Network Working Group                                       J. Rosenberg
Internet-Draft                                                     Five9
Intended status: Standards Track                             C. Jennings
Expires: August 10, 2020                                   Cisco Systems
                                                            A. Minessale
                                                            J. Livingood
                                                               J. Uberti
                                                        February 7, 2020

                RealTime Internet Peering for Telephony


   This document specifies the Realtime Internet Peering for Telephony
   (RIPT) protocol.  RIPT is used to provide peering of voice and video
   communications between entities.  These include a traditional voice
   trunking provider (such as a telco), and a trunking consumer (such as
   an enterprise PBX or contact center), or between a video conferencing
   endpoint deployed in an enterprise, and a video conferencing SaaS
   service.  RIPT is an alternative to SIP, SDP and RTP for these use
   cases, and is designed as a web application using HTTP/3.  Using
   HTTP/3 allows implementors to build their applications on top of
   cloud platforms, such as AWS, Azure and Google Cloud, all of which
   are heavily focused on HTTP based services.  RIPT also addresses many
   of the challenges of traditional SIP-based peering.  It supports
   modern techniques for load balancing, autoscaling, call-preserving
   failover, graceful call migrations, security by default, STIR-based
   caller ID, provisioning, and capabilities - all of which have been
   challenges with traditional SIP peering and voice trunking.  Since it
   runs over HTTP/3, it works through NATs and firewalls with the same
   ease as HTTP does.

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

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   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 August 10, 2020.

Copyright Notice

   Copyright (c) 2020 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
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Background  . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Structure of this Document  . . . . . . . . . . . . . . . . .   5
   3.  Solution Requirements . . . . . . . . . . . . . . . . . . . .   5
   4.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   6
   5.  Reference Architecture  . . . . . . . . . . . . . . . . . . .   7
   6.  Web Resource Model  . . . . . . . . . . . . . . . . . . . . .   9
   7.  Deployment Examples . . . . . . . . . . . . . . . . . . . . .  10
     7.1.  Enterprise Voice Trunking . . . . . . . . . . . . . . . .  11
     7.2.  BYO Voice for CCaaS . . . . . . . . . . . . . . . . . . .  12
     7.3.  Inter-Carrier Voice Peering . . . . . . . . . . . . . . .  12
     7.4.  Video Endpoint to Meetings Provider . . . . . . . . . . .  13
   8.  Overview of Operation . . . . . . . . . . . . . . . . . . . .  13
     8.1.  Bootstrap . . . . . . . . . . . . . . . . . . . . . . . .  13
     8.2.  Login . . . . . . . . . . . . . . . . . . . . . . . . . .  14
     8.3.  TG Discovery  . . . . . . . . . . . . . . . . . . . . . .  14
     8.4.  Customer TG Registration  . . . . . . . . . . . . . . . .  15
     8.5.  Handler Registration  . . . . . . . . . . . . . . . . . .  16
     8.6.  Obtaining Certificates  . . . . . . . . . . . . . . . . .  19
     8.7.  Call Establishment  . . . . . . . . . . . . . . . . . . .  19
     8.8.  Media Exchange  . . . . . . . . . . . . . . . . . . . . .  22
     8.9.  Graceful Migration  . . . . . . . . . . . . . . . . . . .  23
     8.10. Non-Graceful Migration  . . . . . . . . . . . . . . . . .  24
   9.  Normative Protocol Specification  . . . . . . . . . . . . . .  26

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     9.1.  Bootstrapping . . . . . . . . . . . . . . . . . . . . . .  26
     9.2.  TG Discovery  . . . . . . . . . . . . . . . . . . . . . .  26
     9.3.  TG Construction . . . . . . . . . . . . . . . . . . . . .  26
     9.4.  Consumer TG Registration  . . . . . . . . . . . . . . . .  27
     9.5.  Handler Registration and Lifecycle Management.  . . . . .  28
     9.6.  Handler Description Format  . . . . . . . . . . . . . . .  28
       9.6.1.  ABNF  . . . . . . . . . . . . . . . . . . . . . . . .  30
     9.7.  Certificate Enrollment  . . . . . . . . . . . . . . . . .  30
     9.8.  Call Establishment  . . . . . . . . . . . . . . . . . . .  31
     9.9.  Signaling and Media Byway Establishment . . . . . . . . .  32
     9.10. Basic Call State Management . . . . . . . . . . . . . . .  33
     9.11. Sending and Receiving Media . . . . . . . . . . . . . . .  34
       9.11.1.  The Media Chunk Format . . . . . . . . . . . . . . .  35
       9.11.2.  The Control Chunk  . . . . . . . . . . . . . . . . .  36
       9.11.3.  Client Media handling  . . . . . . . . . . . . . . .  37
       9.11.4.  Server Media Handling  . . . . . . . . . . . . . . .  37
     9.12. Connection and Byway Lifecycle Management . . . . . . . .  38
     9.13. Graceful Call Migration . . . . . . . . . . . . . . . . .  38
     9.14. Ungraceful Call Migration . . . . . . . . . . . . . . . .  39
   10. RAML API  . . . . . . . . . . . . . . . . . . . . . . . . . .  39
   11. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  39
     11.1.  Registration of Well-Known Web Services  . . . . . . . .  39
     11.2.  RIPT . . . . . . . . . . . . . . . . . . . . . . . . . .  39
     11.3.  RIPT-Oauth . . . . . . . . . . . . . . . . . . . . . . .  40
   12. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  40
   13. Security Considerations . . . . . . . . . . . . . . . . . . .  40
   14. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  40
   15. References  . . . . . . . . . . . . . . . . . . . . . . . . .  40
     15.1.  Normative References . . . . . . . . . . . . . . . . . .  40
     15.2.  Informative References . . . . . . . . . . . . . . . . .  41
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  41

1.  Introduction

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in [RFC2119].

1.1.  Background

   Cloud computing platforms, such as those provided by Amazon, Azure,
   and Google, have now become mainstream for the development of
   software applications.  These platforms are targeted at enabling web
   applications, and as such many of their features are based on the
   usage of HTTP.

   One example are HTTP load balancers.  Cloud computing platforms
   provide highly scalable, geographically distributed, redundant load

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   balancers.  These load balancers can monitor the state of downstream
   servers and can uniformly distribute load amongst them.  The load
   balancers can compensate for failure of individual nodes and send new
   traffic to other nodes.

   Autoscaling is another example.  The cloud computing platforms can
   automatically add new instances of a server backend, or remove them,
   and automatically configure the load balancers to include them in the
   pool of available servers.

   Yet another example is Kubernetes, which allows web-based
   applications to be deployed into containers (typically Docker), with
   load balancing, scaling, and HTTP request routing.

   Another example are HTTP tracing tools, which facilitate the tracing
   of requests through distributed microservices.  These tools can
   autogenerate sequence diagrams and facilitate in troubleshooting.

   Yet another example are API gateways (such as APIGee and Kong), which
   provide authentication and authorization, provisioning of
   applications, rate limiting, analytics, sandboxing for testing,
   embedded documentation, and so on.

   And yet another example are denial-of-service prevention techniques,
   typically done using BGP peering and re-routing.  Though in principle
   these techniques can work for VoIP, they are deployed in conjunction
   with the load balancers which represent the entry point into these
   cloud provider networks.  Consequently, the protections these cloud
   providers offer do not extend to applications which merely use these
   platforms for virtual machines.

   A more recent technology are service meshes, such as Istio, which
   utilize sidecar HTTP proxies to facilitate inter-service
   communications.  These systems come with robust control planes which
   enable additional routing features, such as canary deploys,
   percentage based routing, and so on.

   None of these capabilities can be used by real-time applications
   based on the Session Initiation Protocol (SIP) [RFC3261], since SIP
   is not an HTTP-based protocol.  The newest HTTP specification -
   HTTP/3 [I-D.ietf-quic-http] - utilizes QUIC
   [I-D.ietf-quic-transport], which runs on top of UDP.  This means that
   it is now possible to utilize HTTP for real-time media and call
   control together.

   This document proposes a new protocol for performing basic call
   control and media processing as an HTTP application.

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2.  Structure of this Document

   The document is broadly split into two parts - explanatory and
   normative materials.  The explanatory text is non-normative, and
   contains no [RFC2119] language.  Much of this text is meant to help
   readers familiar with SIP, understand how SIP concepts translate (or
   don't) into RIPT.  These sections include Requirements Section 3,
   Terminology Section 4, Reference Architecture Section 5, and
   Deployment Examples Section 7.  The remainder of the document
   specifies normative procedures.

3.  Solution Requirements

   The protocol defined here is based on the following requirements:

   REQ1: The solution shall not require extensions or modifications to

   REQ2: The solution shall work with both L4 and L7 HTTP load balancers

   REQ3: The solution shall work in ways that are compatible with best
   practices for load balancers and proxies supporting HTTP/3, and not
   require any special changes to these load balancers in order to

   REQ4: The solution should hide the number of servers behind the load
   balancer, allow the addition or removal of servers from the cluster
   at will, and not expose any of this information to the peer

   REQ5: The solution shall enable the usage of autoscaling technologies
   used in cloud platforms, without any special consideration for RIPT -
   its just a web app

   REQ6: The solution shall provide call preservation in the face of
   failures of the server or client.  It is acceptable for a brief blip
   of media due to transient packet loss, but thats it

   REQ7: The solution shall support built-in migration, allowing a
   server to quickly shed load in order to be restarted or upgraded,
   without any impact to calls in progress

   REQ8: The solution will be easy to interoperate with SIP

   REQ9: The solution shall be incrementally deployable - specifically
   it must be designed for easy implementation by SBCs and easy
   deployment by PSTN termination and origination providers who do not
   utilize cloud platforms

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   REQ10: The solution shall require authentication and encryption, with
   no opportunity to disable them.  Furthermore, it will require secure
   callerID, with no provision for insecure callerID

   REQ11: The solution shall provide low latency for media

   REQ12: The solution shall support audio and video

   REQ13: The solution must support secure caller ID out of the gate and
   not inherit any of the insecure techniques used with SIP

   REQ14: The solution shall include mandatory-to-implement provisioning
   operations for cases where there is a customer-provider relationship

   REQ15: The solution shall make it possible to perform rolling
   upgrades through a cluster many times a day, without call drops

4.  Terminology

   This specification follows the terminology of HTTP/3, but adds the
   following concepts:

   Client: An entity which implements the rules of the client defined in
   this specification.  A RIPP client always acts as an HTTP client.

   Server: An entity which implements the rules of a server as defined
   in this specification.  A RIPP server always acts as an HTTP server.

   Terminal Group (TG): A container for calls between a client and
   server.  A TG is identified by a URI, hosted on the server.  A TG
   acts as a unit of policy and capabilities, including allowed phone
   numbers.  The acronym is a nod to its circuit switched predecessor,
   the Trunk Group.  It exists to facilitate automated configuration of
   call routing and avoid call failures due to mismatched capabilities.

   Call: A real-time voice and/or video session.  A call is always
   associated with a TG, and is identified by a URI hosted on the

   Customer: An end user or administrative entity that utilizes
   communications services from a provider in order to make and receive
   calls.  The relationship between the customer and provider is static
   and does not vary from call to call, and does not vary in call
   direction either. (e.g., Verizon would be the provider to an
   enterprise customer, and the enterprise would be the customer of
   Verizon).  For the purposes of this specification, this matters in
   that a customer, if it wishes to implement the server role, will need
   to configure its provider with credentials and URI needed to enable

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   the provider, acting as a client, to connect to its server.
   Consequently, RIPT provides a facility for this.

   Provider: The administrative entity that provides communications
   services to the customer.

   Byway: A bidirectional byte stream between a client and server.  A
   byway passes its data through HTTP, using a set of techniques which
   depend on the capabilities of both sides and the use cases for which
   they are needed.  This specification considers two types - a
   signaling byway and a media byway.

   Handler: A handler is a "device" - an SBC, a phone, an IP PBX - and
   can be a software or hardware entity which sends and receives media
   associated with a call.  A handler has a description, which includes
   its advertisement, which defines its media capabilities.  The handler
   can change during a call (as in the case of a client failing and its
   calls being picked up by a backup).

   Advertisement: A document which describes a set of semi-static
   capabilities for sending and receiving audio and video with different
   codecs, along with constraints such as maximum resolution or frame
   rates.  An advertisement is semi-static in that it does not change
   from call to call, and is rather a property of the software or
   hardware system, which changes normally only upon upgrade or
   configuration change.

   Directive: The directive is an instruction on how media should be
   sent.  It is communicated from the server, which tells a handler
   where it should send media to for this call.

5.  Reference Architecture

   RIPT is also designed such that all communications between the RIPT
   client - which is just an HTTP client - and the RIPT server - which
   is just an HTTP server - can easily sit behind a typical HTTP load
   balancer, as shown below:

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                                                    |            |
                                                    |  RIPT      |
                                                   >|  Server    |
                                                  / |            |
                                                 /  |            |
                                                /   +------------+
                            +---------+    /
                            |         |   /
         +------------+     |         |  /          +------------+
         |            |     |         | /           |            |
         | RIPT       |     |  HTTP   |/            |  RIPT      |
         | Client     |---->|  LB     | ----------->|  Server    |
         |            |     |         |\            |            |
         |            |     |         | \           |            |
         +------------+     |         |  \          +------------+
                            +---------+   \
                                               \    +------------+
                                                \   |            |
                                                 \  |  RIPT      |
                                                  > |  Server    |
                                                    |            |
                                                    |            |

   The customer and provider role is asymmetric.  Typically the customer
   has purchased services from the provider.  From a protocol
   perspective, RIPT assumes that the customer has an account with the
   provider, and it is possible for the customer to obtain an OAuth
   token which can be used to authenticate.  However, there is no login
   technique which enables the provider to obtain a token to place calls
   towards the customer.

   Since calls always originate from client to server, in order to
   receive inbound calls, a customer can also run a RIPT server.  RIPT
   supports a simple registration mechanism by which the customer -
   acting as a client - can use RIPT to register its TG URI with the
   provider.  This registration also includes a bearer token for

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   It is anticipated that an entity implementing the server role will
   use a load balancer to receive incoming requests to place calls.
   This is not required, of course.  However, this specification
   provides no facility for a customer implementing only the client role
   to receive inbound calls.  It is worth restating that this load
   balancer is NOT specific to RIPT - it is any off-the-shelf HTTP load
   balancer which supports HTTP/3.  No specific support for RIPT is
   required.  RIPT is just an application ontop of HTTP.

   Because RIPT clients and servers are nothing more than HTTP/3
   applications, the behavior or RIPT is specified entirely by
   describing how various RIPT procedures map to the core HTTP/3
   primitives available to applications - opening connections, closing
   connections, sending requests and responses, receiving requests and
   responses, and setting header fields and bodies.  That's it.

6.  Web Resource Model

   The web resource model for RIPT is based on the interplay between
   three key resources held by the server.  These are the TG, the
   handler, and the call.

   The TG is a representation of the service offered by the server to
   the client.  It indicates the allowed directions for calls (in this
   specification, outbound only), the allowed identities that can be
   used for caller ID, the allowed numbers which can be called, and the
   set of advertised media capabilities for the service.  The purpose of
   the TG is to provide the client all of the information it needs to
   know, in advance, whether the placement of a call is within the
   allowed policy scope of the server.  It is also meant to automate
   configuration, providing information which is manually entered today.
   It also facilitates call routing, enabling a client to know where to
   route calls, and minimizes call failures by providing information up
   front about whether a call may be rejected.

   To ensure authenticated caller ID everywhere, the TG specifies the
   set of allowed caller IDs through an [RFC8226] certificate.  This not
   only informs the client about what numbers it can originate with, it
   also proves to the client that it is capable of vouching for those

   For example, a telco might offer an enterprise customer a service in
   which it can place calls to any number in the world, but it must use
   one of the 100 numbers that have been assigned to it as the caller
   ID.  The TG object - literally a JSON document returned by the server
   - also called he representation of the TG - would provide this
   information to the client.

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   In the case of a trunking use case where an enterprise customer is
   implementing the server and client roles, the enterprise, acting as
   client, can inform the provider of its own TGs through a simple TG
   registration.  Consequently, the RIPT protocol allows a server to
   accept TG registrations from a client.  Typically, when an enterprise
   acts as a client and registers its own TGs to the provider's server,
   those TGs do not accept registrations.  In other words, the provider
   (e.g., Verizon) would not register TGs with the enterprise, even
   though the enterprise supports the RIPT server role.

   A handler is a representation of a physical hardware device, such as
   an SBC or IP PBX, or software, which acts as a client and can handle
   calls.  Its primary role is to model the capabilities of the device -
   such as supported media types and codecs.  The server needs this
   information in order to decide how media is to be handled for the
   call.  For any particular call, there is a single handler.  This
   handler is determined when the call is placed, and can change during
   the lifecycle of the call.  For example, if a software SBC instance
   supporting G729 (which is a single handler) places a call, this SBC
   crashes, and a different SBC which only supports G.711 (which is a
   different handler) needs to step in and take over the call, the
   handler would change.  A handler has an advertisement, which is a
   description of its media capabilities (media types and codecs).

   Of course, a single physical device might be represented logically by
   one or more handlers; the mapping of a piece of software or hardware
   to a set of handlers is a matter of local implementation.

   The client registers its handler with the TG.  If a client receives
   services from multiple TGs, it would register that handler to
   multiple TG.

   The final resource is a call, which is exactly what it sounds like.
   Calls are always associated with one and only TG.  When a client
   creates a call, it does so in the context of the TG (i.e., call
   creation is a POST operation on a resource within the TG).
   Similarly, calls are associated with a single handler.  When a client
   creates a call, it tells the server which handler is being used.
   This allows the server to compare the capabilities of the client with
   its own, and make a decision about what media the client should send
   with.  The server communications this directive back to the client.

7.  Deployment Examples

   RIPT enables communications between a pair of entities, which could
   be in different companies or providers.  This enables its usage in
   many use cases where there are bilateral relationships requiring

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   real-time communications.  This section contains several use cases
   which are target use cases for deployment of RIPT.

7.1.  Enterprise Voice Trunking

   In this use case, the provider is a telco that enables connectivity
   to the PSTN.  The customer is an enterprise, utilizing an enterprise
   PBX.  Or, it might be a contact center provider.  This is the use
   case for which SIP trunking is widely deployed today.

   To support this case, both the provider and the customer implement
   both the client and the server roles, as shown below:

                   Customer A                Provider B

                +-------------+  From      +-------------+
                |             |  A to B    |             |
                |             |            |             |
                |  Client     | +--------> |  Server     |
                |             |            |             |
                |             |            |             |
                +-------------+            +-------------+

                +-------------+ From       +-------------+
                |             | B to A     |             |
                |             |            |             |
                |  Server     | <--------+ |  Client     |
                |             |            |             |
                |             |            |             |
                +-------------+            +-------------+

   In this use case, a key goal is to facilitate automated configuration
   of SIP trunks - a process which is, today, complex and error prone.
   This section discusses how RIPT can make such configuration as easy
   as a web app login.

   Consider a simple use case of a small business hosting a single IP
   PBX server, on a single machine, which is being used to make and
   receive calls.  To set up the IP PBX for peering, the administrator
   would go to the configuration web application on the IP PBX.

   The web application would have a page for selecting their trunking
   provider, with a list of supported providers.  The admin clicks on
   "Comcast".  This begins a standard OAuth flow.  The admin would be
   directed to the login page for Comcast, provide their credentials,
   and then be asked to authorize the IP PBX to access their Comcast

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   account on their behalf.  It would also display the permissions being
   requested (which Comcast would define and the IP PBX would request).
   In this case, probably the ability to make and receive calls on their
   behalf.  The admin would click on "approve".  The standard OAuth flow
   would then have deposited an OAuth access token on the IP PBX.

   The IP PBX would use the API defined below to obtain the list of TG
   URIs which are available for that admin (which will often just be
   one), and present the admin with the list of such trunks.  The admin
   would select one, if needed.  At this point, the IP PBX now has the
   OAuth token and the provider TG URI, which are the only two pre-
   requisites for placing calls.

   No further configuration or setup is required.  Using RIPT, the IP
   PBX will read an advertisement associated with the TG URI to learn
   its capabilities and features, create and upload its own TG URI and
   bearer token, and now be able to make and receive calls.

7.2.  BYO Voice for CCaaS

   In a related case, an enterprise has purchased voice services from
   Comcast as in the case above.  However, rather than deploying an IP
   PBX, they wish to utilize the services of a Contact Center as a
   Service (CCaaS), such as Five9.  In this example, Five9 supports a
   "BYO" model where the enterprise customer can purchase their voice
   services from another provider - such as Comcast, and then authorize
   a third party - in this case Five9 - to access them on their behalf.

   The flow works almost identically to the above case.  The end user
   would visit, log in using their Five9 account, and click on
   the button to enable Comcast as their BYO provider.  The user would
   be directed to log into their Comcast account, and then authorize
   Five9 to access their Comcast account for making and receiving calls.
   The user would approve, and then setup is complete.

   In essence, a side effect of using OAuth for authentication and
   authorization is that it enables a wide range of access models for
   telecommunications services.  Access can be granted to servers (as in
   the IP PBX case above), or to other cloud services (as in this case),
   with a common mechanism.

7.3.  Inter-Carrier Voice Peering

   In this case, a pair of telcos exchange voice traffic
   bidirectionally.  Today this is done using SIP NNI interfaces as well
   as traditional SS7.  This is accomplished in RIPT by having both
   telcos implement the provider role with a server only, as well as the
   customer role with a client only.  When initiating outbound calls,

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   the carrier would act as a customer, and when receiving them, as a
   provider.  This enables a fully symmetric interconnect.

   The authentication and authorization flow differs in this use case
   from the ones above.  Since it symmetric, each side will need to have
   an account set up for the other.  Administrators for both sides will
   then need to subsequently log into the other side and perform the
   OAuth flow to obtain an OAuth token for the other.

7.4.  Video Endpoint to Meetings Provider

   In this use case, there is a video conferencing device.  The end user
   wishes to join a meeting hosted on a meeting provider.  The meeting
   would be modeled as a call to number at domain, where the number is
   the meeting number used for the meeting.  Using the domain, the video
   endpoint would construct the RIPT root URI and then query to discover
   the available provider TGs.  This request could be authenticated
   using the caller ID in the passport, even though the video endpoint
   may have no relationship with the meeting provider.

   Of course, if the video endpoint were a user of the meeting provider,
   it could have such a token and use it for the call.

   Finally, the video endpoint would place a call to the meeting

   The video endpoint does not need to register a customer TG URI with
   the meetings provider, because it will never receive incoming calls.

8.  Overview of Operation

   The basic operation of RIPT is broken into several steps - bootstrap,
   login, provider TG discovery, customer TG registration, handler
   registration, call establishment, and media exchange.

8.1.  Bootstrap

   Bootstrap is the process by which a client starts with just the
   domain name of the server - such as or or - and uses that to discover the two URI which may be needed
   to do everything else.

   One of these URI is a URI which can be used to kickoff a web-based
   OAuth flow for generating an OAuth token.  The other is the RIPT root
   URI.  Both of these are well-known URI as defined by
   [I-D.nottingham-rfc5785bis].  Consequently, they are constructed
   using the syntax defined in [I-D.nottingham-rfc5785bis].

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   Their IANA registrations can be found in Section 11.

   For example, for, the RIPT OAuth URI is
   <>, and the RIPT root URI
   is <>.

8.2.  Login

   Login is the process by which an OAuth token is delivered to the
   client.  This step is performed only when the client role is being
   played by the customer.  If the client role is being player by the
   provider, this step is skipped.

   When login is done, it is always done through some technique outside
   of the scope of RIPT.  If the use case is to be done where the user
   is in front of a browser, the application can direct the user to
   visit the OAuth page learned from the discovery process above.

8.3.  TG Discovery

   The client needs to figure out the set of TG which are available to
   it.  There may be cases where the server is offering the client more
   than one.  For example, a telco might offer an enterprise an
   international TG which is used to place calls to non-US numbers, and
   a domestic TG which is used for domestic calls.

   To obtain the list, the client queries the /providertgs resource on
   the well-known RIPT root URI:


   this URI will return the list of TG available to the client.  This
   list has, for each, the TG URI and a name and description in prose,
   meant for a troubleshooting.  The client would normally select a TG
   automatically on a call by call basis based on local policy.  For
   example, if the server provides three TGs, one for receiving calls to
   international numbers and one for domestic numbers, the client would
   choose the TG based on whether a call it wishes to make is towards an
   international or domestic number.

   An example TG supporting outbound dialing from a 2-line IP PBX to
   domestic numbers might look like this:

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      "outbound": {
        "origins" : "RFC 8226 cert with +14085551000 and +14085551002",
        "destinations" : "+1*"

   An enterprise trunk allowing outbound calls to any number worldwide,
   using one of the 1000 numbers assigned to that enterprise, might look
   like this:

              "outbound": {
                "origins" : "(RFC 8226 cert with +14085551* ",
                "destinations" : "*"

   In most cases, the server will require the client to authenticate in
   order to retrieve this document.  Typically, a server would have many
   TGs provisioned, and each TG is associated with a specific customer
   which will connect to it.  The customer ID would be associated with
   the TGs it is permitted to access, and that same customer ID would
   normally be placed into the OAuth token that the client obtains
   during login.  This allows the origin server to determine who the
   customer is, and what TG(s) they are allowed to connect to.  This is
   standard procedure for most web APIs when a query is performed - the
   set of resources returned depend on who is asking.

   Open Issue: how to say all numbers except satellites, or all numbers
   except 1-900.

8.4.  Customer TG Registration

   If the client role is being played by a customer who wishes to
   receive incoming calls, it will need to implement the server role of
   RIPT.  Its customer TG needs to be registered with the server.  To do
   this, it performs a POST to the /customertgs resource on the RIPT
   root URI, and sets two parameters - its own URI, and a bearer token
   used for authorization of inbound requests.

   The customer TG URI has to be reachable by the server in order for
   the it to receive calls, and for security purposes it must also
   support TLS and present a valid domain certificate using the same
   trust chains configured into browsers.  This specification envisions
   an extension which allows a client to receive calls without requiring
   it to run an HTTP server, thus allowing for it to run behind a NAT
   and not have a certificate.

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   An example of a customer TG registration might be this:

           "outbound": {
             "destinations" : "+14085551*"

   Notice how the directionality is "outbound" - this is because a TG
   only supports outbound calls, and from the perspective of the
   provider towards the customer, these are outbound.  Similarly, note
   how the "destinations" parameter indicates the range of numbers
   reachable via this TG.  The provider would validate that these are
   authorized based on prior business relationship, and reject them
   otherwise.  For example, if the customer indicated it wanted to
   receive calls for numbers which were not obtained from that provider,
   the provider's terms of service may not allow that, and it would
   reject the request.  Also note how the origins field is absent - this
   is because the enterprise will accept inbound calls with any

   A success response to this would be a 201 Created, with the TG which
   was created:

    "uri" : "",
    "outbound": {
      "destinations" : "+14085551*"

   The client can use the /consumertgs to modify this later (with a PUT
   to the URI in the "uri" parameter), DELETE it, or create another.

   The server holds onto the consumer TG URI, until such time as the
   consumer is no longer receiving service from the provider.

8.5.  Handler Registration

   The handler is a representation of a device.  Each device has a set
   of capabilities that can be used when placing a call, which are
   contained in its advertisement.  The handler description is a semi-
   static declaration which contains the advertisement along with other
   properties of the handler.  Handler descriptions are semi-static in
   that they do not change on a call by call basis.  They change only
   when some kind of significant configuration change happens.  For
   example, if an SBC receives an upgrade to support a new codec, its

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   handler description would change.  Consequently, handler descriptions
   are expected to change perhaps a few times a year.

   A handler registration is created by having the client perform an
   HTTP POST operation to the /handlers resource on the TG URI.  This
   operation creates a new handler instance on the server and returns
   its URI to the client.  The client is expected to store this URI in
   order to make modifications at a later date (via PUT), or DELETE it.
   The handler registration is not soft state, there is no need for the
   client to refresh it.  The server would typically delete this
   registration when the customer ends its service with the provider.

   The advertisement has a list of media sources and sinks that the
   endpoint has, and an ID for each which monotonically increases from
   1.  Furthermore, each source and sink is of a particular type - audio
   or video.  An advertisement can contain more than one of each.  The
   case of PSTN gateways or traditional voice-only phones is simple -
   they have a single source for audio and a single sink for audio, This
   is true for a PSTN gateway regardless of its capacity.  In other
   words, if a PSTN gateway has a circuit switched line card with 100
   ports, its advertisement still has just one source and one sink.

   A three-screen telepresence system might have three sinks for video,
   one source for audio, two source for video representing a main camera
   and a presentation video feed, and one sink for audio, and represents
   the opposite end of the spectrum in terms of complexity.

   For each codec usable by each source or sink, there are one or more
   parameters that can be specified.  Each parameter in the parameter
   set has a name and a value.  The value is always an integer from -
   2*63 +1 to 2*63 - 1.  Parameters are typically standardized and
   registered with IANA.  The registration indicates the meaning of the
   values - their units, default, and allowed values.  Most importantly,
   the parameter is always expressed in a way where the value represents
   a maximum of some sort.  This enables booleans (where the maximum is
   1), integral ranges (where the maximum is a large integer), or
   ordered enums (where the enum values correspond to integers in
   order).  When a parameter is not specified, it takes on a default.
   Similarly, if the parameter is not present, the default can be

   Codec support is signaled using names that match the media subtypes
   defined in the IANA protocol registry for media types [RFC4855].  If
   the source or sink is audio or video can be determined from the
   codec(s) associated with it.

   They syntax for the advertisement consists list of descriptions for
   each source and sink.  Each description starts with the source or

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   sink ID followed by a direction of in or out then a colon followed by
   a semicolon list of codec descriptions.  Each codec description
   consists of the name of the codec followed by a comma separated list
   of parameters for that codec on that source or sink.

   Open Issue: Consider if JSON would be better syntax or define mapping
   between this and JSON.  This is syntax is optimized for developer
   debugging of what has proven to be one of the difficult parts of
   debugging WebRTC.

   An IP phone with a single microphone and speaker that support G.711
   and opus might create its handler thusly:

     "handler-id": "982akca99283",
     "advertisement" : "1 in: opus; PCMU; PCMA;
                                  2 out: opus; PCMU; PCMA;"

   and the reply would be:

201 Created

   Notice how the server as added the "uri" parameter.  The "handler-id"
   is client-specified and allows for correlation across different TGs
   for the same handler.

   A device with a camera that could support H.264 at 4K and av1 at
   1080p might have an advertisement that looked like:

1 out: H264,max-width=3840,max-height=2160; AV1,max-width=1920,max-height=1080;

   A video phone that could support opus and H.264 at 720p @ 60 fps
   might have an advertisement which looks like:

           1 in: opus;
           2 out: opus;
           3 in: H264,max-width=1280,max-height=720,max-fps=60;
           3 out: H264,max-width=1280,max-height=720,max-fps=60;

   As with any kind of capabilities technology, the one defined here
   cannot describe arbitrary constraints.  It cannot express
   dependencies that may arise between audio and video, or between
   encoding and decoding.  It, in essence, represents each as an OR-of-

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   ANDS construct, and while all boolean expressions can be reduced to
   this form, it will produce extremely unwieldy results with complex
   expressions.  It cannot represent ranges of values.

   We believe this represents the minimum technique which can be used to
   describe modern AV systems.

8.6.  Obtaining Certificates

   In order to place an outbound call, a client must have a certificate
   which is valid and attests to the number that is to be used.  In
   order to enable that, RIPT requires that a client obtain a passport
   for each number it wishes to place a call from.

   To do that, the server provides a "/cert" resource on the TG.  The
   client can perform a POST to this resource, and the request would
   contain a CSR and the response would have the certificate.  The CSR
   describes an [RFC8226] certificate which is valid for a single phone
   number - the one the client wishes to use for a call.  This request
   is, as with all others, authorized with the OAuth token.  If the
   holder of the token is permitted to utilize that number in phone
   calls, the server generates a certificate and returns the certificate
   in the POST response.  These certificates can last as long as the
   server desires.  This means that acquisition of such certificates can
   be done in advance of the call, and regardless of when they are
   acquired, can be cached for subsequent use.

   Open Issue: Do we need an async mechanics to get the certificate.

   In the case of a mobile phone as the RIPP client (for example), it
   would learn its own mobile number from the TG, construct a CSR for
   that number, and request the server to create a certificate that it
   would cache until it expires.

   An important benefit of this approach is that the server can use the
   passport itself as the technique to authenticate the client as
   defined in [RFC8224], and merely pass on the passport provided by the
   client rather than generating one.  This avoids the computational
   cost of creating a signature, and also means that secure caller ID is
   provided by default, without requiring active action on behalf of the
   server.  If the server does nothing, the call still has a valid

8.7.  Call Establishment

   The client can initiate calls by POSTing to /calls on the TG URI.
   The request contains:

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   1.  the target phone number or email address (TODO: need to define
       normalization procedures),

   2.  A passport [RFC8225] identifying the calling identity,

   3.  The handler URI from which the call is being placed,

   in the body.

   For example, to place a call to a phone number from the handler

    POST "
      "handler": "
      "destination": "+14089529999",
      "passport": "{passport encoding}"

   The server takes the advertisement from the client's handler, takes
   its own advertisement (which it has never exchanged, but merely
   knows), and figures out what it will send, and what the client must
   send.  It then constructs two directives - the client directive
   indicating what the client must send, and the server directive which
   describes what it will send.  The directives have the similar syntax
   as the handler descriptions.  They contain a set of streams, each
   with a source and the ID of the sink on the remote side which is
   included in the media packet so that the far side understands where
   to render the media.  For the client directive, the source specifies
   one of the sources in the client handler, and the sink specifies a
   sink on the server.  It is the inverse for the server directive.  For
   For each stream, there is one parameter set, and for each parameter,
   the value indicates the upper bound of what client can sent.  Each
   directive is always specified in a way that makes the value of each
   parameter less than the maximum value between the advertisements from
   the client and server.

   The server places the call, and returns the call description back to
   the client.  The call description includes the directives along with
   core meta-data about the call - directionality, handler, caller,
   callee and a URI for the call.  The server directive is optional and
   does not need to be known by the client, and it is not processed in
   any way.  It is included only for diagnostic and troubleshooting

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   The syntax for the description is simular to the advertisement except
   that the lines starts with the id of the source on the client,
   followed by "to" and the ID of the sink on the remote side.

201 Created
  "uri" :
  "clientDirectives": "1 to 1: opus;
                              2 to 2:  H264,max-width=1280,max-height=720",
   "serverDirectives": "1 to 1: opus;"

   Note how the client and server directives are both for Opus.  Also
   note how the server directive has a single stream for audio only.
   Clients MUST always prepared to receive media for any sink they've
   specified in their advertisement.

   In the (unlikely) case that this directive cannot be followed (due,
   perhaps to a unexpected change in capabilities as a result of a GPU
   or CPU spike), the client updates its handler with updated
   capabilities.  To tell the server to create a new proposal for the
   call, it performs a POST against the existing call URI, this time
   without parameters, and the server will respond with an updated call
   description, including the new directive.

   Another important consequence of this design is that media packets
   must be self-describing, without any kind of reference to a specific
   call.  This is because the directive is constructed from the handler
   descriptions only, and the handler descriptions are semi-static.
   This means RIPT does not use dynamic payload types to identify

   Once a call has been created, a pair of long-lived HTTP transactions
   is initiated from the client to the server for purposes of signalling
   (this only happens if the call was created successfully and the
   directive could be followed).  One is a GET to the /events resource
   on the call URI, retrieving call events from the server.  The other
   is a PUT to the same /events URI, used by the client to send call
   events to its peer.  The combination of these two is called the
   signalling byway.  HTTP/3 allows for an zero RTT for setup of these

   Signaling commands are encoded into the signalling byway using
   streaming JSON in both directions.  Each JSON object encodes an event
   and its parameters.  A set of events common to all deployments of
   RIPT are defined for proceeding, alerting, answered, declined, ended,
   migrate, moved, ping, and pong.  An additional set are defined

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   targeted at server to server cases, such as SIP trunking and inter-
   server peering.  These include transfer-and-takeback.

8.8.  Media Exchange

   Media exchange makes use of web-transport over HTTP3
   [I-D.vvv-webtransport-http3] when it is available, falling back to
   media byways when it is not.

   Once the call signaling is complete, the client attempts to open a
   webtransport session within the current HTTP3 connection.  If this
   succeeds, the client and server can both send media chunks as
   webtransport datagrams.

   If the connection does not support webtransport, the client takes
   responsibility for opening media byways, which carry media chunks in
   both directions.  Even though data can flow in both directions, a
   media byway is unidirectional in terms of media transmission.  A
   forward media byway carries media from the client to the server, and
   a reverse byway carries media from the server to the client.  These
   map directly to HTTP PUT and GET respectively, as they do for
   signaling byways.

   Unlike signaling byways where sequenced, in-order and reliable
   delivery is desired, these are undesirable for media.

   To eliminate them for media in the forward direction, for each and
   every media chunk to send, the client will create a new PUT
   transaction, send the media chunk, and immediately close the
   transaction.  When run over HTTP3, this process can take place with
   zero RTT.  When the server receives the request and associated media
   chunk, it immediately sends a 200 OK response, includes a payload
   containing an acknowledgement media chunk, thus completing the
   transaction.  Since HTTP3 will use a different QUIC stream for each
   HTTP transaction, each of these transactions occurs on a different
   stream, and therefore there is no HOL blocking between media packets.

   Receiving media is more complex, since the server cannot initiate
   HTTP requests in the reverse direction.  To solve this, the client
   initiates 20 GET requests to the server to retrieve media chunks.
   These GET requests contain no body.  The server, when it wishes to
   send a media chunk, chooses an existing GET transaction and sends the
   chunk as a response body.  The client, upon receiving this,
   immediately opens a new GET transaction to replace the one it just
   received a response on.  Consequently, a client always maintains 20
   open GET transactions.  TODO: need to tune this and justify its
   value.  In addition, the client will include an acknowledgement media
   chunk along with its next media chunk in a PUT request.

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   The use of acknowledgements provides the ability for clients and
   servers to elect to retransmit media as well as to generate detailed
   statistics on packet loss and delay on a packet-by-packet basis.
   Consequently, the equivalent of RTCP sender and receiver reports are
   not needed.

8.9.  Graceful Migration

   RIPT provides a simple technique for allowing a call to gracefully
   migrate from one client instance to another on a different host, or
   from one server instance to another on a different host.  The client
   always performs the migration action.  It can be instructed by the
   server to do so via a migrate event.  Or, it can decide to do so on
   its own.

   In the first case, there is a traditional HTTP load balancer fronting
   a farm of HTTP origin servers, and the goal is to move the call to
   any one of the available instances behind the load balancer.  A
   server can request this in order to shed load.  Or, it can request
   this because it has been instructed to shut down and restart, perhaps
   to perform a software upgrade, or to migrate its docker container to
   another node.  It is assumed that some kind of shared database is
   used to store any state that needs to be passed between origin
   servers.  RIPT is designed such that the only state which needs to be
   stored is call state, and not media state.

   In this use case, if the origin server handling the call does not
   desire to receive new calls (including the one it is about to
   migrate), it would be removed from the pool of available servers in
   the load balancer.  The means for this is outside the scope of this
   specification.  Then, the origin server sends a migration event to
   the client.  No additional attributes are provided in the event.

   Alternatively, the server may wish to migrate the call to a specific
   instance amongst those in the cluster.  One example of that is to
   move calls from specific users or domains to specific hosts for
   reason of isolation.  There are many other reasons.  In this case,
   the event from the server will contain a URI which replaces the
   existing call URI, since it has a different authority component.

   Either way, the receipt of this event will cause the client to end
   its current signaling and media transactions.  It then initiates new
   transactions for signaling (the GET and PUT to /events), and media
   (the GET and PUTs to /media), targeting the URI for the call or the
   replacement URI for the call if the server provided one in the
   migrate event.  Some load balancers use cookies to keep subsequent
   transactions "sticky" to a given backend, In RIPT, cookies are
   handled in the same way cookies are handled for any other HTTP

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   session so that load balancers can change the cookies to stick to a
   new backend server to enable sticky session routing for subsequent
   requests for this call.

   Whether client or server initiated, when a migration occurs, both
   sides buffer their media packets and signaling events until the
   byways are ready to carry them.  This enables the migrations to occur
   with minimal impact to the media.  The migration takes only a single
   RTT between client and server, and thus can occur within the
   boundaries of jitter buffer depths.

   Note that the call state persists independently of the state of the
   HTTP connection or the byways embedded in HTTP transactions, so that
   a reconnect can continue where things left off.  This is why, when a
   call migrates, the client does NOT initiate a new call, it opens
   signaling and media byways to the existing call URI it already has,
   or the new one that replaces it.

8.10.  Non-Graceful Migration

   RIPT is also designed to handle failures of clients, servers and
   network connectivity issues, and to reconnect calls with a minimum of
   disruption.  It does this by providing implementors with tools to
   enable such resiliency, without being overly prescriptive on exactly
   how it is implemented.

   The asymmetric nature of RIPT means that it is always the
   responsibility of the client to recover from network failures and
   failures of a downstream server or load balancer.

   The first failure case is that of a closure of the actual HTTP
   connection, either gracefully or non-gracefully.  In this case, the
   client retries the connection with an exponential backoff, the
   backoff timer being another parameter which can be specified in the
   TG.  Once the connection is re-established, it initiates the
   signaling and media byways, and will reuse any session cookies it had
   previously received.

   RIPT also requires clients to send keepalive signaling events
   periodically for each call, and the server responds to these events.
   If the client ceases to receive the server keepalives for a certain
   duration (a value again that has a default but which can be tuned in
   the TG description), it treats this identically to a request for a
   migration.  This will cause it to end its signaling and media byway
   transactions, and re-initiate them without session cookies.

   Finally, the media byways are used for rapid failure detection.  If
   the client fails to receive acknowledgements for the media it has

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   sent for period of time (this timer not being subject to
   standardization as it is entirely a matter of local configuration),
   the client can re-establish the signaling and media byways as if it
   were a migration.

   The above cases handle failures in the network and of the origin
   servers and intermediate load balancers.  If there was a failure of
   the origin server and it was not able to recover sufficient call
   state for re-establishment, it rejects the re-establishment of the
   media and signaling byways.  This causes a call termination from the
   perspective of the client, which it would typically pass upstream if
   needed, as an event.  And indeed for the origin server to recover, it
   will need to stash state.  Specifically, it will need to stash the
   callID and the downstream entity to which the call was set when it
   was established.  If the downstream entity is a RIPT server, the TG
   URI, call URI, selected IP address (from either DNS or using the IP
   in the advertisement), and session cookie are stored.  These
   facilitate reconnection to the same downstream instance.

   In inter-company peering arrangements, such as enterprise voice
   trunking or inter-carrier NNI voice peering, both sides will
   typically have a cluster of software agents (VMs, docker containers,
   bare-metal servers, dedicated hardware products, or whatever) acting
   as both clients and servers.  Consequently, we need to consider what
   happens to the client transactions when these fail.

   In such a case, failure recovery is always delegated to the upstream
   client.  In this case, let us assume that the client had this cluster
   of agents at its perimeter.  It would be receiving calls from some
   further upstream elements.  If those elements were using RIPT, it
   would mean that the roles were reversed - a particular agent was
   acting as a server to receive a call, and then as a client to send it
   downstream.  If that agent fails, the upstream client would detect
   this, and migrate to a different agent.  This new agent - which has
   no in-memory state for the call - would look up the state of the call
   in some shared database, see where it was routed (in this case, to
   the TG), and retrieve the session cookie which had also been stored
   after the call was established.  This new agent then re-establishes
   the media and signaling byways, perhaps even opening a fresh HTTP
   connection to do so.  The use of the session cookie enables the
   byways to be re-established to the server's origin server that has
   been, and is still - handling the call.

   This allows a chain of RIPT clients and servers in back-to-back
   configuration to recover from failures of any intermediate element
   without dropping calls.  Furthermore, failure detection and recovery
   are rapid, especially when using load balancers.  When an origin
   server fails, the client can still reuse its connection to the load

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   balancer, such that connection establishment is not needed.  Indeed,
   the time to recover from failure is only 1/2 RTT between client and
   server, once the client has detected the downstream failure.

   Of course, the downstream element may be SIP-based and not RIPT.  In
   such a case, INVITE with Replaces is a good solution, and vendors of
   highly reliable SIP gear are encouraged to support it.

9.  Normative Protocol Specification

   This section contains the normative specification of RIPT.

9.1.  Bootstrapping

   A RIPT client that wishes to obtain an OAuth token to a specified
   authority through a web interface MUST construct a well known RIPT
   Oauth URI using [I-D.nottingham-rfc5785bis] with ript-oauth as the
   well-known service, and use this to trigger the Oauth process.

   Similarly, a RIPT client wishing to access the resources defined in
   this specification, against an authority (such as MUST
   use [@!I-D.nottingham-rfc5785bis} with ript as the well-known
   service.  The result is the RIPT root URI for that authority.

9.2.  TG Discovery

   All clients MUST perform a GET query to the /providertgs on RIPT root
   API for an authority they wish to connect to.  Origin servers MUST
   provide a resource at this URI.  It MUST be constructed using the
   JSON syntax described in Section 10, and MUST have one or more TG URI
   in the list.  It is RECOMMENDED that the origin server specify that
   this document can be cached.

   If the client receives a document and there is only one TG URI, it
   uses this for subsequent outbound calls to the server.  If there is
   more than one, the client can use any algorithm and policy it desires
   to choose the target TG for any call.

   At the end of this process, the client will have a TG URI.  It MUST
   retrieve the value of this URI, and use it to process calls.

9.3.  TG Construction

   When a server (either for the consumer or the provider) retrieves a
   GET against the a valid TG URI, it MUST return a document with a set
   of parameters.

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   The document MUST contain an "outbound" element.  The "origins" field
   specifies the permitted caller ID values which can be present in the
   passport used in a call setup towards this TG.  If the server will
   act as an authentication service as defined in [RFC8224], it MUST
   include a value for this parameter.  When present, it MUST be an
   [RFC8226] certificate, whose TelephoneNumberRange indicates the
   numbers which the server can vouch for.  In cases where the server is
   not acting as the authentication service, the "origins" field MUST be
   absent.  This would be the case in inter-carrier peering links, or in
   a consumer TG registered to the provider.

   The "destinations" field specifies the allowed targets for calls.
   The server MUST include this element if it will reject a call based
   on policy for a specific destination.  The default is "*" meaning the
   TG will accept any calls.

   In addition, the TG URI contains a set of configuration values.  If
   absent, these take their default.  The following are defined:

   o  retry-backoff: In the event of a closure of the HTTP connection,
      this parameter specifies the initial time value to use before
      retrying the connection.  If a retry fails again, the client will
      try again but this time wait twice the value of this timer, then
      four times, eight times, etc.  The value of this parameter is an
      integer, in units of milliseconds.  Its default is 2000, and the
      client MUST NOT honor values less than 2000, rounding up to 2000

   o  media-timeout: If a client fails to receive media ack packets
      after the timeout specified in this parameter, it considers the
      call dead and initiates migration.  The value of this parameter is
      an integer, in units of milliseconds.  Its default is 5000.

   OPEN ISSUE: Need to consider realistic ways to incrementally
   introduce this into carrier networks without requiring massive
   forklifts of new load balancers, anycast and so on.

9.4.  Consumer TG Registration

   If an entity needs to receive inbound calls from its provider, it
   MUST register a consumer TG.  To do that, it MUST initiate an HTTPS
   PUT request towards /consumertgs on the TG URI.  This request MUST be
   authenticated by the origin server.  The request MUST include a TG
   description object in the body.  This object is specified in
   Section Section 10.

   The TG description object MUST contain a consumer TG URI and a bearer
   token.  The client TG URI MUST be unique for each distinct provider

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   TG URI to which it is registered.  This URI MUST support HTTP/3, and
   MUST implement the /handlers, /calls, and /events resources and their
   associated behaviors.  This URI MUST be reachable by the provider.
   The URI MUST utilize HTTPS, and MUST utilize a domain name for the
   authority component.

   The destinations parameter in the consumer TG description MAY be
   included.  If it is included, the destinations MUST be a subset of
   the addresses present in the certificate found in the origins element
   in the provider TG description.  Since a consumer does not act as an
   authentication service as specified in [RFC8224], the origins
   parameter MUST be omitted.

9.5.  Handler Registration and Lifecycle Management.

   Prior to placing a call towards a TG, a client MUST have an active
   handler registered to the server.  To register one, it performs a
   POST on the /handlers resource on the TG URI.  This request MUST
   contain a handler description conformant to the syntax in Section 10
   and following the rules defined in the next section.

   If the request is valid and authorized, the origin server MUST return
   a 201 Created response, with the URI for the new handler in the
   Location header field.  It MUST echo back the handler description,
   and MUST add or replace the "uri" parameter in that description to
   contain this URI.

9.6.  Handler Description Format

   An handler description has a "handler-id", which is a unique
   identifier for the handler on the client, and then an advertisement.

   The advertisement contains, for each codec on each source or sink, a
   set of parameters, each of which is a name-value pair.  This
   specification defines several well-known names and establishes an
   IANA registry for future extensions.  Every capability has a default,
   so that if it is not included in the advertisement, the capability
   for the peer is understood.

   The advertisement and directives are a list of source or sink
   descriptions.  Each one starts with a source or sink ID followed by
   'in' or 'out' for advertisements and followed 'to' and a remote
   source ID for descriptions.  Next is a colon followed by a semicolon
   separated list of codec descriptions and includes a semicolon at end.
   Each description starts with a codec name followed by a comma
   separated list of parameter values.  Codecs names MUST not start with
   a digit.  Each paramater consists of a name optional followed by qual

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   sign then numeric value.  A parameter with no equal sign after is
   consider to be set to value of 1.

   There MUST be one "source" and "sink" instance ID for each
   corresponding source and sink which can simultaneously send or
   receive its media in a single call.  Instance ID start at 1 with a
   general convention that the lower numbers would be more important
   than the upper numbers and for video of equal importance, the
   convention is to increase the ID for viewers left to right then top
   to bottom.  Each parameter MUST specify the maximum that the sink can
   receive, or source can send, for that parameter.  A client or server
   MUST include a parameter and its value when it differs from the
   default, and SHOULD NOT include it when it matches the default.

   This specification defines the following parameters for audio:

   o  sr: The maximum sample rate for audio.  This is specified in Hz.
      The default is 48000.

   o  ss: The maximum sample size for audio.  This is specified in bits.
      It MUST be greater than or equal to 8.  The default is 16.

   o  cbr: If present, indicates whether the entity requires constant
      bitrate (CBR) media only.  It has value of 1 (indicating CBR only)
      or zero (no constraint on rate).  Zero is default.

   o  ch: Indicates whether the entity supports multiple audio channels.
      The default is 1 for support.

   o  ptime: max duration of media encoded in single packet in ms.
      Default is 30.

   This specification defines the following parameters for video:

   o  fps: The maximum frame rate for video.  This is specified in
      frames per second.  The default is 30.

   o  max-res: the max width in pixels.  Default is TBD.

   o  pr: the max pixel rate ( pixels / second ).  Default is 2^64-1.

   o  depth: the max pixel depth in bits.  Default is 8.

   This specification defines the following capability set for scalable
   video codecs:

   o  tlay: max number of temporal layers.  Each sub layer runs at twice
      the rate of the later it depends on.  Default is 1.

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   o  slay: max numer of spacial layers.  Each sub layer has 2 times the
      number of pixel of layer it depended on.  Default is 1.  (Open
      issue 2 times or 4 times)

   This specification defines the following capability set for both
   audio and video codecs.

   o  br: The maximum bitrate in kilo bits per second for receiving
      audio or video.  This is specified in bits per second.  This is
      the rate of encoded media from the codec and not the rate
      transmitted over the network.  Default is 2^64-1.

   o  codec names - The codec names are defined in the IANA registry
      "Media Types" at <
      media-types.xhtml>.  For each one, the value of the parameter is
      either 1 or 0, 1 indicating support, and 0 indicating no support.

   All RIPT implementations MUST support G.711 and Opus audio codecs.
   All implementations MUST support [RFC2833] for DTMF, and MUST support
   [RFC3389] for comfort noise, for both sending and receiving.

   In general, an entity MUST declare a capability for any
   characteristic of a call which may result in a proposal being
   unacceptable to the client.  This requirement facilitates prevention
   of call failures.

9.6.1.  ABNF

   TODO - Add ABNF for advertisements and directives

9.7.  Certificate Enrollment

   Prior to placing a call, a client MUST have a valid [RFC8226]
   certificate which has a subject field that contains the phone number
   or email address from which the call is to be placed.

   A server acting as an authenticator as defined in [RFC8224] MUST
   implement the "/certs" sub-resource of its TG.  A client wishing to
   obtain a certificate can perform a POST operation to this resource.
   The body MUST contain a valid CSR.  The server MUST validate the CSR
   is valid, and MUST validate that the phone number or email address
   that is requested, is amongst ones that the server is willing to
   vouch for.  It MUST generate a certificate and return it in the 200
   OK response to the request.

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9.8.  Call Establishment

   To place a call, the client performs a POST request to /calls
   resource on the TG URI URI.  This request MUST include a body,
   formatted according to the syntax of Section 10.  The body MUST
   contain the "handler", "destination" and "passport" values.

   The handler value MUST be a valid HTTPS URI and MUST point to a
   hander registered and active against the same authority in the POST's
   request URI.  If there is no matching handler currently registered,
   the server MUST return a 500.

   The destination MUST be either an E.164 value or an email address.
   If an E164 address, it SHOULD be a subset of the addresses defined in
   the "destinations" parameter of the TG description.  A server MUST
   reject a request whose target does not so match.  RIPT also supports
   private trunks, in which case the destination MUST take the form
   "<number>@<domain>", where the number is a non-E164 number scoped to
   be valid within the domain.  This form MUST NOT be used for E.164

   The passport URI parameter MUST be a valid passport as defined by
   [RFC8224].  It identifies the calling party and includes signature
   information which can be used to verify it.  The caller ID and called
   party values in the passport MUST be within the allowed values
   defined in the "origins" and "destinations" parameters of the TG,

   The server MAY authorize creation of the call using any criteria it
   so desires.  If it decides to create the call, the server MUST return
   a 201 Created response, and MUST include a Location header field
   containing an HTTPS URI which identifies the call that has been
   created.  The call URI MUST contain a UUID.

   The server MUST construct a client directive, which tells the client
   what media to send.  This directive MUST include zero or more stream
   elements, each of which MUST specify a source on the handler
   specified by the client, and a sink on the server.  The server MUST
   specify the values for any codec which are not the default.  The
   value for each parameter MUST be less than the value specified in the
   handler description from the client, and also MUST be less than its
   own maximum value from its own handler description.  If a selected
   value differs from the default, it MUST be included in the directive.

   Similarly, the server chooses which of its sources will send, and to
   which sinks on its peer it will send.  It MUST NOT send media for
   which there is not a corresponding sink on its peer which is a match
   for the media type.  It MUST send utilizing media parameters which

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   are less than the values specified in the handler description in the
   peer for that sink.  It MUST codify this into a server directive
   describing what it will actually send.

   The server MUST include the client directive in the body of the 201
   response, MUST include the URI for the handler that was used, MUST
   include the call direction, MUST include the from and to
   participants, and MUST include its own server directive.

   If the request is otherwise valid, but the target of the call cannot
   be reached through the TG URI, it MUST generate a 404 response.  If
   the request is valid and the target can be reached, but the client is
   not allowed to do so for policy reasons it MUST generate a 403

9.9.  Signaling and Media Byway Establishment

   If the server returns a 201 Created to the client, the client MUST
   establish the forward and reverse signaling byways by sending a PUT
   request and GET request, respectively, to the /events resource on the
   call URI obtained from the Location header field in the 201 response.
   The GET request MUST NOT contain a body.  The PUT request and GET
   response MUST contain an infinitely long JSON document.  The client
   MUST hold both transactions open until the call has ended.

   The body begins with an open square bracket to indicate the start of
   an JSON array, and after that is a series of JSON objects, each
   starting with a curly bracket, and ending with a curly bracket.
   Consequently, each side MUST immediately send their respective open
   square brackets after the HTTP header fields.  We utilize streaming
   JSON in order to facilitate usage of tools like CURL for signalling

   The server MUST immediately send the current state of the call as far
   as it is concerned, by generating an event in the response which
   indicates this state.  For a brand new call, this MUST be the
   proceeding event.  For any call, a server MUST support multiple
   signaling byways, in which case it MUST send all call events on all
   open reverse signaling byways, and MUST accept commands from any
   forward one.  A server MUST NOT terminate the call if there are no
   signaling byways established; rather it utilizes timeouts as
   described below.

   Each event is a JSON object embedded in the signalling stream, which
   conveys the event as perceived by the client or server.  The event
   MUST contain a direction field, which indicates whether the event was
   sent from client to server, or server to client.  It also MUST
   contains a timestamp field, representing universal time, which

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   indicates the time of the event as perceived by the sender.  The
   event MUST contain a call field, which contains the URI of the call
   in question.  The event MUST contain an event type field, which
   conveys the type of event.  This is followed by additional fields
   which are specific to the event type.

   Events are also defined so that the resulting state is uniquely
   defined by the event itself.  This ensures that knowing the most
   recent event is sufficient to determine the state of the call.

   In addition, the client MUST immediately establish 20 reverse media
   byways by initiating 20 GET requests to the /media resource on the
   call URI.  The media byways are required before call answer to
   support early media.  For any call, a server MUST support up to 30
   reverse media byways open.

9.10.  Basic Call State Management

   The server always maintains definitive state for the call.  The basic
   state of the call is manipulated through events passed from one side
   to another.

   This specification defines the following events which MUST be
   implemented by all RIPT client and servers: These events are
   asymmetric, in that the direction of call establishment impacts who
   can send which events, and when.

   proceeding: Passed from server to client, indicating that call setup
   is in progress but has not reached the recipient.

   alerting: Passed from server to client, indicating that the recipient
   is alerting.

   answered: Passed from server to client, indicating that the call was
   accepted by the recipient.  At this point, the call is considered

   declined: Passed from server to client, indicating that the call was
   rejected by the user.

   failed: Passed from server to client, indicating that the call was
   rejected by server or downstream servers, not by the user, but due to
   some kind of error condition.

   noanswer: Passed from server to client, indicating that the call was
   delivered to the receiving user but was not answered, and the server
   or a downstream server timed out the call.

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   end: initiated by either client or server, it indicates that the call
   is to be terminated.  Note that this does NOT delete the HTTP
   resource, it merely changes its state to call end.  Furthermore, a
   call cannot be ended with a DELETE against the call URI; DELETE is
   not permitted and MUST be rejected by the server.

   migrate: sent from server to client, it instructs the client to
   terminate the media and signaling byways, and re-establish them.  The
   event MAY contain an URI which replaces the current call URI, thus
   indicating the destination to which the media and signaling byways
   will be established.

   ping: This event is always initiated by the client.  When received by
   a server, the server MUST generate a pong response.  The ping MAY
   contain a nonce, and if so, the server MUST echo it in the pong

   The client can obtain the current state of the call at any time by
   querying the call URI.  The server MUST return a call description
   which includes the client and server directives, call direction,
   calling and called parties, and the handler which is being used.
   This aspect of RIPT is essential for allowing clients to fail,
   recover, and re-obtain the state of the call.

   When a server has a signaling event to send, and there are multiple
   GET requests open for /events, the server MUST send the event on all
   such open requests.  This enables seamless migration.  Similarly, the
   server MUST be prepared to receive events on multiple distinct PUT
   transactions, and process them in the order in which they are
   received by the server.

9.11.  Sending and Receiving Media

   Media is always associated with a call.  Within a call, media has a
   direction (c2s or s2c).  Within a direction, media belongs to a
   stream, where a stream is transmitted between a source and a sink.
   As such there can be many streams in each direction.  A stream is an
   ordered sequence of media chunks.  There is a sequence number space
   and timestamp space associated with each stream.  A stream is
   uniquely identified by the call URI, the direction, the source ID and
   sink ID.

   The forward and reverse byways each contain an unordered sequence of
   media chunks and control chunks.  The only control chunk defined by
   this specification is the acknowledgement chunk.  Any media or
   control chunk, for any stream, MAY be sent in any forward or reverse
   byway for c2s and s2c streams respectively.

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9.11.1.  The Media Chunk Format

   Each ript media chunk has an outer envelope that is not encrypted,
   except by the transport the chunk is sent over, and an inner package
   that is encrypted.  Inside both the envelope and package are a set of
   tag length values (TLV) tuples.

   Each media chunk has the following properties in the envelope part:

   o  Media or Control: a boolean which indicates whether this is a
      media or control packet.  The remaining parameters apply when it
      is media.  If not present, defaults to media. (tag=TBD)

   o  The truncated sequence number, which provides the absolute
      ordering of the packets for the codec.  The sequence number is
      actually 64 bits long, but the encoding is of variable length.
      This allows for any number of bits to be sent on the wire.  If N
      bits are sent, where N is less than 64, the value is treated as
      the truncated version of the actual sequence number, and the
      recipient computes it from prior packets.  Clients MUST send full
      values until such time as they receive an acknowledgement of the
      media packet containing a full value, and after that, SHOULD send
      truncated values.  For video, if the media is an intra-frame, the
      full value MUST be present. (tag = 1)

   o  The truncated timestamp, which provides the wall clock time in ms
      of when the first sample of media in the chunk was recorded.  Like
      sequence number, this value is represented with 64 bits, but
      truncated values can be sent.  The client MUST send full values
      until the first packet containing a full value has been
      acknowledged, and after that, SHOULD send truncated values.  For
      video, if the media is an intra-frame, the full value MUST be
      present. (tag = 2)

   o  The truncated payload type, which is a 32 bit integer specifying
      the codec in use.  The integral values are maintained an in IANA
      registry and directly map to the media type names.  It is encoded
      as a variable length field, enabling a small number of bits for
      frequent codecs. (tag = 3)

   o  The media source, which is an 8 bit integer specifying the ID of
      the media source for this chunk. (tag=TBD)

   o  The media sink, which is an 8 bit integer specifying the ID of the
      media sync for this chunk.  (tag=TBD)

   o  Optional Reference frame flag.  Indicates other chunks are encoded
      based on this frame.  Typically used with video codecs and only

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      set true on chunks that are part of an I Frame.  Default value is
      false.  (tag=TBD)

   o  Optional Spacial Layer scale: TBD - indicates which spacial layer
      this packet is part of when using a scalable codec.  (tag=TBD)

   o  Optional Temporal Layer scale: TBD - indicates which temporal
      layer this packet is part of when using a scalable codec.

   o  Active Level: Provides indication of audio energy in media chunk
      and can be used by conferring bridges to do speaker selection.

   o  Optional extension parameters in the envelope

   Each media chunk has the following properties in the package part:

   o  Media.  Then output from the codec (tag=4)

   o  optional extension parameters in the package

   The chunk is encoded by passing the envelope as authenticated data,
   and the package to an AEAD cipher.  The chunk is created by taking
   the truncated auth tags of the cipher followed by output of the AEAD
   cipher.  The keys for the cipher to use, the keys for the cipher, and
   the authentication tag truncation are set by mechanisms outside this
   section.  It not specified, they default to the NULL cipher with no

   TODO - need to describe more cipher and keys for this

9.11.2.  The Control Chunk

   The control chunk also has an envelope and payload, and is structured
   identically to media chunks.  Its envelope has the following

   o  Media or Control: a boolean which indicates whether this is a
      media or control packet.  The remaining parameters apply when it
      is control.

   o  Control Type: a 16 bit integer which indicates the type.  Only one
      type is defined by this specification - acknowledgement.

   The payload then depends on the control type.  For an
   acknowledgement, the payload contains the direction, media sourceID

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   and media sink ID of the chunk being acknowledged.  The final
   component needed to identify a chunk - the callID - is inferred from
   that of the media byway in which the acknowledgement is sent.

9.11.3.  Client Media handling

   If the HTTP connection supports webtransport, media chunks MUST be
   sent (and received) as datagrams over the webtrasport session.

   If not, to send media, the client MUST create a new HTTP PUT request
   to the /media resource on call URI for each media chunk to send.  The
   body of the request MUST include the media chunk.  If the client has
   received any chunks from the peer for this call since the last time
   it has sent a media packet for this call, it MUST include an
   acknowledgement control packet in the body as well, one for each
   unacknowledged media chunk.  Once the one media chunk and zero or
   more acknowledgement packets have been sent, the client MUST end the

   To receive media, the client will have a large number of reverse
   media byways open (as GET requests to the /media resource for the
   call).  If a response is received, the client extracts the contents,
   which will always be one media chunk and zero or more control chunks.

   The media sent by the client MUST match the client directive received
   from the server.

9.11.4.  Server Media Handling

   If the HTTP connection supports webtransport, media chunks MUST be
   sent (and received) as datagrams over the webtrasport session.

   If not, to send media, the server MUST select an open GET request to
   /media.  The server MUST select the most recent, in order to enable
   migration of media streams from one client to another.  The server
   MUST place its media chunk in the body of the response.  If the
   server has received any media chunks from the peer for this call
   since the last time it has sent a media packet for this call, it MUST
   include an acknowledgement control packet in the body as well, one
   for each unacknowledged media chunk.  Once the one media chunk and
   zero or more acknowledgement packets have been sent, the server MUST
   end the transaction.  If the server is unable to send media for the
   call because there are no open requests, it MUST generate the "media-
   panic" event on the event stream towards the client, which indicates
   this problem.  The server MUST drop the media chunk.  The call will
   eventually trigger migration as the client will timeout on

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   The server will receive one media and zero or more control chunks as
   the body of the PUT requests.  Note that these may actually originate
   from different clients during migration events; this is irrelevant to
   server processing of media.

9.12.  Connection and Byway Lifecycle Management

   The state of the connection, the QUIC streams, and byways, is
   separate from the state of the call.  The client MAY terminate an
   HTTP connection or byway at any time, and re-establish it.
   Similarly, the server or client may end the a byway at any time,
   neither affects call state directly.

   If a byway ends or the connection breaks or is migrated, the client
   MUST re-initiate the byways immediately, or risk loss of media and
   signalling events.  However, to deal with the fact that re-
   establishment takes time, both client and server MUST buffer their
   signalling and media streams for at least 5 seconds, and then once
   the connections and byways are re-established, it sends all buffered
   data immediately.

   A server MUST maintain a timer, with a value equal to 30 seconds, for
   which it will hold the call in its current state without any active
   signalling byway.  If the server does not receive a signalling byway
   before the expiration of this timer, it MUST consider the call as
   ended.  Once the call has ended, the call resource SHOULD be

   If the server receives a signalling or media byway for a call that is
   TERMINATED, it MUST reject the transaction with an 404 response code,
   since the resource no longer exists.

   Note that it is the sole responsibility of the client to make sure
   byways are re-established if they fail unexpectedly.

9.13.  Graceful Call Migration

   A server MAY initiate a call migration at any time for a specific
   call.  To do so, it MUST send a migration event to the client over
   the signaling byway.  The client MUST honor this request for
   migration.  The client MAY choose to migrate the call to a different
   server at any time.

   To perform the migration, the client MUST end all outstanding HTTP
   transactions for the call (signaling and media byways).  If the
   server provided a URI in the migrate event, the client MUST treat
   this as a new URI for the call, replacing the prior one.  Then, It
   MUST initiate a new signaling byway towards the call URI.  This may

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   require the client to open a new HTTP connection, if it does not
   already have one open to the target.  The client MUST NOT include a
   session cookie in the PUT to /events.  This is essential to cause the
   request to route to a different instance behind the load balancer,
   which will typically be configured with sticky routing.  The server
   MAY respond to this request with a new session cookie.  The client
   MUST NOT initiate forward or reverse media byways until the client
   receives response headers to the signaling byway creation.  This is
   to enable session cookies to be included in these requests for sticky

   The client SHOULD cache all media that the client wishes to transmit,
   but was unable to do so during the migration.  The media MUST then be
   sent in a burst once the media byways are re-established.  This
   ensures there is no packet loss (though there will be jitter) during
   the migration period.

9.14.  Ungraceful Call Migration

   If the origin server or HTTP load balancer fails unexpectedly, this
   is handled by the client.  If the client (1) receives a reset on the
   signaling byway, or (2) receives no media packets for 5s, or (3)
   receives no acknowledgements for media packets it has sent for 1s, or
   (4) receives a ungraceful closure of the HTTP connection, it MUST
   initiate a migration as defined in the prior section.


   TODO - add in RAML

11.  IANA Considerations

11.1.  Registration of Well-Known Web Services

   This specification defines two new well-known web services as defined
   in [I-D.nottingham-rfc5785bis] - ript and ript-oauth.

11.2.  RIPT

   URI suffix: ript

   Change controller: IETF

   Specification document(s): This specification.

   Status: Permanent.

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11.3.  RIPT-Oauth

   URI suffix: ript-oauth

   Change controller: IETF

   Specification document(s): This specification.

   Status: Permanent.

12.  IANA Considerations

13.  Security Considerations

14.  Acknowledgements

   Thanks you for review and edits to: Giacomo Vacca.  Thank you to Mo
   Zanaty for greatly simplifying the advertisement proposal for video.

15.  References

15.1.  Normative References

              Bishop, M., "Hypertext Transfer Protocol Version 3
              (HTTP/3)", draft-ietf-quic-http-25 (work in progress),
              January 2020.

              Nottingham, M., "Well-Known Uniform Resource Identifiers
              (URIs)", draft-nottingham-rfc5785bis-11 (work in
              progress), April 2019.

              Vasiliev, V., "WebTransport over HTTP/3", draft-vvv-
              webtransport-http3-01 (work in progress), November 2019.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC2833]  Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF
              Digits, Telephony Tones and Telephony Signals", RFC 2833,
              DOI 10.17487/RFC2833, May 2000,

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   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
              September 2002, <>.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007,

   [RFC8224]  Peterson, J., Jennings, C., Rescorla, E., and C. Wendt,
              "Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 8224,
              DOI 10.17487/RFC8224, February 2018,

   [RFC8225]  Wendt, C. and J. Peterson, "PASSporT: Personal Assertion
              Token", RFC 8225, DOI 10.17487/RFC8225, February 2018,

   [RFC8226]  Peterson, J. and S. Turner, "Secure Telephone Identity
              Credentials: Certificates", RFC 8226,
              DOI 10.17487/RFC8226, February 2018,

15.2.  Informative References

              Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
              and Secure Transport", draft-ietf-quic-transport-25 (work
              in progress), January 2020.

   [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,

Authors' Addresses

   Jonathan Rosenberg


   Cullen Jennings
   Cisco Systems


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   Anthony Minessale


   Jason Livingood


   Justin Uberti


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