Automatic Peering for SIP Trunks
draft-ietf-asap-sip-auto-peer-00

Document Type Active Internet-Draft (asap WG)
Authors Kaustubh Inamdar  , Sreekanth Narayanan  , Cullen Jennings 
Last updated 2021-04-01
Replaces draft-kinamdar-dispatch-sip-auto-peer
Stream Internet Engineering Task Force (IETF)
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ASAP                                                          K. Inamdar
Internet-Draft                                              S. Narayanan
Expires: October 3, 2021                                     C. Jennings
                                                           Cisco Systems
                                                           April 1, 2021

                    Automatic Peering for SIP Trunks
                                draft-ietf-asap-sip-auto-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 October 3, 2021.

Copyright Notice

   Copyright (c) 2021 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
   include Simplified BSD License text as described in Section 4.e of

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   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  . . . . . . . . . . . . . . . . . . . . . . .   8
     6.1.  HTTP Methods  . . . . . . . . . . . . . . . . . . . . . .   8
     6.2.  Integrity and Confidentiality . . . . . . . . . . . . . .   8
     6.3.  Authenticated Client Identity . . . . . . . . . . . . . .   8
     6.4.  Encoding the Request  . . . . . . . . . . . . . . . . . .   8
     6.5.  Generating the response . . . . . . . . . . . . . . . . .   9
     6.6.  Identifying the Request Target  . . . . . . . . . . . . .   9
   7.  State Deltas  . . . . . . . . . . . . . . . . . . . . . . . .   9
   8.  Encoding the Service Provider Capability Set  . . . . . . . .  10
   9.  Data Model for Capability Set . . . . . . . . . . . . . . . .  10
     9.1.  Tree Diagram  . . . . . . . . . . . . . . . . . . . . . .  10
     9.2.  YANG Model  . . . . . . . . . . . . . . . . . . . . . . .  12
     9.3.  Node Definitions  . . . . . . . . . . . . . . . . . . . .  17
     9.4.  Extending the Capability Set  . . . . . . . . . . . . . .  24
   10. Example Capability Set Document Encoding  . . . . . . . . . .  25
     10.1.  XML Capability Set Document  . . . . . . . . . . . . . .  26
   11. Example Exchange  . . . . . . . . . . . . . . . . . . . . . .  27
   12. Security Considerations . . . . . . . . . . . . . . . . . . .  28
   13. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  28
   14. References  . . . . . . . . . . . . . . . . . . . . . . . . .  28
     14.1.  Normative References . . . . . . . . . . . . . . . . . .  28
     14.2.  URIs . . . . . . . . . . . . . . . . . . . . . . . . . .  30
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  31

1.  Introduction

   The deployment of a Session Initiation Protocol [RFC 3261 [1]] (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, extensions to baseline protocols and
   operating principles ought to be implemented by service provider and
   enterprise networks to ensure successful peering.

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   The SIP Connect technical recommendations [2] 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.  However, this
   workflow complements the SIP Connect technical recommendations, in
   that both endeavours aim to promote/achieve interop between the
   enterprise and service provider.

   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 RFC 2119 [3]

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 [RFC 7092 [4]].  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 HTTPS [RFC 7231 [5]] server.

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       +-----------------------------------------------------+
       | +---------------+         +-----------------------+ |
       | |               |         |                       | |
       | | +----------+  |         |   +-------+           | |
       | | |   Cap    |  | HTTPS   |   |       |           | |
       | | |  Server  |<-|---------|-->|       |           | |
       | | |          |  |         |   |       |   +-----+ | |
       | | +----------+  |         |   |       |   | SIP | | |
       | |               |         |   |       |<->| PBX | | |
       | |               |         |   |       |   +-----+ | |
       | | +----------+  |         |   |  SBC  |           | |
       | | |          |  |   SIP   |   |       |           | |
       | | | SP - SSE |<-|---------|-->|       |   +-----+ | |
       | | |          |  |         |   |       |<->| M.E.| | |
       | | +----------+  |         |   |       |   |     | | |
       | |               |         |   |       |   +-----+ | |
       | | +----------+  | (S)RTP  |   |       |           | |
       | | |  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) [RFC 7092 [6]] 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 HTTPS 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.

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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 collectivity to a set of
      characteristics within the service provider network, which when
      communicated to the enterprise network, provides the enterprise
      network the information required to interconnect with the service
      provider network.  The various parameters that constitute the
      capability set relate to characteristics that are specific to
      signalling, media, transport and security.  Certain aspects of
      interconnecting with service providers are out of scope of the
      capability set.  For example, the access technology used to
      interconnect with service provider networks.

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

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   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.

   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 [RFC
   6665 [7]], 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 barrier to adoption for SIP
   service providers and enterprise networks as equipment manufacturers
   would have to first add support for such a SIP extension.  A HTTPS-
   based approach would be relatively easier to adopt as most edge
   devices deployed in enterprise networks today already support HTTPS;
   from the perspective of service provider networks, all that is
   required is for them to deploy HTTPS servers that function as
   capability servers.  Additionally, most SIP service providers require
   enterprise networks to register with them (using a SIP REGISTER
   message) before any other SIP methods that initiate subscriptions
   (SIP SUBSCRIBE) or calls (SIP INVITE) are processed.  As a result, a
   SIP-based framework to obtain a capability set would require
   operational changes on the part of service provider networks.

   Yet another example of an alternative mechanism would be for service
   providers and enterprise equipment manufacturers to agree on YANG
   models [RFC 6020 [8]] that enable configuration to be pushed over
   NETCONF [RFC 6241 [9]] to enterprise networks from a centralised
   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 a SIP service provider, the edge
   element in an 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:

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   1.  Edge elements are devices that normalise 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 interconnection.

   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
   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.

   There could be situations wherein an enterprise telephony network
   interconnects with its SIP service provider such that traffic between
   the two networks traverses an intermediary SIP service provider
   network.  This could be a result of interconnect agreements between
   the terminating and transit SIP service provider networks.  In such
   situations, the capability set provided to the enterprise network by
   its SIP service provider must account for the characteristics of the
   transit SIP service provider network from a signalling and media
   perspective.  For example, if the terminating SIP service provider
   network supports the G.729 codec and the transit SIP service provider
   network does not, G.729 must not be advertised in the capability set.
   As another example, if the transit SIP service provider network
   doesn't support a SIP extension, for instance, the SIP extension for
   Reliable Provisional Responses as defined in RFC 3262, the
   terminating SIP service provider network must not advertise support
   for this extension in the capability set provided to the enterprise
   network.  How a terminating SIP service provider obtains the
   characteristics of the intermediary SIP service provider network is
   out of the scope of this document; however, one method could be for
   the terminating SIP service provider to obtain the characteristics of
   the intermediary SIP service provider by leveraging the YANG model
   introduced in this document.

   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

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   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 HTTPS as a transport protocol for
   the peering workflow.  This workflow is based on HTTP version 1.1,
   and as such is compatible with any future version of HTTP that is
   backward compatible with HTTP 1.1.

6.1.  HTTP Methods

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

6.2.  Integrity and Confidentiality

   Peering requests and responses are defined over HTTPS.  However, due
   to the sensitive nature of information transmitted between client and
   server, it is required to secure HTTP using Transport Layer Security
   [RFC 5246 [10]].  The enterprise edge element and capability server
   MUST be compliant with RFC 7235 [11].  The enterprise edge element
   and capability server MUST support the use of the https uri scheme as
   defined in RFC 7230 [12].

6.3.  Authenticated Client Identity

   It is only required for the SIP service provider to authenticate the
   client (enterprise edge element).  How the SIP service provider
   authenticates the client is out of the scope of this document,
   however, methods such as HTTP Digest Authentication may be used.

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

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   HTTPS GET request such that the Accept header field includes both
   MIME types along with the corresponding "qvalue" for each MIME type.

   The generated HTTPS GET request must not use the "Expect" and "Range"
   header fields.  The requests must also 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 must 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

   3.  307 Temporary Redirect

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

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.  1.  Manual Configuration

   2.  2.  Discovery [TBD]

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

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   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 [RFC 8340 [13]] 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).

   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 (":").

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

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

           +--rw peering-response
        +--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 numRange
              |     +--rw numRangeType*    string
              |     +--rw count*           int32
              |     +--rw value*           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 certLocation     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 [RFC 6991 [14]].

    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 numRange {
            leaf numRangeType {
              type string;
              description "String indicating whether the DID number range is passed
              by value or by reference"
            }

            leaf count {
              when "../numRangeType = 'range' or ../numRangeType = 'block'";
              type int32;
              description "Number of DID numbers present in the number range."
            }

            leaf-list value {
              type string;
              description "Value of the DID number range or URL being passed as
              reference."
            }

          }
        }

        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.";
            }
          }

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          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].";
            }

            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 {

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              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.";
            }
          }

          leaf certLocation {
            type string;
            description "Location of the service provider certificate chain for SIP over TLS.";
          }
        }

        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

   *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.

   *variant*: This node identifies the version number of the capability
   set document.  This draft defines the parameters for variant 1.0;

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   future specifications might define a richer parameter set, in which
   case the variant must 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.

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

   *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).

   *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 must be populated with one or
   more valid DNS server IP addresses.

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

   *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

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   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

   *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

   *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.

   *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 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.

   *signalingForking*: A leaf that specifies whether outbound call
   requests from the enterprise might be forked on the service provider
   network that MAY lead 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.

   *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

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   enumerates support for the OPTIONS method, the enterprise network
   could periodically send OPTIONS requests as a keep-alive mechanism.

   *numRange*: Is a container that specifies the Direct Inward Dial
   (DID) number range allocated to the enterprise network by the SIP
   service provider.  The DID number range allocated by the service
   provider to the enterprise network might be a contiguous or a non-
   contiguous block.  The number range allocated to an enterprise can be
   communicated as a value or as a reference.  For large enterprise
   networks, the size of the DID range might run into several hundred
   numbers.  For situations in which the enterprise is allocated a large
   DID number range or a non-contiguous number range it is RECOMMENDED
   that the SIP service provider communicate this information by
   reference, that is, through a URL.  The enterprise network is
   required to de-reference this URL in order to obtain the DID number
   range allocated by the SIP service provider.  The numRange container
   can be used more than once.  Refer to the example provided in
   Section 10.1.

   *numRangeType*: A leaf node that indicates whether the DID range is
   communicated by value or by reference.  It can have a value of
   'range', 'block' or 'reference'.

   *count*: A leaf node that indicates the size of the DID number range.
   The number range may be contiguous or non-contiguous.  This leaf node
   MUST NOT be included when using the 'reference' numRangeType value.

   *value*: A leaf-list that encapsulates the DID number range allocated
   to the enterprise.  If the numRangeType value is set to 'range' or
   'block', this is the list of numbers allocated to the enterprise.  If
   the numRangeType value is set to 'reference', this is the URL of the
   resource containing the DID number range.  To ensure ease of parsing,
   it is RECOMMENDED that the resource contain a number range formatted
   as if it were being passed as a block or range.

   *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.

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

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   *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 required and
   optional parameters for the given media format as specified when the
   media format is registered [RFC 4855 [15]].  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.

   *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.

   *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.

   *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

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   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 eavesdrop on the conversation between the offnet parties.

   *symmetricRTP*: A leaf node indicating whether the SIP service
   provider expects the enterprise network to use symmetric RTP as
   defined in RFC 4961 [16].  Uncovering this expectation is useful in
   scenarios where "latching" [RFC 7362 [17]] 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.

   *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.

   *RTCPFeedback*: A leaf node that indicates whether the SIP service
   provider supports the RTP profile extension for RTCP-based feedback
   RFC 4585 [18].  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 RFC 4585 [19] 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.

   *symmetricRTCP*: A leaf node indicating whether the SIP service
   provider expects the enterprise network to use symmetric RTCP as
   defined in RFC 4961 [20].  This node is a Boolean type, a value of 1
   indicates that the service provider expects symmetric RTCP reports,

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   whereas a value of 0 indicates that the enterprise can use asymmetric
   RTCP.

   *dtmf*: A container that describes the various aspects of DTMF relay
   via RTP Named Telephony Events.  The dtmf container allows 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 RFC 2833 [21] or RFC 4733 [22] 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.

   RFC 4733 [23] iterates over RFC 2833 [24] by introducing certain
   changes in the way NTE events are transmitted.  SIP service providers
   can indicate support for RFC 4733 [25] by setting the iteration flag
   to 1 or indicating support for RFC 2833 [26] by setting the iteration
   flag to 0.

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

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

   *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".

   *version*: A leaf node that specifies the version(s) of TLS supported
   in decimal format.  If multiple versions of TLS are supported, they
   should be separated by semi-colons.  If the service provide does not

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   support TLS for protecting SIP sessions, the signalling element is
   set to the string "NULL".

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

   *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 RFC 4568 [27] for the
   purpose of key management.  A value of "DTLS-SRTP" signifies that the
   SIP service provider uses the methods defined in RFC 5764 [28] 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".

   *certLocation:*: If the enterprise network is required to exchange
   SIP traffic over TLS with the SIP service provider, and if the SIP
   service provider is capable of accepting TLS connections from the
   enterprise network, it may be required for the SIP service provider
   certificates to be pre-installed on the enterprise edge element.  In
   such situations, the certLocation leaf node is populated with a URL,
   which when dereferenced, provides a single PEM encoded file that
   contains all certificates in the chain of trust.  This is an optional
   leaf node.

   *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

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   enterprise SBC.  The "vendorA-config" YANG module is configured as
   follows:

    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 [RFC 7950 [29]] 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.

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10.1.  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>
        <numRange>
          <type>range</type>
          <count>20</count>
          <value>19725455000</value>
        </numRange>
        <numRange>
          <type>block</type>
          <count>2</count>
          <value>1972546000</value>
          <value>1972546001</value>
        </numRange>
      </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>

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          <symmetricRTCP>true</symmetricRTCP>
          <RTCPFeedback>true</RTCPFeedback>
        </rtcp>
      </media>
      <dtmf>
        <payloadNumber>101</payloadNumber>
        <iteration>0</iteration>
      </dtmf>
      <security>
        <signaling>
          <type>TLS</type>
          <version>1.0;1.2</version>
        </signaling>
        <mediaSecurity>
          <keyManagement>SDES;DTLS-SRTP,version=1.2</keyManagement>
          </mediaSecurity>
        <certLocation>https://sipserviceprovider.com/certificateList.pem</certLocation>
        </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 has been pre-configured with the request
   target for the capability set document or has dynamically found the
   request target, the edge element generates a HTTPS GET request.  This
   request can be challenged by the service provider to authenticate the
   enterprise.

       GET //capdoc?trunkid=trunkent1456 HTTP/1.1
       Host: capserver.ssp1.com
       Accept:application/peering-info+xml

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

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

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

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12.  Security Considerations

   [TBD]

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>.

   [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>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <https://www.rfc-editor.org/info/rfc3261>.

   [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>.

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   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246,
              DOI 10.17487/RFC5246, August 2008,
              <https://www.rfc-editor.org/info/rfc5246>.

   [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>.

   [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>.

   [RFC6749]  Hardt, D., Ed., "The OAuth 2.0 Authorization Framework",
              RFC 6749, DOI 10.17487/RFC6749, October 2012,
              <https://www.rfc-editor.org/info/rfc6749>.

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

   [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>.

   [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>.

   [RFC7362]  Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT
              Traversal (HNT) for Media in Real-Time Communication",
              RFC 7362, DOI 10.17487/RFC7362, September 2014,
              <https://www.rfc-editor.org/info/rfc7362>.

   [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>.

   [RFC8446]  Rescorla, E., "The Transport Layer Security (TLS) Protocol
              Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
              <https://www.rfc-editor.org/info/rfc8446>.

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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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