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Automatic Peering for SIP Trunks
draft-ietf-asap-sip-auto-peer-30

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RFC 10006

The information below is for an old version of the document.

Document	Type	This is an older version of an Internet-Draft that was ultimately published as RFC 10006.
	Authors	Kaustubh Inamdar , Sreekanth Narayanan , Cullen Fluffy Jennings
	Last updated	2025-09-11 (Latest revision 2025-08-09)
	Replaces	draft-kinamdar-dispatch-sip-auto-peer
	RFC stream	Internet Engineering Task Force (IETF)
	Formats	txt html xml htmlized bibtex bibxml
	Reviews	OPSDIR IETF Last Call review (of -32) by Jen Linkova Ready YANGDOCTORS IETF Last Call review by Ebben Aries Almost ready ARTART Telechat review (of -23) by Harald Alvestrand Ready w/issues YANGDOCTORS Telechat review (of -23) by Ebben Aries Not ready ARTART Telechat review (of -21) by Harald Alvestrand Ready w/issues TSVART Telechat review (of -20) by Joerg Ott Ready w/nits GENART IETF Last Call review (of -18) by Joel Halpern Ready w/issues SECDIR Telechat review (of -16) by Dan Harkins Ready
	Additional resources	Mailing list discussion
Stream	WG state	In WG Last Call
	Document shepherd	Marc Petit-Huguenin
	Shepherd write-up	Show Last changed 2025-02-02
IESG	IESG state	Became RFC 10006 (Proposed Standard)
	Consensus boilerplate	Yes
	Telechat date	(None) Has enough positions to pass.
	Responsible AD	Andy Newton
	Send notices to	snandaku@cisco.com, marc@petit-huguenin.org
IANA	IANA review state	Version Changed - Review Needed

Email authors Email WG IPR References Referenced by Nits Nits v3 Search email archive

draft-ietf-asap-sip-auto-peer-30

ASAP K. Inamdar
Internet-Draft S. Narayanan
Intended status: Standards Track Unaffiliated
Expires: 10 February 2026 C. Jennings
Cisco Systems
9 August 2025

Automatic Peering for SIP Trunks
draft-ietf-asap-sip-auto-peer-30

Abstract

This document specifies a framework that enables enterprise telephony
Session Initiation Protocol (SIP) networks to solicit and obtain a
capability set document from a SIP service provider. The capability
set document encodes a set of characteristics that enable easy
peering between enterprise and service provider SIP 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 10 February 2026.

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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 Revised BSD License text as
described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Revised BSD License.

Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Overview of Operations . . . . . . . . . . . . . . . . . . . 4
2.1. Reference Architecture . . . . . . . . . . . . . . . . . 4
2.2. Configuration Workflow . . . . . . . . . . . . . . . . . 6
2.3. Transport . . . . . . . . . . . . . . . . . . . . . . . . 7
3. Conventions and Terminology . . . . . . . . . . . . . . . . . 8
4. HTTP Transport . . . . . . . . . . . . . . . . . . . . . . . 8
4.1. HTTP Methods . . . . . . . . . . . . . . . . . . . . . . 8
4.2. Integrity and Confidentiality . . . . . . . . . . . . . . 8
4.3. Authenticated Client Identity . . . . . . . . . . . . . . 9
4.4. Encoding the Request . . . . . . . . . . . . . . . . . . 11
4.5. Identifying the Request Target . . . . . . . . . . . . . 11
4.6. Generating the response . . . . . . . . . . . . . . . . . 13
5. State Deltas . . . . . . . . . . . . . . . . . . . . . . . . 13
6. Encoding the Service Provider Capability Set . . . . . . . . 14
7. Data Model for Capability Set . . . . . . . . . . . . . . . . 14
7.1. Tree Diagram . . . . . . . . . . . . . . . . . . . . . . 14
7.2. YANG Model . . . . . . . . . . . . . . . . . . . . . . . 16
7.3. Extending the Capability Set . . . . . . . . . . . . . . 36
8. Processing the Capability Set Response . . . . . . . . . . . 37
9. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 38
9.1. JSON Capability Set Document . . . . . . . . . . . . . . 38
9.2. Example Exchange . . . . . . . . . . . . . . . . . . . . 41
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42
11. Security Considerations . . . . . . . . . . . . . . . . . . . 42
11.1. OAuth Credentials . . . . . . . . . . . . . . . . . . . 42
11.2. Client-Server Communication . . . . . . . . . . . . . . 43
12. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 43
13. Informative References . . . . . . . . . . . . . . . . . . . 43
14. Normative References . . . . . . . . . . . . . . . . . . . . 44
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46

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1. Introduction

The deployment of a Session Initiation Protocol [RFC3261] (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 conventional 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.

The SIP Connect technical recommendations [SIP-Connect-TR] aim to
solve this problem by providing a central 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 interoperability 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 network, 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 automation or
an administrator to generate configuration blocks across one or more
devices within the enterprise network to ensure successful service
provider peering.

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2. Overview of Operations

This section provides a reference architecture against which the SIP
Auto Peer framework may be implemented. Additionally, terms that are
commonly used in the context of the document are defined. Lastly,
considerations for the choice of network transport between enterprise
and service provider telephony networks are discussed.

2.1. 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 (M.E.) and a Session
Border Controller [RFC7092]. 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 for handling media streams
of calls setup by the SP-SSE and a HTTPS [RFC9110] server.

+-----------------------------------------------------+
| +---------------+ +-----------------------+ |
| | | | | |
| | +----------+ | | +-------+ | |
| | | 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:

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* 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.

* 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) [RFC7092] such as a Session Border
Controller (SBC).

* 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.

* 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, such as,
registration of endpoints, integration with application servers
(voicemail, instant messaging, presence), among others.

* 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.

* Capability Set: The term capability set (or capability set
document) refers collectively 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.

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2.2. 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:

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

* 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.

* Capability set documents obtained as a result of the configuration
workflow must be conducive to easy parsing by automation.
Subsequently, automation may be used for the generation of
appropriate configuration blocks on the edge element or across one
or more elements in the enterprise network.

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 HTTP 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 JSON,
thus ensuring that the response can be easily parsed by automation.

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 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

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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 [RFC6020] that enable configuration to be pushed over NETCONF
[RFC6241] 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.

2.3. Transport

To solicit the capability set of a SIP service provider, the edge
element in an enterprise network generates a well-formed HTTP 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 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 HTTP 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 JSON. More details about the formatting of
the HTTP request and response are provided in Section 4.

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

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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.

3. Conventions and Terminology

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.

4. HTTP Transport

This section describes the use of HTTPS as a transport protocol for
the peering workflow. This workflow is based on HTTP/1.1, and as
such is compatible with any future version of HTTP that is backward
compatible with HTTP/1.1 including HTTP/3 [RFC9114].

4.1. 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.

4.2. Integrity and Confidentiality

Peering requests and responses are defined over HTTP [RFC9110].
However, due to the sensitive nature of information transmitted
between client and server, it is required to secure HTTP
communications using Transport Layer Security (TLS) [RFC8446];
therefore the enterprise edge element and the capability server MUST
support TLS. When HTTP/3 is used, TLS is incorporated within QUIC.
Additionally, the enterprise edge element and capability server MUST

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support the use of the HTTPS URI scheme as defined in [RFC9110].

4.3. Authenticated Client Identity

HTTP usually adopts asymmetric methods of authentication. For
example, clients typically use certificate based authentication to
verify the server they are talking to, whereas, servers typically use
methods such as HTTP digest authentication or OAuth 2.0 [RFC6749] to
authenticate clients. Though OAuth 2.0 is not an authentication
protocol, it nonetheless allows for client authentication to be
carried out with the use of OAuth tokens.

In the context of the SIP Auto Peer framework, OAuth 2.0 MUST be used
to carry out client authentication. Enterprise edge elements could
use the various grant types outlined in the OAuth 2.0 specification
and supported by the service provider in order to obtain the
capability set document. This draft does not mandate a specific
grant type. The implementation of OAuth 2.0 to obtain the capability
set are beyond the scope of this document. However, it provides an
example of how an enterprise SBC could leverage the "Authorization
Code Grant" (Section 4.1 of [RFC6749]) flow to acquire the capability
set document from the service provider in Figure 2.

Using the "Resource Owner Password Credentials" grant type
(Section 1.3.3 of [RFC6749]) requires the existence of a trust
relationship between the resource owner (in this context, the
administrator/enterprise network) and the client (in this context, an
edge element such as an SBC). In SIP trunking deployments between
enterprise and service provider networks, such a trust relationship
between the administrator/resource owner/enterprise network and the
client (edge element) already exists, as SIP trunk registration (and
refreshing registrations) require credentials - typically a username
and password, that are configured on the edge element by the
administrator. However, it is important for the enterprise network
administrator and service provider to factor in security issues
associated with this grant type.

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+---------------+
| Resource |
| Owner |
| (Enterprise) |
+---------------+
^
|
(B)
+----|-----+ Client Identifier +---------------+
| -+----(A)-- & Redirection URI ---->| Service |
| User- | | Provider |
| Agent -+----(B)-- User authenticates --->| Authorization |
| | | Server |
| -+----(C)-- Authorization Code ---<| |
+-|----|---+ +---------------+
| | ^ v
(A) (C) | |
| | | |
^ v | |
+---------+ | |
| |>---(D)-- Authorization Code ---------' |
| Client | & Redirection URI |
| (SBC) | |
| |<---(E)----- Access Token -------------------'
+---------+ (w/ Optional Refresh Token)
^ v
| |
| | +--------------+
| -------(F)---- Access Token --------->| Capability |
-----------(G)---- Capability set -------<| Server |
+--------------+

Figure 2: Client Authentication Mechanism

The flow illustrated in Figure 2 includes the following steps:

A. The enterprise SBC (client) initiates the flow by directing the
resource owner's (enterprise network administrator) user-agent to
the authorization endpoint. The SBC includes its client
identifier, requested scope, local state, and a redirection URI
to which the authorization server will send the user-agent back
once access is granted (or denied). As a precursor to the flow,
the enterprise network administrator has already obtained a
unique client identifier for their network and provided a
redirection URI populated with a target within their network to
obtain the authorization code.

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B. The authorization server within the service provider network
authenticates the network administrator (via the user-agent) and
establishes whether the network administrator grants or denies
the client's access request.

C. Assuming the network administrator grants access, the
authorization server redirects the user-agent back to the
enterprise SBC using the redirection URI provided earlier (in the
request or during client registration). The redirection URI
includes an authorization code and any local state provided by
the client earlier.

D. The enterprise SBC requests an access token from the
authorization server's token endpoint by including the
authorization code received in the previous step. When making
the request, the enterprise SBC authenticates with the
authorization server and includes the redirection URI used to
obtain the authorization code for verification.

E. The authorization server authenticates the enterprise SBC,
validates the authorization code, and ensures that the
redirection URI received matches the URI used to redirect the SBC
in step (C). If valid, the authorization server responds back
with an access token and, optionally, a refresh token.

F. The enterprise SBC then contacts the capability server located in
the service provider network with an HTTP GET request along with
the access token to retrieve the capability set document.

G. The capability server checks for a valid access token and returns
the capability set document to the enterprise SBC. The service
provider will host a unique document for each enterprise network
that will peer with it.

4.4. Encoding the Request

The edge element in the enterprise network generates a HTTP GET
request such that the request-target is obtained using the procedure
outlined in section 4.5. This document does not specify any content
negotiation. The server MUST set the response content type header to
the application/json media type.

4.5. Identifying the Request Target

HTTP 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:

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1. Manual Configuration

2. Discovery 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, to the edge element), the WebFinger protocol [RFC7033] and
the 'sipTrunkingCapability' [RFC9409] link relation type may be
leveraged assuming that the service SIP service provider has
implemented WebFinger within their network and hosts the capability
set at the respective location.

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).

GET /.well-known/webfinger?
resource=http%3A%2F%2Fssp1.example.com
rel=sipTrunkingCapability
HTTP/1.1
Host: ssp1.example.com

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/jrd+json

{
"subject" : "http://ssp1.example.com",
"links" :
[
{
"rel" : "sipTrunkingCapability",
"href" :
"https://capserver.ssp1.com/capserver/capdoc.json"
}
]
}

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Once the target URI is obtained by an enterprise telephony network,
the URI may be dereferenced to obtain a unique capability set
document that is specific to that given enterprise telephony network.
The ITSP may use credentials to determine the identity of the
enterprise telephony network and provide the appropriate capability
set document.

4.6. Generating the response

Capability servers include the capability set documents in the body
of a successful response. Capability set documents MUST be formatted
in JSON. For requests that are incorrectly formatted, an example
being an incorrect query parameter in the URI, the capability server
must generate a "400 Bad Request" response for the incorrect request.
If requests contain an invalid token, the capability server must
generate a "403 Forbidden" response clearly indicating that this
token does not have the permission to view the capability set
document.

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. If the Location header isn't included in the response,
this can lead to the client being unable to find the capability set
document, leading to a failure in the peering process or requiring
manual intervention by an administrator.

5. 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.

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6. 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.

7. Data Model for Capability Set

This section defines a YANG module [RFC7950] for encoding the service
provider capability set. Section 7.1 provides the tree diagram,
which is followed by a description of the various nodes within the
module defined in this draft.

7.1. Tree Diagram

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

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

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

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

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

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

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

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7.2. YANG Model

This section defines the YANG module for the peering capability set
document. This module depends on existing YANG modules that provide
common YANG data types [RFC6991] and system management [RFC7317].

<CODE BEGINS> file "ietf-sip-auto-peering@2025-08-10.yang"
module ietf-sip-auto-peering {
yang-version 1.1;
namespace "urn:ietf:params:xml:ns:yang:ietf-sip-auto-peering";
prefix "sipap";

import ietf-inet-types {
prefix "inet";
reference
"RFC 6991: Common YANG Data Types";
}

import iana-crypt-hash {
prefix "ianach";
reference
"RFC 7317: A YANG Data Model for System Management";
}

organization
"IETF ASAP (Automatic SIP trunking And Peering) Working Group";

contact

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"WG Web: <https://datatracker.ietf.org/wg/asap/>
WG List: <mailto:asap@ietf.org>

Editor: Kaustubh Inamdar
<mailto:kaustubh.ietf@gmail.com>

Editor: Sreekanth Narayanan
<mailto:sknth.n@protonmail.com>

Editor: Cullen Jennings
<mailto:fluffy@iii.ca>";

description
"Data model for encoding SIP Service Provider Capability Set

This YANG module defines a read-only data model intended for
exchanging SIP service provider capabilities with enterprise
networks. The data is published by service providers and
consumed by enterprises via standard YANG-based interfaces
(RESTCONF, NETCONF, etc.).

This module does NOT provide configuration capabilities - it
serves purely as a standardized format for capability exchange.
Service providers generate and host capability documents based
on this schema, which enterprises retrieve and use to configure
their SIP equipment.

Redistribution and use in source and binary forms, with or
without modification, is permitted pursuant to, and subject to
the license terms contained in, the Revised BSD License set
forth in Section 4.c of the IETF Trust's Legal Provisions
Relating to IETF Documents
(https://trustee.ietf.org/license-info).

This version of this YANG module is part of RFC XXXX
(https://www.rfc-editor.org/info/rfcXXXX); see the RFC itself
for full legal notices.

The key words 'MUST', 'MUST NOT', 'REQUIRED', 'SHALL', 'SHALL
NOT', 'SHOULD', 'SHOULD NOT', 'RECOMMENDED', 'NOT RECOMMENDED',
'MAY', and 'OPTIONAL' in this document are to be interpreted as
described in BCP 14 (RFC 2119) (RFC 8174) when, and only when,
they appear in all capitals, as shown here.";

revision 2025-10-08 {

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description "Initial version";
reference
"NOTE TO RFC EDITOR: Please replace 'RFC XXXX' with the actual
RFC number of this document when published, and delete this
sentence. Also replace the revision with the date of publication
of this document.
RFC XXXX: Automatic Peering for SIP Trunks";
}

identity sip-extension {
description
"Base identity for SIP extensions/option tags as defined by IANA
SIP Parameters registry.";
}

identity reliable-provisional-responses {
base sip-extension;
description
"This extension indicates support for reliable provisional
responses as defined in RFC 3262.";
reference "RFC 3262";
}

identity session-timers {
base sip-extension;
description
"This extension indicates support for session timers
as defined in RFC 4028.";
reference "RFC 4028";
}

identity replaces {
base sip-extension;
description
"This extension indicates support for the Replaces header
as defined in RFC 3891.";
reference "RFC 3891";
}

identity path {
base sip-extension;
description
"This extension indicates support for the Path header
as defined in RFC 3327.";
reference "RFC 3327";
}

grouping entity {

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description
"Grouping that provides a reusable list named
'entity', with each entry containing a host and a port.";

leaf host {
type union {
type inet:ip-address;
type inet:domain-name;
}
description
"IP Address or host name of the entity";
}

leaf port {
type inet:port-number;
description "Entity's port number.";
}
}

container sip-auto-peering {
config false;
description
"Root container for SIP service provider capability
data. This container holds read-only operational data that
represents the capabilities and requirements of a SIP service
provider. Enterprise networks retrieve this data to automatically
configure their SIP trunking parameters.";

leaf index {
type uint8;
description
"Index for the peering-info document.";
}

leaf variant {
type enumeration {
enum v1_0 {
description
"Variant 1.0 of the capability set document is defined in
this draft";
}
}
mandatory true;
description
"A node that 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 must be changed to

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2.0, 3.0 and so on. Future extensions to the capability set
document MUST also ensure that the corresponding YANG module
is defined.";
}

container revision {
description
"A container that encapsulates information
regarding the availability of a new version of the
capability set document for the enterprise.";

leaf not-before {
type uint32;
units "seconds";
mandatory true;
description
"A node that identifies the unix epoch time at which the
parameters in this capability set document are activated
or considered valid. This node has been set to mandatory
as it is the service provider's responsibility to inform
when new peering settings take effect. Without being aware
of a start time, the enterprise network will experience
failures.";
}

leaf location {
type inet:uri;
mandatory true;
description "A node that identifies the URL of a new
revision of the service provider capability set document.
Without this URL, an enterprise network wouldn't be aware
of changes that have occurred in the service provider
network.";
}
}

container transport-info {
description
"A container that encapsulates transport
characteristics of SIP sessions between enterprise and
service provider networks.";

leaf-list transport {
type enumeration {
enum tcp {
description
"Transmission Control Protocol";
}

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enum tls {
description
"Transport Layer Security (over TCP)";
}
enum udp {
description
"User Datagram Protocol";
}
}
min-elements 1;
description
"A list that enumerates the different Transport
Layer protocols supported by the SIP service provider.
Valid transport layer protocols include: UDP, TCP and TLS";
}

list registrar {
key "host port";
uses entity;
max-elements 3;
description
"A 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.";
}

list realms {
key "name";
description
"A container that encapsulates the set of realms
or protection domains the SIP service provider is
responsible for.";

leaf name {
type string;
mandatory true;
description
"A node specifying the SIP service provider
realm or protection domain. This node is encoded as a
string; the value of this node must be identical to the

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value of the 'realm' parameter in a WWW-Authenticate
header field that the SIP service provider might send in
response to requests that do not contain a valid
Authorisation header field.";
}

leaf username {
type string;
description
"A node that encodes the username for the
given realm. The username is one of many inputs used by
the enterprise network in generating the response
parameter of the Authorization header field.";
}

leaf password {
type ianach:crypt-hash;
description
"A node that encodes the password for the
given realm. The password is one of many inputs used by
the enterprise network in generating the response
parameter of the Authorization header field. The
password is stored as a cryptographic hash.";
}
}

list call-control {
key "host port";
uses entity;
max-elements 3;
description
"A 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.";
}

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leaf-list dns {
type inet:ip-address;
max-elements 2;
description
"A 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 call-control lists or both
are populated with a valid IP address and port pair, the dns
element can be omitted.";
}

list outbound-proxy {
key "host port";
uses entity;
description
"A 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 list is populated with
a valid transport address, it represents the default
destination for all outbound SIP requests and therefore, the
registrar and call-control lists can be omitted.";
}
}

container call-specs {
description
"A container that encapsulates information about
call specifications, restrictions and additional handling
criteria for SIP calls between the enterprise and service
provider network.";

leaf early-media {
type boolean;
description
"A node that specifies whether the service
provider network is expected to deliver in-band
announcements/tones before call connect. The

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'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 true signifies
that the service provider is capable of early media. A
value of false signifies that the service provider is not
expected to generate early media.";
}

leaf signaling-forking {
type boolean;
description
"A node 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 boolean type, where a value of true signifies
that the service provider network can potentially fork
outbound call requests from the enterprise. A value of
false indicates that the service provider will not fork
outbound call requests.";
}

leaf-list supported-methods {
type enumeration {
enum invite {
description "Initiate a dialog or session.";
}
enum ack {
description "Acknowledge final response to INVITE.";
}
enum bye {
description "Terminate a dialog or session.";
}
enum cancel {
description "Cancel a pending request.";
}
enum register {
description "Register contact information.";
}

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enum options {
description "Query capabilities of a server.";
}
enum prack {
description "Provisional acknowledgement.";
}
enum subscribe {
description "Subscribe to an event.";
}
enum notify {
description "Notify subscriber of an event.";
}
enum publish {
description "Publish an event state.";
}
enum info {
description "Send mid-session information.";
}
enum refer {
description "Refer recipient to a third party.";
}
enum message {
description "Instant message transport.";
}
enum update {
description
"Update session parameters within a dialog.";
}
}
description
"A list that specifies the various SIP methods
supported by the SIP service provider. The list of
supported methods help to appropriately configure
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.";
}

container caller-id {
description
"A container that encodes the preferences of SIP
service providers in terms of calling number presentation
by the enterprise network. Certain ITSPs require that the
calling number be formatted in E.164, whereas others place
no such restrictions. Additionally, some ITSPs require
that the calling number be included in a specific SIP

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header field, for example, the P-Asserted-ID header field
or the From header field, whereas others place no
restrictions on the specific SIP header field used to
convey the calling number.";

leaf e164-format {
type boolean;
description
"A node that indicates whether the service
provider requires the enterprise network to normalize
the calling number into E.164 format. A value of true
mandates the enterprise network to format calling
numbers to E.164 format, while a false leaves the
formatting of the calling number up to the enterprise
network.";
}

leaf preferred-method {
type enumeration {
enum p-asserted-identity {
description
"Use the 'P-Asserted-Identity' header to
determine remote party identity.";
}
enum from {
description
"Use the 'From' header to determine remote
party identity.";
}
}
description
"A node that specifies which SIP header MUST
be used by the enterprise network to communicate caller
information. The value of this node is a string that
contains the name of the SIP header required to carry
caller information.";
}
}

list number-range {
key index;
description
"A list that specifies the Direct Inward Dial (DID) number
range allocated to the enterprise network by the SIP service
provider. The DID number ranges allocated by the service
provider to the enterprise network might be a contiguous or a
non-contiguous block. The number ranges allocated to an
enterprise can be communicated as a value or as a reference.

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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 ranges
allocated by the SIP service provider. Refer to the example
provided in Section 9.1.";

leaf index {
type uint16;
description
"Index for the number ranges.";
}

leaf type {
type enumeration {
enum range {
description
"Numbers specified as a range.";
}
enum collection {
description
"Numbers specified in the form of a collection.";
}
enum reference {
description
"Number range available at a URL.";
}
}
description
"A node that indicates whether the DID range
is communicated by value or by reference. It can have a
value of 'range', 'collection' or 'reference'.";
}

leaf count {
when "../type = 'range' or ../type = 'collection'";
type uint16;
description
"Indicates the size of the DID number range.
This leaf MUST NOT be included when using the
'reference' type.";
}

leaf-list value {
type string;

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description
"A list that encapsulates the DID number range
allocated to the enterprise. If the num-ranges 'type' is
set to 'range' or 'collection', the 'count' node MUST
have a valid, non-zero, positive integer. If the
num-ranges 'type' value is set to 'range', then, the
number in this field represents the first phone number
of a DID range allocated to the enterprise. The value
of subsequent numbers of the given DID range are
obtained by adding one to the value of this field. The
number of times we need to add one is indicated by the
'count' field.";
}
}
}

container media {
description
"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. At the time of writing this specification,
video media streams aren't exchanged between enterprise and
service provider SIP networks.";

list media-type-audio {
key "media-format";
description
"A list encoding the various audio media formats
supported by the SIP service provider. The relative
ordering of different media formats in the list indicates
preference from the perspective of the service provider.
Each element in the list 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 the sampling rate for the
encoding and the packetization time. Additionally, any
other required and optional parameters for the given media
format as specified when the media format is registered
[@RFC4855] are described the 'param' field.
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

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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.";

leaf media-format {
type enumeration {
enum pcmu {
description
"PCMU format.";
}
enum g722 {
description
"G722 format.";
}
enum g729 {
description
"G729 format.";
}
}
description
"The audio media format.";
}

leaf rate {
type uint16;
units "Hz";
description
"Sampling rate in Hz.";
}

leaf ptime {
type uint8;
units "milliseconds";
description
"Packetization time in milliseconds.";
}

leaf parameter {
type string;
description
"Optional parameter for additional media details regarding
the encoding.";
}
}

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

leaf-list protocol {
type enumeration {
enum pass-through {
description
"Protocol-based fax passthrough.";
}
enum t38 {
description
"T38 relay.";
}
}
max-elements 2;
description
"List indicating the different fax protocols
supported by the service provider.";
}
}

container rtp {
description
"A container that encapsulates generic
characteristics of RTP sessions between the enterprise
and service provider network.";

leaf rtp-trigger {
type boolean;
description
"A 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

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callee. This node is a boolean type. A value of true
indicates that the service provider expects the
enterprise network to send the first RTP packet, whereas
a value of 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 eavesdrop on the conversation
between the offnet parties.";
}

leaf symmetric-rtp {
type boolean;
description
"A node indicating whether the SIP service
provider expects the enterprise network to use symmetric
RTP as defined in [@RFC4961]. Enforcement of this
requirement by service providers on enterprise networks
is typically useful in scenarios such as media latching
[@RFC7362]. This node is a boolean type, a value of true
indicates that the service provider expects the
enterprise network to use symmetric RTP, whereas a value
of false indicates that the enterprise network can use
asymmetric RTP.";
}
}

container rtcp {
description
"A container that encapsulates generic
characteristics of RTCP sessions between the enterprise
and service provider network.";

leaf symmetric-rtcp {
type boolean;
description
"A 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 true indicates that the service
provider expects symmetric RTCP reports, whereas a
value of false indicates that the enterprise can use
asymmetric RTCP.";
}

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leaf rtcp-feedback {
type boolean;
description
"A 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 of
boolean type, a value of true indicates that the service
provider supports the RTP profile extension for
RTP-based feedback and a value of false indicates that
the service provider does not support the RTP profile
extension for RTP-based feedback.";
}
}
}

container dtmf {
description
"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.";

leaf payload-number {
type uint8 {
range "96..127";
}
description
"Indicates the payload type number.";
}

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leaf iteration {
type boolean;
description
"A value of true indicates that the service
provider supports RFC4733 while a value of false indicates
that the service provider prefers RFC2833";
}
}

container security {
description
"A container that encapsulates characteristics
about encrypting signalling streams between the enterprise
and SIP service provider networks.";

container signaling {
description
"A container that encapsulates the type of
security protocol for the SIP communication between the
enterprise SBC and the service provider.";

leaf secure {
type boolean;
description
"A node that specifies whether the service
provider allows the use of TLS to secure SIP signalling
messages between the enterprise and service provider
network. This node is of boolean type, a value of true
indicates that the service provider supports SIP
sessions over TLS, wheras a value of false indicates
that the service provider does not support SIP over
TLS.";
}

leaf-list version {
when "../secure = 'true'";
type enumeration {
enum v1_2 {
description
"TLS version 1.2.";
}
enum v1_3 {
description
"TLS version 1.3.";
}
}
description
"A list that specifies the version(s) of TLS supported.";

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}
}

container media-security {
description
"A container that describes the various
characteristics of securing media streams between
enterprise and service provider networks.";

leaf-list key-management {
type enumeration {
enum sdes {
description
"Simplified Data Encryption Standard
key management.";
}
enum dtls-srtp {
description
"SRTP keys managed using DTLS.";
}
}
description
"A list that specifies the key management
method(s) used by the service provider. Possible values
in this list 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.";
}
}

leaf cert-location {
type inet:uri;
description
"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 cert-location node is populated with
a URL, which when dereferenced, provides a single PEM
encoded file that contains all certificates in the chain
of trust.";
}

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container secure-telephony-identity {
description
"Encapsulates Secure Telephony Identity (STIR)
characteristics.";

leaf stir-compliance {
type boolean;
description
"A node that indicates whether the SIP service
provider is STIR compliant. This node is of boolean
type, a value of true indicates that the SIP service
provider is STIR compliant. A value of false indicates
that the SIP service provider is not STIR compliant. A
SIP service provider being STIR compliant has
implications for inbound and outbound calls, from the
perspective of the enterprise network.";
}

leaf cert-delegation {
type boolean;
description
"A node that indicates whether a SIP service
provider that allocates one or more number ranges to an
enterprise network, is willing to delegate authority to
the enterprise network over that number range(s). This
node is of boolean type, a value of true indicates that
the SIP service provider is willing to delegate authority
to the enterprise network over one or more number
ranges. A value of false indicates that the SIP service
provider is not willing to delegate authority to the
enterprise network over one or more number ranges. This
node MUST only be included in the capability set if the
value of the stir-compliance leaf node is set to true.
In order to obtain delegate certificates, the enterprise
network must be made aware of the scope of delegation -
the number or number range(s) over which the SIP service
provider is willing to delegate authority. This
information is included in the num-ranges container.";
}

leaf acme-directory {
when "../cert-delegation = 'true'";
type inet:uri;
description
"A node that provides the URL of the directory
object for delegate certificates using Automatic
Certificate Management Environment (ACME) [@RFC8555].
The directory object URL, when de-referenced, provides a

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collection of field name-value pairs. Certain field
name-value pairs provided in the response are used to
bootstrap the process the obtaining delegate
certificates. This node MUST only be included in the
capability set if the value of the cert-delegation
leaf node is set to true.";
}
}
}

leaf-list extensions {
type identityref {
base sip-extension;
}
description
"A list of SIP option tags (extensions) supported by the service
provider network. Each extension is represented as an identity
derived from the sip-extension base identity. This provides
type safety and allows for proper validation of supported
extensions.";
}
}
}
<CODE ENDS>

7.3. 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,
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.

8. Processing the Capability Set Response

This section provides a non-normative description of the procedures
that could be carried out by the enterprise network after obtaining
the SIP service provider capability set. On obtaining the capability
set, the enterprise edge element can parse the various fields within
the capability set and generate configuration blocks. For example,
the configuration required to successfully register a SIP trunk with
the SIP registrar hosted in the service provider network, the
configuration required to ensure that fax calls are handled

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appropriately, the configuration required to advertise only audio
codecs supported by the SIP service provider, among many other
configuration blocks. A configuration block generated for an almost
identical SIP service provider capability set document is likely
going to differ drastically from one vendor to the next.

Enterprise edge elements are usually capable of normalising
mismatches in the signalling and media planes between the enterprise
and service provider SIP networks. As a result, most, if not all of
the configuration blocks required to enable successful SIP service
provider peering might need to be added on the edge element. In
situations wherein configuration blocks need to be distributed across
multiple devices, some mechanism, that is out of scope of this
document might be used to communicate the specific fields of capacity
set and their corresponding value. Alternatively, a human
administrator could go through the capability set document and
configure the edge element (and if required, other devices in the
enterprise network appropriately.

9. Examples

This section provides examples of how capability set documents that
leverage the YANG module defined in this document can be encoded over
JSON as well as the exchange of messages between the enterprise edge
element and the service provider to acquire the capability set
document. The service provider will create a unique document for
each enterprise network that will peer with it.

9.1. JSON Capability Set Document

<CODE BEGINS> file "asap-example.json"
{
"ietf-sip-auto-peering:sip-auto-peering":
{
"index": 0,
"variant": "v1_0",
"revision": {
"not-before": 1742330340,
"location":
"https://capserver.example.org/capserver/capdoc.json"
},
"transport-info": {
"transport": [
"tcp",
"tls",
"udp"
],
"registrar": [

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{
"host": "registrar1.voip.example.com",
"port": 5060
},
{
"host": "registrar2.voip.example.com",
"port": 5060
}
],
"realms": [
{
"name": "voip.example.com",
"username": "voip",
"password": "$6$OoEJwExxp6U/FRFq$4RkL2lSSGLoKdfGjX4lQLFXo89gc0wtJsKiBxg/BBz6aNwu7C.D3kRUwD7lvJm6rhaCdhSzVh/XfkkAUY2dTu0"
}
],
"call-control": [
{
"host": "callServer1.voip.example.com",
"port": 5060
},
{
"host": "192.0.2.40",
"port": 5065
}
],
"dns": [
"192.0.2.50",
"192.0.2.51"
],
"outbound-proxy": [{
"host": "192.0.2.35",
"port": 5060
}]
},
"call-specs": {
"early-media": true,
"signaling-forking": false,
"supported-methods": [
"invite",
"options",
"bye",
"cancel",
"ack",
"prack",
"subscribe",
"notify",
"register"

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],
"caller-id": {
"e164-format": true,
"preferred-method": "from"
},
"number-range": [
{
"index": 0,
"type": "range",
"count": 20,
"value": [
"19725455000"
]
},
{
"index": 1,
"type": "collection",
"count": 2,
"value": [
"19725455000",
"19725455001"
]
}
]
},
"media": {
"media-type-audio": [
{
"media-format": "pcmu",
"rate": 8000,
"ptime": 20
},
{
"media-format": "g729",
"rate": 8000,
"ptime": 20,
"parameter": "annexb"
}
],
"fax": {
"protocol": [
"t38",
"pass-through"
]
},
"rtp": {
"rtp-trigger": true,
"symmetric-rtp": true

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},
"rtcp": {
"symmetric-rtcp": true,
"rtcp-feedback": true
}
},
"dtmf": {
"payload-number": 101,
"iteration": false
},
"security": {
"signaling": {
"secure": true,
"version": ["v1_2", "v1_3"]
},
"media-security": {
"key-management": ["sdes", "dtls-srtp"]
},
"cert-location":
"https://sipserviceprovider.com/certificateList.pem",
"secure-telephony-identity": {
"stir-compliance": true,
"cert-delegation": true,
"acme-directory": "https://sipserviceprovider.com/acme.html"
}
},
"extensions": [
"ietf-sip-auto-peering:reliable-provisional-responses",
"ietf-sip-auto-peering:session-timers",
"ietf-sip-auto-peering:replaces",
"ietf-sip-auto-peering:path"
]
}
}
<CODE ENDS>

9.2. Example Exchange

This section is an informational example depicting 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 HTTP GET request. This
request can be challenged by the service provider to authenticate the
enterprise.

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GET /capdoc?trunkid=trunkent1456 HTTP/1.1
Host: capserver.ssp1.com
Authorization: Bearer <clientToken>

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

HTTP/1.1 200 OK
Content-Type: application/json
Content-Length: nnn

{
"peering-info": ...
}

10. IANA Considerations

This document has no IANA actions.

11. Security Considerations

The capability set document contains sensitive information that must
be protected from attackers. A capability set document leak can
inflict considerable damage to both the enterprise as well as the
service provider. An attacker that gains access to the capability
set document can cause problems in multiple ways.

There are multiple attack points in the ASAP workflow. The sections
below deal with the different points at which the workflow is
vulnerable to attackers.

11.1. OAuth Credentials

In scenarios wherein client authentication is carried out using OAuth
resource owner credentials, it is required to ensure that these
credentials cannot be acquired by any unauthorized third-party. If
acquired by an unauthorized third-party, these credentials may be
used to obtain the capability set document from the SIP service
provider and subsequently use the information in such a document to
make unauthorized calls while posing as an enterprise telephony
network that has legitimately paid for calling services from a SIP
service provider.

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11.2. Client-Server Communication

All communication used by the edge element to obtain the capability
set document from the capability server MUST be secured using HTTPS.
Failure to do so, results in the capability set document being
transmitted over clear text, thus exposing sensitive information such
as targets for trunks registration, targets for outbound calling
requests and credentials used in building the Authorisation header
field provided in response to authentication challenges.

12. Acknowledgments

We would like to thank those who provided detailed and thoughtful
comments on this draft, especially Marc Petit-Huguenin, Paul Jones,
Ram Mohan R, Nicola Serafini, Jonathan Rosenberg, Jon Peterson, Chris
Wendt and Henning Schulzrinne. Additional thanks to Murray
Kucherawy, Joel Halpern, Dan Harkins, Éric Vyncke, Joerg Ott, Mahesh
Jethanandani, Orie Steele, Harald Alvestrand and Ebben Aries for
their reviews and feedback.

13. 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>.

[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>.

[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>.

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[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>.

[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>.

[RFC7092] Kaplan, H. and V. Pascual, "A Taxonomy of Session
Initiation Protocol (SIP) Back-to-Back User Agents",
RFC 7092, DOI 10.17487/RFC7092, December 2013,
<https://www.rfc-editor.org/info/rfc7092>.

[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>.

[RFC8555] Barnes, R., Hoffman-Andrews, J., McCarney, D., and J.
Kasten, "Automatic Certificate Management Environment
(ACME)", RFC 8555, DOI 10.17487/RFC8555, March 2019,
<https://www.rfc-editor.org/info/rfc8555>.

[RFC9114] Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114,
June 2022, <https://www.rfc-editor.org/info/rfc9114>.

[RFC9409] Inamdar, K., Narayanan, S., Engi, D., and G. Salgueiro,
"The 'sip-trunking-capability' Link Relation Type",
RFC 9409, DOI 10.17487/RFC9409, July 2023,
<https://www.rfc-editor.org/info/rfc9409>.

[SIP-Connect-TR]
"SIP Connect Technical Recommendation",
<https://www.sipforum.org/download/sipconnect-technical-
recommendation-version-2-0/?wpdmdl=2818>.

14. 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>.

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[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>.

[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.B., "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>.

[RFC7317] Bierman, A. and M. Bjorklund, "A YANG Data Model for
System Management", RFC 7317, DOI 10.17487/RFC7317, August
2014, <https://www.rfc-editor.org/info/rfc7317>.

[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>.

[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.

[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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[RFC9110] Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
Ed., "HTTP Semantics", STD 97, RFC 9110,
DOI 10.17487/RFC9110, June 2022,
<https://www.rfc-editor.org/info/rfc9110>.

Authors' Addresses

Kaustubh Inamdar
Unaffiliated
Email: kaustubh.ietf@gmail.com

Sreekanth Narayanan
Unaffiliated
Email: sknth.n@protonmail.com

Cullen Jennings
Cisco Systems
Email: fluffy@iii.ca

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Automatic Peering for SIP Trunks draft-ietf-asap-sip-auto-peer-30

Automatic Peering for SIP Trunks
draft-ietf-asap-sip-auto-peer-30