providers could extend the YANG
   module to include information that further simplifies direct IP
   peering.  Such information could include: trunk group identifiers,
   direct-inward-dial (DID) number ranges allocated to the enterprise,
   customer/enterprise account numbers, service provider support
   numbers, among others.

   Extension of the module can be achieved by importing the module
   defined in this draft.  An example is provided below:

   Consider a new YANG module "vendorA" specified for VendorA's
   enterprise SBC.  The "vendorA-config" YANG module is configured as
   follows:














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    module vendorA-config {
      namespace "urn:ietf:params:xml:ns:yang:vendorA-config";
      prefix "vendorA";

      description
      "Data model for configuring VendorA Enterprise SBC";

      revision 2020-05-06 {
      description "Initial revision of VendorA Enterprise SBC configuration data model";
      }

      import ietf-peering {
        prefix "peering";
      }

      augment "/peering:peering-info" {
        container vendorAConfig {
          leaf vendorAConfigParam1 {
            type int32;
            description "vendorA configuration parameter 1 (SBC Device ID)";
          }

          leaf vendorAConfigParam2 {
            type string;
              description "vendorA configuration parameter 2 (SBC Device name)";
          }
          description "Container for vendorA SBC configuration";
        }
      }
    }

   In the example above, a custom module named "vendorA-config" uses the
   "augment" statement as defined in Section 4.2.8 of [RFC7950] to
   extend the module defined in this draft.

10.  Example Capability Set Document Encoding

   This section provides examples of how capability set documents that
   leverage the YANG module defined in this document can be encoded over
   JSON or XML.

10.1.  JSON Capability Set Document









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    {
      "peering-info:variant": "1.0",
      "transport-info": {
        "transport": "TCP;TLS;UDP",
        "registrar": ["registrar1.voip.example.com:5060", "registrar2.voip.example.com:5060"],
        "registrarRealm": "voip.example.com",
        "callControl": ["callServer1.voip.example.com:5060", "192.168.12.25:5065"],
        "dns": [8.8.8.8, 208.67.222.222],
        "outboundProxy": "0.0.0.0"
      },
      "call-specs": {
        "earlyMedia": "true",
        "signalingForking": "false",
        "supportedMethods": "INVITE;OPTIONS;BYE;CANCEL;ACK;PRACK;SUBSCRIBE;NOTIFY;REGISTER"
      },
      "media": {
        "mediaTypeAudio": {
          "mediaFormat": ["PCMU;rate=8000;ptime=20","G729;rate=8000;annexb=yes","G722;rate=8000;bitrate=56k,64k"]
        },
        "fax": {
          "protocol": ["pass-through", "t38"]
        },
        "rtp": {
          "RTPTrigger": "false",
          "symmetricRTP": "true"
        },
        "rtcp": {
          "symmetricRTCP": "true",
          "RTCPFeedback": "true"
        }
      },
      "dtmf": {
        "payloadNumber": "101",
        "iteration": "0"
      },
      "security": {
        "signaling": {
          "type": "TLS",
          "version": "1.0;1.2"
        },
        "mediaSecurity": {
          "keyManagement": "SDES;DTLS-SRTP,version=1.2"
        }
      },
      "extensions": "timer;rel100;gin;path"
    }





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10.2.  XML Capability Set Document

    <peering-info xmlns="urn:ietf:params:xml:ns:yang:ietf-peering"
      xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
      xsi:schemaLocation="urn:ietf:params:xml:ns:yang:ietf-peering ietf-peering.xsd">
      <variant>1.0</variant>
      <transport-info>
        <transport>TCP;TLS;UDP</transport>
        <registrar>registrar1.voip.example.com:5060</registrar>
        <registrar>registrar2.voip.example.com:5060</registrar>
        <registrarRealm>voip.example.com</registrarRealm>
        <callControl>callServer1.voip.example.com:5060</callControl>
        <callControl>192.168.12.25:5065</callControl>
        <dns>8.8.8.8</dns>
        <dns>208.67.222.222</dns>
        <outboundProxy>0.0.0.0</outboundProxy>
      </transport-info>
      <call-specs>
        <earlyMedia>true</earlyMedia>
        <signalingForking>false</signalingForking>
        <supportedMethods>INVITE;OPTIONS;BYE;CANCEL;ACK;PRACK;SUBSCRIBE;NOTIFY;REGISTER</supportedMethods>
      </call-specs>
      <media>
        <mediaTypeAudio>
          <mediaFormat>PCMU;rate=8000;ptime=20</mediaFormat>
          <mediaFormat> G729;rate=8000;annexb=yes</mediaFormat>
          <mediaFormat>G722;rate=8000;bitrate=56k,64k</mediaFormat>
        </mediaTypeAudio>
        <fax>
          <protocol>pass-through</protocol>
          <protocol>t38</protocol>
        </fax>
        <rtp>
          <RTPTrigger>true</RTPTrigger>
          <symmetricRTP>true</symmetricRTP>
        </rtp>
        <rtcp>
          <symmetricRTCP>true</symmetricRTCP>
          <RTCPFeedback>true</RTCPFeedback>
        </rtcp>
      </media>
      <dtmf>
        <payloadNumber>101</payloadNumber>
        <iteration>0</iteration>
      </dtmf>
      <security>
        <signaling>
          <type>TLS</type>



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          <version>1.0;1.2</version>
        </signaling>
        <mediaSecurity>
          <keyManagement>SDES;DTLS-SRTP,version=1.2</keyManagement>
          </mediaSecurity>
        </security>
      <extensions>timer;rel100;gin;path</extensions>
    </peering-response>

11.  Example Exchange

   This section depicts an example of the configuration flow that
   ultimately results in the enterprise edge element obtaining the
   capability set document from the SIP service provider.

   Assuming the enterprise edge element isn't pre-configured with the
   request target for the capability set document and is required to
   authenticate with the SIP service provider capability server, the
   following sequence of events are put into motion to obtain the
   capability set document:

   The enterprise edge element generates a WebFinger query to discover
   endpoints hosted in the ssp1.example.com domain (line wraps are for
   display purposes only)

       GET /.well-known/webfinger?
         resource=http%3A%2F%2Fssp1.example.com
         &rel=http%3A%2f%2fsipserviceprovider%2fcapserver
         &rel=http%3A%2f%2fsipserviceprovider%2auth
         &rel= http%3A%2f%2fsipserviceprovider%2token
         HTTP/1.1
       Host: ssp1.example.com

   The resulting WebFinger response, contains the URLs of the capability
   server, the OAuth 2.0 authorization endpoint and OAuth 2.0 token
   endpoint.















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           HTTP/1.1 200 OK
           Access-Control-Allow-Origin: *
           Content-Type: application/jrd+json
           {
             "subject" : "http://ssp1.example.com",
             "links" :
             [
               {
                 "rel" : "http://sipserviceprovider/capserver",
                 "href" : "https://capserver.ssp1.com"
               },
               {
                 "rel" : "http://sipserviceprovider/auth",
                 "href" : "https://ssp1.com/authorize"
               },
               {
                 "rel" : "http://sipserviceprovider/token",
                 "href" : "https://ssp1.com/token"
               },
             ]
           }


The endpoint URLs returned in the WebFinger response are stored by the
edge element and referenced when required. Then, the administrator logs
into the GUI of the edge element and initiates the download of the
service provider capability set (perhaps by clicking on a button). This
triggers the edge element to redirect the administrator to the OAuth 2.0
authorization endpoint (discovered via WebFinger). Once the
administrator is authenticated and provides authorization, flow is
redirected to the callback URL of the edge element application. The edge
element then contacts the OAuth 2.0 token endpoint (discovered via
WebFinger) to authenticate itself and obtain access and refresh tokens.
Accordingly, the edge element mints a JWT bearer token to authenticate
itself with the token endpoint and obtain an access and refresh token.
Below is an example of client authentication using a JWT during the
presentation of an authorization code grant for an access token request
(line wraps are for display purposes only).













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           POST /token  HTTP/1.1
           Host: ssp1.com
           Content-Type: application/x-www-form-urlencoded

           grant_type=authorization_code&
           code=n0esc3NRze7LTCu7iYzS6a5acc3f0ogp4&
           client_assertion_type=urn%3Aietf%3Aparams%3Aoauth%3A
           client-assertion-type%3Ajwt-bearer&
           client_assertion=eyJhbGciOiJSUzI1NiIsImtpZCI6IjIyIn0.
           eyJpc3Mi[...omitted for brevity...].
           cC4hiUPo[...omitted for brevity...]


 If the request is acceptable to the token endpoint, an access token and
 a refresh token is provided in the response. For example:


               HTTP/1.1 200 OK
               Content-Type: application/json;charset=UTF-8
               Cache-Control: no-store
               Pragma: no-cache

               {
                 "access_token":"sF_9.B5f-4.1JqM",
                 "token_type":"Bearer",
                 "expires_in":86400,
                 "refresh_token":"hGzv3JOkF0XG5Qx2TlKWIA"
               }


 The obtained bearer access token can subsequently be used to obtain the
 capability set document for the capability server. The edge element
 generates a HTTPS GET request with the bearer token included in the
 Authorization header field.


                GET //capdoc?trunkid=trunkent1456 HTTP/1.1
                Host: capserver.ssp1.com
                Authorization: Bearer mF_9.B5f-4.1JqM
                Accept:application/peering-info+xml


 The capability set document is obtained in the body of the response and
 is encoded in XML.







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                HTTP/1.1 200 OK
                Content-Type: application/peering-info+xml
                Content-Length: nnn

                <peering-info>
                ...
                </peering-info>



12.  Security Considerations

   Capability set documents have a significant bearing on the quality of
   the peering relationship between an enterprise and service provider
   network.  These documents can be modified by an attacker to
   drastically impact the quality of communication sessions between
   enterprise and service provider networks.  Additionally, capability
   set documents contain parameters that may be considered sensitive
   from the perspective of the SIP service provider.  For example, the
   YANG model defined in this document might be extended by SIP service
   providers to include account sensitive information such as the
   username and password to used when generating an MD5 response to
   401/407 SIP challenges.

   To ensure the problems discussed in the previous paragraph are
   accounted for, the following considerations MUST be taken into
   account:

   o  Integrity and Confidentiality

   Request and responses for the capability set documents are defined
   over HTTP.  However, due to the sensitive nature of information
   transmitted between client and server, it is required to secure HTTP
   using Transport Layer Security.  The enterprise edge element and
   capability server MUST be compliant to [RFC7235].  The enterprise
   edge element and capability server MUST support the use of the https
   uri scheme as defined in [RFC7230].

   o  Authenticated Client Identity

   While this draft does not enforce client authentication, there are
   situations in which client need to authenticated by SIP service
   providers before they are provided capability set documents.  In such
   situations client MUST be authenticated using the procedures outlined
   in section 6.3 of this draft.

   o  The "trunkid" parameter




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   It is RECOMMENDED that enterprise edge elements use the "trunkid"
   parameter in query strings when requesting for the capability set
   documents.  The value of "trunkid" parameter is generated by the SIP
   service provider and provided to the administrator via some out-of-
   band mechanism.  SIP service providers MUST ensure that value of the
   "trunkid" parameters does not inadvertently communicate sensitive
   information to an attacker such as a username or password credential.

   In addition to the considerations listed above, all the security
   considerations that are part of the WebFinger and OAuth 2.0
   specifications are applicable to this draft.

13.  Acknowledgments

   TBD

14.  References

14.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

   [RFC6020]  Bjorklund, M., Ed., "YANG - A Data Modeling Language for
              the Network Configuration Protocol (NETCONF)", RFC 6020,
              DOI 10.17487/RFC6020, October 2010,
              <https://www.rfc-editor.org/info/rfc6020>.

   [RFC6991]  Schoenwaelder, J., Ed., "Common YANG Data Types",
              RFC 6991, DOI 10.17487/RFC6991, July 2013,
              <https://www.rfc-editor.org/info/rfc6991>.

14.2.  Informative References

   [RFC2833]  Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF
              Digits, Telephony Tones and Telephony Signals", RFC 2833,
              DOI 10.17487/RFC2833, May 2000,
              <https://www.rfc-editor.org/info/rfc2833>.

   [RFC3762]  Levin, O., "Telephone Number Mapping (ENUM) Service
              Registration for H.323", RFC 3762, DOI 10.17487/RFC3762,
              April 2004, <https://www.rfc-editor.org/info/rfc3762>.







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   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
              <https://www.rfc-editor.org/info/rfc4568>.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,
              <https://www.rfc-editor.org/info/rfc4585>.

   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
              Digits, Telephony Tones, and Telephony Signals", RFC 4733,
              DOI 10.17487/RFC4733, December 2006,
              <https://www.rfc-editor.org/info/rfc4733>.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007,
              <https://www.rfc-editor.org/info/rfc4855>.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
              <https://www.rfc-editor.org/info/rfc4961>.

   [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,
              DOI 10.17487/RFC5764, May 2010,
              <https://www.rfc-editor.org/info/rfc5764>.

   [RFC6241]  Enns, R., Ed., Bjorklund, M., Ed., Schoenwaelder, J., Ed.,
              and A. Bierman, Ed., "Network Configuration Protocol
              (NETCONF)", RFC 6241, DOI 10.17487/RFC6241, June 2011,
              <https://www.rfc-editor.org/info/rfc6241>.

   [RFC6665]  Roach, A., "SIP-Specific Event Notification", RFC 6665,
              DOI 10.17487/RFC6665, July 2012,
              <https://www.rfc-editor.org/info/rfc6665>.

   [RFC6750]  Jones, M. and D. Hardt, "The OAuth 2.0 Authorization
              Framework: Bearer Token Usage", RFC 6750,
              DOI 10.17487/RFC6750, October 2012,
              <https://www.rfc-editor.org/info/rfc6750>.

   [RFC7033]  Jones, P., Salgueiro, G., Jones, M., and J. Smarr,
              "WebFinger", RFC 7033, DOI 10.17487/RFC7033, September
              2013, <https://www.rfc-editor.org/info/rfc7033>.




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   [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Message Syntax and Routing",
              RFC 7230, DOI 10.17487/RFC7230, June 2014,
              <https://www.rfc-editor.org/info/rfc7230>.

   [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
              DOI 10.17487/RFC7231, June 2014,
              <https://www.rfc-editor.org/info/rfc7231>.

   [RFC7235]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Authentication", RFC 7235,
              DOI 10.17487/RFC7235, June 2014,
              <https://www.rfc-editor.org/info/rfc7235>.

   [RFC7565]  Saint-Andre, P., "The 'acct' URI Scheme", RFC 7565,
              DOI 10.17487/RFC7565, May 2015,
              <https://www.rfc-editor.org/info/rfc7565>.

   [RFC7950]  Bjorklund, M., Ed., "The YANG 1.1 Data Modeling Language",
              RFC 7950, DOI 10.17487/RFC7950, August 2016,
              <https://www.rfc-editor.org/info/rfc7950>.

   [RFC8340]  Bjorklund, M. and L. Berger, Ed., "YANG Tree Diagrams",
              BCP 215, RFC 8340, DOI 10.17487/RFC8340, March 2018,
              <https://www.rfc-editor.org/info/rfc8340>.

Authors' Addresses

   Kaustubh Inamdar
   Cisco Systems

   Email: kinamdar@cisco.com


   Sreekanth Narayanan
   Cisco Systems

   Email: sreenara@cisco.com


   Cullen Jennings
   Cisco Systems

   Email: fluffy@iii.ca






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