Automatic SIP Trunking and Peering
RFC 10006
| Document | Type | RFC - Proposed Standard (August 2026) | |
|---|---|---|---|
| Authors | K. Inamdar , S. Narayanan , C. Jennings | ||
| Last updated | 2026-08-11 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| IESG | Responsible AD | Andy Newton | |
| Send notices to | (None) |
RFC 10006
Internet Engineering Task Force (IETF) K. Inamdar
Request for Comments: 10006 S. Narayanan
Category: Standards Track Unaffiliated
ISSN: 2070-1721 C. Jennings
Cisco Systems
August 2026
Automatic SIP Trunking and Peering
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 is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc10006.
Copyright Notice
Copyright (c) 2026 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 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
2. Requirements Language
3. Overview of Operations
3.1. Reference Architecture
3.2. Terminology
3.3. Configuration Workflow
3.4. Transport
4. HTTP Transport
4.1. HTTP Methods
4.2. Integrity and Confidentiality
4.3. Authenticated Client Identity
4.4. Encoding the Request
4.5. Identifying the Request Target
4.6. Generating Status Codes
5. Monitoring for Updates
6. Encoding the Service Provider Capability Set
7. Data Model for Capability Set
7.1. Tree Diagram
7.2. YANG Data Model
7.3. Extending the Capability Set
8. Processing the Capability Set Response
9. Examples
9.1. JSON Capability Set Document
9.2. Example Exchange
10. IANA Considerations
10.1. IANA-Maintained Module for SIP Option Tags
11. Security Considerations
11.1. OAuth Credentials
11.2. Client-Server Communication
11.3. YANG Security Considerations
12. References
12.1. Normative References
12.2. Informative References
Appendix A. Alternative Mechanisms to Transmit the Capability Set
Acknowledgments
Authors' Addresses
1. Introduction
The deployment of an infrastructure based on SIP [RFC3261] 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 (note that "peering" and
"trunking" can be used interchangeably). 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 SIPconnect technical recommendations [SIPconnect-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 the guidelines of the technical specifications are not hard
requirements that can be enforced by the peer. 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 SIPconnect technical
recommendations, in that both endeavors aim to promote and 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 document introduces the framework for Automatic Peering and
Trunking over SIP by 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.
2. Requirements Language
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.
3. Overview of Operations
This section provides a reference architecture against which the SIP
Automatic Peering framework may be implemented. Additionally, terms
that are commonly used in the context of the document are defined.
Last, considerations for the configuration workflow and the choice of
network transport between enterprise and service provider telephony
networks are discussed.
3.1. 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 Private Branch Exchange (SIP-PBX), media
endpoints (ME), and a Session Border Controller (SBC) [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 set up by the SP-SSE, and an HTTP
[RFC9110] server that stores the capability set document (indicated
as Cap Server in Figure 1).
+-----------------------------------------------------+
| +---------------+ +-----------------------+ |
| | | | | |
| | +----------+ | | +-------+ | |
| | | Cap | | HTTPS | | | | |
| | | Server |--|---------|-->| | | |
| | | |<-|---------|---| | +-----+ | |
| | +----------+ | | | |-->|SIP- | | |
| | | | | |<--|PBX | | |
| | | | | | +-----+ | |
| | +----------+ | | | SBC | | |
| | | | | SIP | | | | |
| | | SP-SSE |--|---------|-->| | +-----+ | |
| | | |<-|---------|---| |-->| ME | | |
| | +----------+ | | | |<--| | | |
| | | | | | +-----+ | |
| | +----------+ | (S)RTP | | | | |
| | | Media |--|---------|-->| | | |
| | | |<-|---------|---| | | |
| | +----------+ | | +-------+ | |
| +---------------+ +-----------------------+ |
| |
+-----------------------------------------------------+
Figure 1: Reference Architecture
3.2. Terminology
This document makes use of the following terminology:
Enterprise Network:
A communications network infrastructure deployed by an enterprise
that 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 document, the service provider
network is accessible over SIP for the establishment,
modification, and termination of calls and is accessible over HTTP
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 telephony networks refers to software that is
responsible for delivering core telephony functions. Call control
not only provides the basic functionality of setting up,
sustaining, and terminating calls, but it 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 (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 signaling, 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.
3.3. Configuration Workflow
A workflow that enables 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
an HTTP-based workflow that the enterprise network can use to 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 HTTP 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.
3.4. 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 HTTP request:
1. Edge elements are devices that normalize any mismatches between
the enterprise and service provider networks in the media and
signaling planes. As a result, when the capability set is
received from the SIP service provider network, the edge element
can generate appropriate configuration blocks (possibly across
multiple devices) to enable 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 to the guidelines of the YANG data model
described in this document. 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
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 signaling 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 does
not support a SIP extension, for instance, the SIP extension for
reliable provisional responses as defined in [RFC3262], 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 data
model introduced in this document.
4. HTTP Transport
This section describes the use of HTTP [RFC9110] as a transport
protocol for the peering workflow.
4.1. HTTP Methods
The workflow defined in this document leverages the HTTP GET method
and its corresponding response(s) to request 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 version 1.2 [RFC5246] or later [RFC8446]. When HTTP/3
[RFC9114] is used, TLS is incorporated within QUIC for the transport
of the capability set document. The usage of SIP or RTP-over-QUIC is
beyond the scope of this document. Additionally, the enterprise edge
element and capability server MUST 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 Automatic Peering 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 document does not mandate a
specific grant type. The implementation of OAuth 2.0 to obtain the
capability set is beyond the scope of this document. However, it
provides an example of how an enterprise SBC could leverage the
authorization code grant flow (Section 4.1 of [RFC6749]) 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 client (edge element) and the administrator, resource
owner, and enterprise network 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.
+---------------+
| 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.
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 an 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:
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 sip-trunking-capability [RFC9409] link relation type may be
leveraged assuming that the 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.
GET /.well-known/webfinger?
resource=https%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" : "https://ssp1.example.com",
"links" :
[
{
"rel" : "sipTrunkingCapability",
"href" :
"https://capserver.ssp1.com/capserver/capdoc.json"
}
]
}
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 Status Codes
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 (e.g., an
incorrect query parameter in the URI), the capability server MUST
generate a "400 Bad Request" status code for the incorrect request.
If requests contain an invalid token, the capability server MUST
generate a "403 Forbidden" status code 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
status codes (3xx).
The server SHOULD include the Location header field in such
responses. If the Location header is not included with the status
code, 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.
The enterprise edge element SHOULD handle the 3xx status codes from
the capability server in accordance with [RFC9110].
5. Monitoring for Updates
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 predefined
intervals to obtain the full capability set document. It is
recommended that capability servers are polled every 24 hours.
Alternatively, the enterprise edge elements can leverage
preconditions specified in [RFC9110] to conditionally retrieve the
capability set document if any changes have occurred.
6. Encoding the Service Provider Capability Set
In the context of this document, 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 practices and common problem sets typically seen between
enterprise and service provider SIP networks.
7. Data Model for Capability Set
This section contains a tree diagram (Section 7.1), the YANG module
[RFC7950] for encoding the service provider capability set
(Section 7.2), and a discussion about extending the capability set
(Section 7.3).
7.1. Tree Diagram
The meanings of the symbols in YANG tree diagrams are defined in
"YANG Tree Diagrams" [RFC8340].
The data model for the peering capability document has the following
structure:
module: ietf-sip-auto-peering
+--ro sip-auto-peering
+--ro variant identityref
+--ro revision
| +--ro not-before yang:date-and-time
| +--ro location inet:uri
+--ro transport-info
| +--ro transport* identityref
| +--ro registrar* [host port]
| | +--ro host union
| | +--ro port inet:port-number
| +--ro realm* [name]
| | +--ro name string
| | +--ro username? string
| | +--ro password? ianach:crypt-hash
| +--ro call-control* [host port]
| | +--ro host union
| | +--ro port inet:port-number
| +--ro dns-server* inet:ip-address
| +--ro outbound-proxy* [host port]
| +--ro host union
| +--ro port inet:port-number
+--ro call-spec
| +--ro early-media? boolean
| +--ro signaling-forking? boolean
| +--ro supported-method* enumeration
| +--ro caller-id
| | +--ro e164-format? boolean
| | +--ro preferred-method? enumeration
| +--ro number-range* [index]
| +--ro index uint16
| +--ro type? enumeration
| +--ro count? uint16
| +--ro value* string
+--ro media
| +--ro media-type-audio* [media-format]
| | +--ro media-format identityref
| | +--ro rate? uint16
| | +--ro ptime? uint8
| | +--ro parameter? string
| +--ro fax
| | +--ro protocol* enumeration
| +--ro rtp
| | +--ro rtp-trigger? boolean
| | +--ro symmetric-rtp? boolean
| +--ro rtcp
| +--ro symmetric-rtcp? boolean
| +--ro rtcp-feedback? boolean
+--ro dtmf
| +--ro payload-number? uint8
| +--ro iteration? boolean
+--ro security
| +--ro signaling
| | +--ro secure? boolean
| | +--ro version* identityref
| +--ro media-security
| | +--ro key-management* enumeration
| +--ro certificate-location? inet:uri
| +--ro secure-telephony-identity
| +--ro stir-compliance? boolean
| +--ro certificate-delegation? boolean
| +--ro acme-directory? inet:uri
+--ro extension* iana-sip-option-tags:sip-option-tag
7.2. YANG Data 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 [RFC9911] and system management [RFC7317]. In
addition, this YANG module references [RFC2833], [RFC4585],
[RFC4568], [RFC4733], [RFC4855], [RFC4961], [RFC5764], [RFC6716],
[RFC7362], [RFC8555], [RFC9645], [iana-crypt-hash], and
[iana-sip-option-tags].
<CODE BEGINS> file "ietf-sip-auto-peering@2026-06-15.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 9911: Common YANG Data Types.";
}
import ietf-yang-types {
prefix yang;
reference
"RFC 9911: Common YANG Data Types.";
}
import iana-crypt-hash {
prefix ianach;
reference
"https://www.iana.org/assignments/iana-crypt-hash/";
}
import ietf-tls-common {
prefix tlscmn;
reference
"RFC 9645: YANG Groupings for TLS Clients and TLS Servers.";
}
import iana-sip-option-tags {
prefix iana-sip-option-tags;
reference
"https://www.iana.org/assignments/iana-sip-option-tags/";
}
organization
"IETF ASAP (Automatic SIP trunking And Peering) Working Group";
contact
"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 an out-of-band interface.
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.
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.
Copyright (c) 2026 IETF Trust and the persons identified as
authors of the code. All rights reserved.
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).
All revisions of IETF and IANA published modules can be found
at the YANG Parameters registry group
(https://www.iana.org/assignments/yang-parameters).
This version of this YANG module is part of RFC 10006; see the
RFC itself for full legal notices.";
revision 2026-06-15 {
description
"Initial version";
reference
"RFC 10006: Automatic SIP Trunking and Peering";
}
identity capability-doc-variant {
description
"Base for capability document variants.";
}
identity v1-0 {
base capability-doc-variant;
description
"Variant 1.0 of the capability set document.";
}
identity sip-transport-protocol {
description
"Base for transport protocols used to send SIP requests
across.";
}
identity udp {
base sip-transport-protocol;
description
"UDP used for SIP requests and responses.";
}
identity tcp {
base sip-transport-protocol;
description
"TCP used for SIP requests and responses.";
}
identity codec-variant {
description
"Base for variants of codec supported by the service
provider.";
}
identity pcmu {
base codec-variant;
description
"PCMU (G.711 μ-law) audio codec.";
}
identity pcma {
base codec-variant;
description
"PCMA (G.711 A-law) audio codec.";
}
identity opus {
base codec-variant;
description
"Opus audio codec.";
reference
"RFC 6716: Definition of the Opus Audio Codec.";
}
identity g722 {
base codec-variant;
description
"G.722 audio codec.";
}
identity g729 {
base codec-variant;
description
"G.729 codec.";
}
grouping entity {
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 variant {
type identityref {
base capability-doc-variant;
}
mandatory true;
description
"A node that identifies the version number of the capability
set document. RFC 10006 defines the parameters for
variant 1.0; 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.";
reference
"RFC 10006: Automatic SIP Trunking and Peering";
}
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 yang:date-and-time;
mandatory true;
description
"A node that identifies the absolute UTC 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 would not 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 identityref {
base sip-transport-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, 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 FQDN, the DNS
element must be populated with one or more valid DNS
server IP addresses.";
}
list realm {
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;
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 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 Authorization
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, a subdomain of the
SIP service provider network, or a FQDN of the SIP service
provider network. If the transport address of a call
control server(s) is specified using either a subdomain or
a FQDN, 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.";
}
leaf-list dns-server {
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
FQDN 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 FQDN
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; therefore, the registrar
and call-control lists can be omitted.";
}
}
container call-spec {
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 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
signaling 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 beforehand 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-method {
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.";
}
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
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 value of 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. 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
dereference this URL in order to obtain the DID number
ranges allocated by the SIP service provider.";
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;
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 number-range
'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 RFC 10006 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
RFC 10006, video media streams are not exchanged
between enterprise and service provider SIP networks.";
reference
"RFC 10006: Automatic SIP Trunking and Peering";
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
are described the 'param' field.
Given that the parameters of media formats can vary from
one communication session to another (e.g., across two
separate communication sessions), the packetization
time (ptime) used for the PCMU media format might vary
from 10 to 30 ms, and 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.";
reference
"RFC 4855: Media Type Registration of RTP Payload Formats";
leaf media-format {
type identityref {
base codec-variant;
}
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.";
}
}
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 T.38 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
"T.38 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
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 practice of
preserving the enterprise network in a hairpinned call
flow is fairly common, it is recommended that SIP
service providers avoid this practice. 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.
Enforcement of this requirement by service providers
on enterprise networks is typically useful in scenarios
such as media latching. 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.";
reference
"RFC 4961: Symmetric RTP / RTP Control Protocol (RTCP),
RFC 7362: Latching: Hosted NAT Traversal (HNT) for Media
in Real-Time Communication";
}
}
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. 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.";
reference
"RFC 4961: Symmetric RTP / RTP Control Protocol (RTCP)";
}
leaf rtcp-feedback {
type boolean;
description
"A node that indicates whether the SIP service
provider supports the RTP profile extension for
RTCP-based feedback. 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 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 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.";
reference
"RFC 4585: Extended RTP Profile for Real-time Transport
Control Protocol (RTCP)-Based Feedback (RTP/AVPF)";
}
}
}
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.";
}
leaf iteration {
type boolean;
description
"A value of true indicates that the service provider
supports the newer standard while a value of false
indicates that the service provider prefers the
older standard";
reference
"RFC 4733: RTP Payload for DTMF Digits, Telephony
Tones, and Telephony Signals,
RFC 2833: RTP Payload for DTMF Digits, Telephony
Tones and Telephony Signals";
}
}
container security {
description
"A container that encapsulates characteristics about
encrypting signaling 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 signaling
messages between the enterprise and service provider
network. This node is a boolean type. A value of
true indicates that the service provider supports
SIP sessions over TLS, whereas a value of false
indicates that the service provider does not support
SIP over TLS.";
}
leaf-list version {
when "../secure = 'true'";
type identityref {
base tlscmn:tls-version-base;
}
description
"A list that specifies the version(s) of TLS supported.";
}
}
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 (SDES)
key management.";
}
enum dtls-srtp {
description
"Secure Real-time Transport Protocol (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'.";
reference
"RFC 4568: Session Description Protocol (SDP) Security
Descriptions for Media Streams,
RFC 5764: Datagram Transport Layer Security (DTLS)
Extension to Establish Keys for the Secure Real-time
Transport Protocol (SRTP)";
}
}
leaf certificate-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 certificate-location node is populated
with a URL, which when dereferenced, provides a single
Privacy-Enhanced Mail (PEM) encoded file that contains all
certificates in the chain of trust.";
}
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 a 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 certificate-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 a 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, i.e., 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 "../certificate-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). The directory object
URL, when dereferenced, provides a collection of field
name-value pairs. Certain field name-value pairs
provided in the response are used to bootstrap the
process of obtaining delegate certificates.
This node MUST only be included in the capability
set if the value of the certificate-delegation leaf node
is set to true.";
reference
"RFC 8555: Automatic Certificate Management Environment
(ACME)";
}
}
}
leaf-list extension {
type iana-sip-option-tags:sip-option-tag;
description
"A list of SIP option tags (extensions) supported by the
service provider network.";
reference
"https://www.iana.org/assignments/iana-sip-option-tags/";
}
}
}
<CODE ENDS>
7.3. Extending the Capability Set
There are situations in which equipment manufacturers or service
providers would benefit from extending the YANG module defined in
this document. 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, and service provider support
numbers, among others. Extensions of the module can be achieved by
importing the module defined in this document. An example is
provided below.
Consider a new YANG module "example-vendor-config" specified for
Vendor's enterprise SBC. The "example-vendor-config" YANG module is
configured as follows:
module example-vendor-config {
yang-version 1.1;
namespace "urn:ietf:params:xml:ns:yang:example-vendor-config";
prefix vendor;
import ietf-sip-auto-peering {
prefix sipap;
reference
"RFC 10006: Automatic SIP Trunking and Peering";
}
organization
"Vendor Enterprise.";
contact
"Vendor Enterprise
1234 Vendor Street
Anytown, State 12345
Tel: +1 424 254 5300
<mailto:vendor@vendor.com>";
description
"Example of a vendor configuration data model that augments the
IETF SIP auto-peering model to include vendor-specific SBC
configuration parameters.";
revision 2026-12-06 {
description
"Initial revision of Vendor Enterprise SBC
configuration data model.";
reference
"RFC 10006: Automatic SIP Trunking and Peering";
}
augment "/sipap:sip-auto-peering" {
description
"Augmentation of the SIP auto-peering model to include vendor-
specific SBC configuration parameters.";
container vendorConfig {
leaf vendorConfigParam1 {
type int32;
description
"Vendor configuration parameter 1
(SBC Device ID).";
}
leaf vendorConfigParam2 {
type string;
description
"Vendor configuration parameter 2
(SBC Device name).";
}
description
"Container for vendor SBC configuration.";
}
}
}
In the example above, a custom module named "example-vendor-config"
uses the "augment" statement as defined in Section 4.2.8 of [RFC7950]
to extend the module defined in this document.
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. Examples of
this include 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 appropriately, and 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 normalizing
mismatches in the signaling 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
NOTE: '\' line wrapping per [RFC8792].
<CODE BEGINS> file "asap-example.json"
{
"ietf-sip-auto-peering:sip-auto-peering":
{
"variant": "ietf-sip-auto-peering:v1-0",
"revision": {
"not-before": "2026-06-15T10:30:00Z",
"location":
"https://capserver.example.org/capserver/capdoc.json"
},
"transport-info": {
"transport": [
"ietf-sip-auto-peering:tcp",
"ietf-sip-auto-peering:udp"
],
"registrar": [
{
"host": "registrar1.voip.example.com",
"port": 5060
},
{
"host": "registrar2.voip.example.com",
"port": 5060
}
],
"realm": [
{
"name": "voip.example.com",
"username": "voip",
"password":
"$6$OoEJwExxp6U/FRFq$4RkL2lSSGLoKdfGjX4lQLF\
Xo89gc0wtJsKiBxg/BBz6aNwu7C.D3kRUwD7lvJm6rhaCdhSzVh/XfkkAUY2dTu0"
}
],
"call-control": [
{
"host": "callServer1.voip.example.com",
"port": 5060
},
{
"host": "192.0.2.40",
"port": 5065
}
],
"dns-server": [
"192.0.2.50",
"192.0.2.51"
],
"outbound-proxy": [{
"host": "192.0.2.35",
"port": 5060
}]
},
"call-spec": {
"early-media": true,
"signaling-forking": false,
"supported-method": [
"invite",
"options",
"bye",
"cancel",
"ack",
"prack",
"subscribe",
"notify",
"register"
],
"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": "ietf-sip-auto-peering:pcmu",
"rate": 8000,
"ptime": 20
},
{
"media-format": "ietf-sip-auto-peering:g729",
"rate": 8000,
"ptime": 20,
"parameter": "annexb"
}
],
"fax": {
"protocol": [
"t38",
"pass-through"
]
},
"rtp": {
"rtp-trigger": true,
"symmetric-rtp": true
},
"rtcp": {
"symmetric-rtcp": true,
"rtcp-feedback": true
}
},
"dtmf": {
"payload-number": 101,
"iteration": false
},
"security": {
"signaling": {
"secure": true,
"version": ["ietf-tls-common:tls12", "ietf-tls-comm\
on:tls13"]
},
"media-security": {
"key-management": ["sdes", "dtls-srtp"]
},
"certificate-location":
"https://sipserviceprovider.com/certificateList.pem",
"secure-telephony-identity": {
"stir-compliance": true,
"certificate-delegation": true,
"acme-directory":
"https://sipserviceprovider.com/acme.html"
}
},
"extension": [
"one-hundred-rel",
"timer",
"replaces",
"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 preconfigured with the request
target for the capability set document or has dynamically found the
request target, the edge element generates an HTTP 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
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
{
"ietf-sip-auto-peering:sip-auto-peering": ...
}
10. IANA Considerations
This document registers two new URIs in the "IETF XML Registry"
[RFC3688]. Following the format in [RFC3688], the following
registrations have been made.
URI: urn:ietf:params:xml:ns:yang:ietf-sip-auto-peering
Registrant Contact: The IESG.
XML: N/A; the requested URI is an XML namespace.
URI: urn:ietf:params:xml:ns:yang:iana-sip-option-tags
Registrant Contact: The IESG.
XML: N/A; the requested URI is an XML namespace.
This document registers two new YANG modules in the "YANG Module
Names" registry [RFC6020].
Name: ietf-sip-auto-peering
Maintained by IANA? N
Namespace: urn:ietf:params:xml:ns:yang:ietf-sip-auto-peering
Prefix: sipap
Reference: RFC 10006
Name: iana-sip-option-tags
Maintained by IANA? Y
Namespace: urn:ietf:params:xml:ns:yang:iana-sip-option-tags
Prefix: sip-option-tags
Reference: RFC 10006
10.1. IANA-Maintained Module for SIP Option Tags
This document defines the initial version of the IANA-maintained
"iana-sip-option-tags" YANG module. The most recent version of the
YANG module is available in the "YANG Parameters" registry group
[YANG-PARAMS].
IANA has added the following to the "Notes" field of the "iana-sip-
option-tags" entry in the "YANG Module Names" registry within the
"YANG Parameters" registry group:
| New values must not be directly added to the "iana-sip-option-
| tags" YANG module. They must instead be added to the "Option
| Tags" registry [SIP-PARAMS].
When a value is added to the "Option Tags" registry, a new "enum"
statement must be added to the "iana-sip-option-tags" YANG module.
The "enum" statement, and substatements thereof, should be defined:
"enum": Replicates a name from the registry.
"description": Replicates the description from the registry.
"reference": Replicates the reference(s) from the registry with the
title of the document(s) added.
Unassigned or reserved values are not present in the module.
When the "iana-sip-option-tags" YANG module is updated, a new
"revision" statement with a unique revision date needs to be added in
front of the existing "revision" statements. The "revision"
statement MUST contain both "description" and "reference"
substatements as follows.
The "description" substatement captures what changed in the revised
version. Typically, the description enumerates the changes such as
updates to existing entries (e.g., update a description or a
reference) or notes about which "enums" were added or had their
status changed (e.g., deprecated, discouraged, or obsoleted).
When such a description is not feasible, the description varies on
how the update is triggered.
* If the update is triggered by an RFC, the "description"
substatement should include or consist of this text:
Applied updates as specified by RFC 10006.
* If the update is triggered following another IANA registration
policy but not all the values in the registry are covered by the
same policy, insert this text (where "Some_IANA_policy" refers to
one of the defined registration policies in Section 4 of
[RFC8126]):
Applied updates as specified by the registration policy
Some_IANA_policy.
The "reference" substatement points specifically to the published
module at [YANG-PARAMS]. It may also point to an authoritative event
triggering the update to the YANG module. In all cases, this event
is cited from the underlying IANA registry. If the update is
triggered by an RFC, that RFC must also be included in the
"reference" substatement.
IANA has added this note to the "Option Tags" registry within the
"Session Initiation Protocol (SIP) Parameters" registry group
[SIP-PARAMS]:
| When this registry is modified, the YANG module "iana-sip-option-
| tags" <https://www.iana.org/assignments/iana-sip-option-tags> must
| be updated as defined in RFC 10006.
The service provider will filter out the advertised extensions using
local policy.
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.
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 Authorization header
field provided in response to authentication challenges.
11.3. YANG Security Considerations
The "ietf-sip-auto-peering" YANG module defines a data model that a
service provider MUST adhere to while creating the capability set
document, preferably in an automated fashion. The capability set
document SHOULD be formatted as a JSON file as exhibited in
Section 9. The service provider communicates the URL of this JSON
file in an out-of-band manner to the enterprise. Alternatively, the
enterprise uses WebFinger to discover the URL of the JSON file. The
enterprise SBC downloads the JSON file and parses it. Once it has
validated that the JSON file is correctly formatted, it applies the
configuration and peers with the service provider's network for SIP
calls to occur.
It is possible that enterprises may purchase numbers in different
countries or regions. In this scenario, there would be multiple SIP
trunks between the enterprise and the service provider. The service
provider is responsible for creating the capability set documents for
each SIP trunk. The capability set document cannot be modified by
the enterprise. It can only be created one time by the service
provider for each enterprise entering into an agreement with the
service provider. Therefore, there are no particularly sensitive
writable data nodes.
There are no particularly sensitive writable data nodes.
Some of the readable data nodes in this YANG module may be considered
sensitive or vulnerable in some network environments. It is thus
important to control read access (e.g., via get, get-config, or
notification) to these data nodes. Specifically, the following
subtrees and data nodes have particular sensitivities/
vulnerabilities:
* registrar: This list contains IP addresses or hostnames belonging
to registration servers in the service provider network, which may
be targeted by malicious actors.
* realms: This list contains sensitive credentials that are utilized
by the enterprise to create a registration with the service
provider's network. The registration is a prerequisite to making
and receiving calls to and from the service provider,
respectively.
* call-control: This list contains IP addresses or hostnames
belonging to call processing servers in the service provider
network, which may be targeted by malicious actors.
* outbound-proxy: This list contains IP addresses or hostnames
belonging to SIP proxies in the service provider network, which
may be targeted by malicious actors.
* number-range: This list contains a range of phone numbers
allocated by the service provider to an enterprise that the
service provider may want to conceal from other enterprises or
customers.
There are no particularly sensitive RPC or action operations.
This YANG module uses groupings from other YANG modules that define
nodes that may be considered sensitive or vulnerable in network
environments. Refer to the Security Considerations of [RFC9911] and
[RFC7317] for information as to which nodes may be considered
sensitive or vulnerable in network environments.
The YANG module "iana-sip-option-tags" defines a set of types. These
nodes are intended to be reused by other YANG modules. This module
by itself does not expose any data nodes that are writable, data
nodes that contain read-only state, or RPCs. As such, there are no
additional security issues related to this YANG module that need to
be considered.
12. References
12.1. Normative References
[iana-crypt-hash]
IANA, "iana-crypt-hash YANG Module",
<https://www.iana.org/assignments/iana-crypt-hash>.
[iana-sip-option-tags]
IANA, "iana-sip-option-tags YANG Module",
<https://www.iana.org/assignments/iana-sip-option-tags>.
[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>.
[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, July 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[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>.
[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>.
[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>.
[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>.
[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>.
[RFC8792] Watsen, K., Auerswald, E., Farrel, A., and Q. Wu,
"Handling Long Lines in Content of Internet-Drafts and
RFCs", RFC 8792, DOI 10.17487/RFC8792, June 2020,
<https://www.rfc-editor.org/info/rfc8792>.
[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>.
[RFC9645] Watsen, K., "YANG Groupings for TLS Clients and TLS
Servers", RFC 9645, DOI 10.17487/RFC9645, October 2024,
<https://www.rfc-editor.org/info/rfc9645>.
[RFC9911] Schönwälder, J., Ed., "Common YANG Data Types", RFC 9911,
DOI 10.17487/RFC9911, December 2025,
<https://www.rfc-editor.org/info/rfc9911>.
12.2. Informative References
[RFC2833] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF
Digits, Telephony Tones and Telephony Signals", RFC 2833,
DOI 10.17487/RFC2833, May 2000,
<https://www.rfc-editor.org/info/rfc2833>.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation Protocol
(SIP)", RFC 3262, DOI 10.17487/RFC3262, July 2002,
<https://www.rfc-editor.org/info/rfc3262>.
[RFC3688] Mealling, M., "The IETF XML Registry", BCP 81, RFC 3688,
DOI 10.17487/RFC3688, January 2004,
<https://www.rfc-editor.org/info/rfc3688>.
[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>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
<https://www.rfc-editor.org/info/rfc4961>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6241] Enns, R., Ed., Bjorklund, M., Ed., Schoenwaelder, J., Ed.,
and A. Bierman, Ed., "Network Configuration Protocol
(NETCONF)", RFC 6241, DOI 10.17487/RFC6241, June 2011,
<https://www.rfc-editor.org/info/rfc6241>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <https://www.rfc-editor.org/info/rfc6716>.
[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>.
[RFC8126] Cotton, M., Leiba, B., and T. Narten, "Guidelines for
Writing an IANA Considerations Section in RFCs", BCP 26,
RFC 8126, DOI 10.17487/RFC8126, June 2017,
<https://www.rfc-editor.org/info/rfc8126>.
[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>.
[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-PARAMS]
IANA, "Session Initiation Protocol (SIP) Parameters",
<https://www.iana.org/assignments/sip-parameters>.
[SIPconnect-TR]
SIP Forum, "SIPconnect 2.0 Technical Recommendation",
<https://www.sipforum.org/download/sipconnect-technical-
recommendation-version-2-0/?wpdmdl=2818>.
[YANG-PARAMS]
IANA, "YANG Parameters",
<https://www.iana.org/assignments/yang-parameters>.
Appendix A. Alternative Mechanisms to Transmit the Capability Set
There are alternative mechanisms that the SIP service provider can
use to 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. An HTTP-
based approach would be relatively easier to adopt, as most edge
devices deployed in enterprise networks today already support HTTP;
from the perspective of service provider networks, all that is
required is for them to deploy HTTP 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
data models [RFC6020] [RFC7950] that enable configuration to be
pushed over NETCONF [RFC6241] to enterprise networks from a
centralized source hosted in service provider networks. The presence
of proprietary software logic for call and media handling in
enterprise devices would preclude the generation of a "one-size-fits-
all" YANG data 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.
Acknowledgments
We would like to thank those who provided detailed and thoughtful
comments on this document, 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, Ebben Aries, Jen
Linkova, David Dong, Gorry Fairhurst, Mohamed Boucadair, Paul
Wouters, Mike Bishop, Andy Newton, and Amanda Baber for their reviews
and feedback.
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