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Versions: 00                                                            
Network Working Group                                            C. Bran
Internet-Draft                                               C. Jennings
Intended status: Standards Track                                   Cisco
Expires: December 7, 2011                                   June 5, 2011

                    RTC-Web Communications Protocols


   The real time communications web (RTC-Web) will enable applications
   such as web browsers to natively support real time interactive voice
   and video.  This document outlines the communication protocols for
   realizing RTC-Web functionality within applications such as web
   browsers.  In addition to communications protocols, this document
   proposes a set of application programming interface requirements for
   controlling the protocol stack.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.  This document may not be modified,
   and derivative works of it may not be created, and it may not be
   published except as an Internet-Draft.

   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 http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on December 7, 2011.

Copyright Notice

   Copyright (c) 2011 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
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect

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   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Protocol Requirements  . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Connection Management Requirements . . . . . . . . . . . .  4
     3.2.  Signaling Protocol Requirements  . . . . . . . . . . . . .  5
       3.2.1.  Client Application SIP Requirements  . . . . . . . . .  5
       3.2.2.  Client Application Optional SIP Support  . . . . . . .  5
       3.2.3.  Required SIP Methods . . . . . . . . . . . . . . . . .  6
       3.2.4.  Multipart SIP Message Requirements . . . . . . . . . .  6
       3.2.5.  SIP Identity Requirements  . . . . . . . . . . . . . .  6
       3.2.6.  SIP Network Address Traversal  . . . . . . . . . . . .  7
     3.3.  Codec Requirements . . . . . . . . . . . . . . . . . . . .  7
       3.3.1.  Audio Codec Requirements . . . . . . . . . . . . . . .  7
       3.3.2.  Video Codec Requirements . . . . . . . . . . . . . . .  8
     3.4.  Real-Time Media Transport Requirements . . . . . . . . . .  8
       3.4.1.  RTP Profile  . . . . . . . . . . . . . . . . . . . . .  9
       3.4.2.  RTP Optimizations  . . . . . . . . . . . . . . . . . .  9
       3.4.3.  RTP Extensions . . . . . . . . . . . . . . . . . . . . 11
       3.4.4.  RTP Transport Robustness . . . . . . . . . . . . . . . 12
       3.4.5.  RTP Rate Control . . . . . . . . . . . . . . . . . . . 13
     3.5.  Non-Media Data Transport Requirements  . . . . . . . . . . 13
       3.5.1.  Non-Media Data In RTP  . . . . . . . . . . . . . . . . 14
       3.5.2.  Transporting Non-Media Data  . . . . . . . . . . . . . 14
   4.  API Requirements . . . . . . . . . . . . . . . . . . . . . . . 16
   5.  Legacy VoIP Interoperability . . . . . . . . . . . . . . . . . 16
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 17
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   8.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
   9.  Normative References . . . . . . . . . . . . . . . . . . . . . 17
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20

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1.  Introduction

   The Internet was, from very early in its lifetime, considered a
   possible vehicle for the deployment of real-time, interactive
   applications - with the most easily imaginable being audio
   conversations (aka "Internet telephony") and videoconferencing.

   The first attempts to build this were dependent on special networks,
   special hardware and custom-built software, often at very high prices
   or at low quality, placing great demands on the infrastructure.

   As the available bandwidth has increased, and as processors and other
   hardware has become ever faster, the barriers to participation have
   decreased, and it is possible to deliver a satisfactory experience on
   commonly available computing hardware.

   Still, there are a number of barriers to the ability to communicate
   universally - one of these is that there are, as of yet, no single
   set of communication protocols that all agree should be made
   available for communication; another is the sheer lack of universal
   identification systems (such as is served by telephone numbers or
   email addresses in other communications systems).

   Development of "The Universal Solution" has proved hard, however, for
   all the usual reasons.  This memo aims to take a more building-block-
   oriented approach, and try to find consensus on a set of substrate
   components that we think will be useful in any real-time
   communications systems.

   The last few years have also seen a new platform rise for deployment
   of services: The browser-embedded application, or "Web application".
   It turns out that as long as the browser platform has the necessary
   interfaces, it is possible to deliver almost any kind of service on

   Traditionally, these interfaces have been delivered by plugins, which
   had to be downloaded and installed separately from the browser; in
   the development of HTML5, much promise is seen by the possibility of
   making those interfaces available in a standardized way within the

   Other efforts, for instance the W3C Web Applications and Device API
   working groups, focus on making standardized APIs and interfaces
   available, within or alongside the HTML5 effort, for those functions;
   this memo concentrates on specifying the protocols and subprotocols
   that are needed to specify the interactions that happen across the

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2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Protocol Requirements

   The section defines the set of protocols and selected subset profiles
   of these protocols that RTC-WEB client applications will need to
   implement.  This set of protocols forms the requirements for the
   controlling APIs in [Section 4].  At a high level this section is
   split into five subsections that address requirements for RTC-WEB
   client application: connection management, signaling protocols, codec
   requirements, transports for real time media such as audio and video
   and transports for non media data .

3.1.  Connection Management Requirements

   It is quite probable that many RTC-WEB client applications, such as
   web browsers will be deployed behind a NAT.  To set up secure data
   plane sessions, all RTC-WEB client application implementations will
   use ICE [RFC5245] or ICE-Lite Section 2.7 of [RFC5245].  ICE is
   leveraged here to address the security concerns discussed in
   [section] Section 7.

   There are two deployment scenarios for RTC-WEB client applications.
   The first scenario is when applications are deployed behind NAT and
   have to worry about NAT traversal.  The second scenario is when the
   application is not behind a NAT, such as an RTC-WEB application that
   is always connected to the public Internet.  As stated in section 2.7
   of [RFC5245], ICE requires that both endpoints to support it in order
   for ICE to be used on a call.

   With regards to RTC-WEB client applications, all applications that
   are deployed behind a NAT or do not have a public IP address are
   REQUIRED to support ICE [RFC5245], applications that are not behind a
   NAT and have a public IP address are REQUIRED to support ICE-Lite and
   MAY fully support ICE.  RTC-WEB client applications that fully
   support ICE are REQUIRED to support AGGRESSIVE NOMINATION, and MAY

   Implicit to supporting ICE, all RTC-WEB client applications are
   REQUIRED to implement Simple Traversal of User Datagram Protocol
   (UDP) Through Network Address Translators (NATs) (STUN) [RFC3489] and
   Traversal Using Relays around NAT (TURN) [RFC5766].

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   [Open Issue: there is a strong interest to define a TURN-like
   protocol that looks like HTTP to intermediaries, so that media can be
   tunneled over HTTP.  Should this be done?]

3.2.  Signaling Protocol Requirements

   This section covers the signaling protocol to be used by RTC-WEB
   applications.  To ensure interoperability not just between RTC-WEB
   applications, but with legacy IPPBX phone systems as well, a small
   subset of SIP will be REQUIRED for all RTC-WEB client application
   implementations.  In addition to the subset of SIP specification
   [RFC3261], RTC-WEB client application implementations will be
   REQUIRED to support DNS resolutions as specified in [RFC3263] and the
   offer/answer model with SDP as specified in [RFC3264].

3.2.1.  Client Application SIP Requirements

   This section focuses on the subset of SIP functionality that will
   exist within all RTC-WEB client applications.  The following User
   Agent Client (UAC) subset of the SIP specification [RFC3261] is

   o  General User Agent Behavior - [Section 8]

   o  Registration - [Section 10]

   o  Client Transaction - [Section 17.1]

   o  Canceling a Request - [Section 9.1]

   o  Terminating a Session - [Section 15.1], [Section 15.1.1]

   o  3.XX Redirect Responses - [Section]

   o  TLS - [Section 26.3.1]

   o  Outbound Proxy

3.2.2.  Client Application Optional SIP Support

   In the SIP specification [RFC3261], the SIP features listed below are
   required for all UAC implementations.  RTC-WEB client applications
   are not a fully featured SIP UAC and will only be implementing a
   subset of the SIP specification.  Thusly, unlike SIP UACs, the
   following list of SIP features is to be considered OPTIONAL for RTC-
   WEB client application implementations.

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   o  INVITEs without an offer

   o  re-INVITEs - [Section 14.1]

   o  forking - [Section 19.3]

   o  S/MIME - [Section 23]

   o  SIPS URI Scheme - [Section 26.2.2]

3.2.3.  Required SIP Methods

   This section outlines the REQUIRED SIP methods for all RTC-WEB client

   o  INVITE - [RFC3261] [Section 13]

   o  REGISTER - [RFC3261] [Section 10]

   o  ACK - [RFC3261] [Section 13.2]

   o  CANCEL - [RFC3261] [Section 13.2]

   o  BYE - [RFC3261] [Section 13.2]

   o  UPDATE - [RFC3311]

3.2.4.  Multipart SIP Message Requirements

   For handling SIP messages RTC-WEB client applications are required to
   implement the multipart MIME handling scheme as specified in

3.2.5.  SIP Identity Requirements

   Identity, for the purposes of this section, is defined as a SIP URI.
   There are two areas concerning SIP identity this specification will

   The first area covers validation of the message originator.  To
   securely validate a the identity of a SIP message originator, all
   RTC-WEB client applications are REQUIRED to implement the mechanism
   specified in [RFC4474].

   To support cases were the identify of a caller/callee may change,
   such as when a call is parked and transferred from the original
   callee to another party, all RTC-WEB client applications are REQUIRED
   to implement the identity mechanism specified in [RFC4916].

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   [RFC3261]implicitly REQUIRES the implementation of the UPDATE method
   as specified in [RFC3311]

3.2.6.  SIP Network Address Traversal

   RTC-WEB client applications MUST support Network Address Translator
   (NAT) traversal.  This section will address SIP-related areas to
   support NAT traversal.

   As called for in [3.1] RTC-WEB client applications will implement
   STUN.  To support client-managed connections, STUN-based keep-alives
   as specified in [RFC5626] are REQUIRED.

   When SIP is used with UDP, responses to requests are returned to the
   source address the request came from, and to the port written into
   the topmost Via header field value of the request.  This behavior is
   not desirable when the RTC-WEB client application is behind a Network
   Address Translator (NAT).  To address UDP traversal problem the
   "rport" extension as specified in [RFC3581] is REQUIRED.

3.3.  Codec Requirements

   This section covers the audio and video codec requirements for RTC-
   WEB client applications.  To ensure a baseline level of
   interoperability between RTC-Web applications, a minimum set of
   required codes is specified below.  While this section specifies the
   codecs that will be supported by all RTC-Web application
   implementations, it leaves the question of supporting additional
   codecs to the will of the implementer.

3.3.1.  Audio Codec Requirements

   RTC-WEB applications are REQUIRED to implement the following audio

   o  PCMA/PCMU - see section 4.5.14 of [RFC3551]

   o  Telephone-event - [RFC4734]

   o  Opus [draft-ietf-codec-opus]

   Implementations of the PCMU and PMCA codecs are REQUIRED to support 1
   channel with a rate of 8000 and a ptime of 20.

   The following codecs are OPTIONAL for RTC-WEB application

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   o  G729

   o  G722

   o  G722.1

   o  G723

   o  AMR

   o  AMR-WB

   o  iLBC

   o  L16

   [Open Issue: minimum profile and identifying any additional mandatory
   to implement audio codecs.]

3.3.2.  Video Codec Requirements

   RTC-WEB applications are REQUIRED to implement the following video

   o  VP8 [I-D.westin-payload-vp8]

   The following codecs are OPTIONAL for RTC-WEB application

   o  H.263

   o  H.264-AVC

   o  H.264-SVC

   [Open Issue: For the mandatory to implement video codec(s) what is
   the minimum profile?]

3.4.  Real-Time Media Transport Requirements

   This section defines the real-time media transport requirements for
   RTC-Web client application implementation.  This section breaks down
   the RTC-WEB RTP requirements into several sections.  The sections
   cover the RTP requirements for: profile, optimizations, extensions,
   transport robustness and rate control.

   [OPEN ISSUE: identify missing requirements]

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3.4.1.  RTP Profile

   RTC-Web applications to will need to provide a secure, interoperable,
   bandwidth friendly, media transport profile.  The Secure Audio-visual
   Profile Feedback (SAVPF) as defined in [RFC5124] will meet the needs
   of RTC-Web applications by providing media encryption,
   interoperability and a flexible, bandwidth conscious RTCP packet
   transmission model.  All RTC-Web applications are REQUIRED to
   implement SAVPF.  Requiring the implementation of SAVPF also means
   that RTC-Web applications MUST implicitly support Audio-visual
   Profile Feedback (AVPF) [RFC4585], Audio-visual Profile (AVP)
   [RFC3551] and Secure Audio-visual Profile (SAVP) [RFC3711].  Profile Encryption Mechanism

   SAVPF supports SRTP by providing media encryption, integrity
   protection, replay protection and a limited form of source
   authentication.  Though the SAVPF profile does support secure media
   transport, it does not specify an encryption keying mechanism.  To
   support keying for SRTP, WEB-RTC application implementors are
   REQUIRED to implement DTLS-SRPT [RFC5764].

3.4.2.  RTP Optimizations

   This section describes the optimization requirements for RTP within
   RTC-Web applications.  RTP and RTCP Multiplexing

   Historically, RTP and RTCP have been run on separate UDP ports.  With
   the increased use of Network Address Port Translation (NAPT) so have
   the problems increased for maintaining multiple, costly NAT bindings
   for each UDP port.  This dual UDP port paradigm also complicates
   firewall administration, since multiple ports must be opened to allow
   for RTP traffic.  To reduce these costs and session setup times,
   support for multiplexing RTP data packets and RTCP control packets on
   a single port [RFC5761] is REQUIRED.

   Note that the use of RTP and RTCP multiplexed on a single port
   ensures that there is occasional traffic sent on that port, even if
   there is no active media traffic.  This may be useful to keep-alive
   NAT bindings.  Reduced-Size RTCP

   RTCP packets are usually sent as compound RTCP packets and [RFC3550]
   demands that the RTCP compound packets always start with a Sender
   Report (SR) or Receiver Report (RR) packet.  The SR and RR packets

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   provide reception quality statistics and increase the mean RTCP
   packet size.  Because the mean compound RTCP packet size is larger,
   the frequency at which RTCP packets can be sent within the RTCP
   bandwidth share decreases.  The decreased transmission frequency
   creates a performance bottleneck that is especially noticeable when
   using frequent feedback messages.

   As mentioned in section [Add ref] RTC-Web applications will be
   required to implement SAVPF, which implicitly requires feedback.
   [RFC5506] specifies how to reduce the mean RTCP message and allow for
   more frequent feedback.  Frequent feedback, in turn, is essential to
   make real-time application quickly aware of changing network
   conditions and allow them to adapt their transmission and encoding
   behavior.  Support for [RFC5506] is REQUIRED  Symmetric RTP/RTCP

   RTP entities choose the RTP and RTCP transport addresses (IP
   addresses and port numbers), to bind to and receive packets on.
   However when sending RTP and RTCP packets, senders may use an IP
   address or port number that is different than the one specified for
   receiving packets.  Using different transport addresses is
   problematic with regards to NAT traversal.  The NAT traversal problem
   can be alleviated using symmetric RTP/RTCP [RFC4961].  Symmetric RTP/
   RTCP requires that the transport addresses for sending and receiving
   RTP/RTCP packets are identical.  All RTC-WEB client applications are
   REQUIRED to implement Symmetric RTP/RTCP [RFC4961].  CNAME Generation

   The RTCP Canonical Name (CNAME) provides a persistent transport-level
   identifier for an RTP endpoint.  While the Synchronization Source
   (SSRC) identifier for an RTP endpoint may change if a collision is
   detected, or when the RTP application is restarted, it's RTCP CNAME
   is meant to stay unchanged, so that RTP endpoints can be uniquely
   identified and associated with their RTP media streams.  For proper
   functionality, RTCP CNAMEs should be unique within the participants
   of an RTP session.

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME.  These guidelines are not sufficient in the
   presence of NAT devices or with regards to addressing privacy
   concerns resulting from the long-term, persistent identifiers.

   To address the shortcomings of CNAME selection in[RFC3550], it is
   RECOMMENDED that RTP CNAME generation follows the approach specified
   in section 5 of [RFC6222].

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   For RTC-WEB client applications, such as a web browser, it may not be
   possible to retrieve the EUI-64 identifier or the host system's MAC
   address which is needed to fulfill the CNAME generation procedure
   outlined in section 5 of [RFC6222].  As an alternative to the EUI-64/
   MAC address, RTC-WEB client applications MAY generate and use a
   random number for the unique CNAME generation procedure.

3.4.3.  RTP Extensions

   .This section describes the RTP extensions that could be very useful
   within the RTC-WEB context.  Conferencing Extensions

   RTC-Web applications will support conferencing capabilities.  While
   this document remains silent regarding what conferencing topology
   should be supported for RTC-Web applications, the following section
   will provide guidance around RTP extensions to support centralized

   For more information on RTP conferencing topologies please refer to
   [RFC5117]  FIR RTCP Feedback Message

   The Full Intra Request (FIR) command and message are defined in
   sections 3.5.1 and 4.3.1 of [RFC5104].  FIR messages will request
   that the currently distributed session participants send new intra
   coded pictures to the mixer.  FIR is used when switching between
   sources to ensure that the receivers can decode the video or other
   predicted media encoding with long prediction chains.  It is
   RECOMMENDED that the FIR message is supported.  PLI RTCP Feedback Message

   The Picture Loss Indicator (PLI) is defined in Section 6.3.1 of
   [RFC4585].  PLI messages tell the encoder that a receiver has lost
   the decoder context and would like it repaired.  It is RECOMMENDED
   that the PLI message is supported.  TMMBR RTCP Feedback Message

   The Temporary Maximum Media Stream Bit Rate Request (TMMBR, "timber")
   message is defined in sections 3.5.4 and 4.2.1 of [RFC5104].  A
   receiver, translator, or mixer uses the TMMBR to request a sender to
   limit the maximum bit rate for a media stream to, or below, the
   provided value.  An example of using TMMBR would be for an RTP mixer
   to constrain the media sender's bit rate to fit within the lower bit

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   rate range of other session participants.  It is RECOMMENDED that the
   TMMBR message be supported.  Header Extensions

   This section describes the requirements for RTC-WEB RTP header
   extensions.  For all RTC-WEB RTP header extensions it is REQUIRED
   that they are formatted and signaled according to the general
   mechanism defined in [RFC5285].

   [Open Issue: should any of the following headers be added to the

   o  Transmission time offsets[RFC5450]

   o  Associating time-codes with RTP streams [RFC5484] [Remove?]

   Open Issue: There is also ongoing work to define RTP header
   extensions for providing audio levels:

   o  From the media sender to the mixer [I-D.ietf-avtext-client-to-

   o  From a mixer to a receiver [I-D.ietf-avtext-mixer-to-client-audio-

   Which, if any of the above should be required? optional?

   ]  Rapid Synchronization

   Basic RTP session synchronization as described in [RFC3550] can be
   slow.  To improve synchronization performance and maintain relative
   backwards compatibility it is RECOMMENDED that the rapid RTP
   synchronization extensions described in [RFC6051] be implemented.

3.4.4.  RTP Transport Robustness

   This section identifies tools that can be used to add robustness to
   the RTP flows.  Adding robustness to the RTP flow can reduce packet
   loss and thus have a positive impact upon media quality.  RTP Retransmission

   The retransmission scheme in RTP allows for flexibility of
   retransmissions.  From the receiving side, only selected missing
   packets can be requested.  From the sending side, packets can be

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   prioritized based upon the senders knowledge of the receiver's
   missing packets.  Support for RTP retransmission as defined by
   [RFC4588] is RECOMMENDED.

   [Open Issue: is [RFC4588] the way we want to tackle this issue?]  Forward Error Correction

   [Open issue - should there be a FEC scheme recommendation?]  Multicast

   RTC-WEB client applications support for multicast RTP is NOT

3.4.5.  RTP Rate Control

   [OPEN ISSUE - There are currently no available, standardized RTP rate
   control mechanism that uses media adaptation.  Having a mechanism in
   place will be REQUIRED for RTC-WEB applications and which means there
   is a need for the IETF to produce this specification.

   A potential starting point for defining a solution is "RTP with TCP
   Friendly Rate Control" [rtp-tfrc].]

3.5.  Non-Media Data Transport Requirements

   The RTC-WEB will enable for rich voice and video communications from
   client applications, such as a web browser.  One of the natural
   extensions of the RTC-WEB and the work emerging from the HTML5
   community is video games.  Video games have a similar stringent real-
   time requirement for exchanging non-media data types such as a
   player's screen position.

   The question of how best to handle non-media data types has been
   raised.  There have been proposals to address this problem.  Common
   to all proposals is how the data transport session is set up, using
   ICE [RFC5245] in a similar manner to that of RTP [RFC3550].  The
   proposals vary from once the session is set up; one proposal is just
   to use a thin shim on top of UDP or DTLS to de-multiplex the packets
   from other packets such as RTP on the same connection.  Another
   proposal is DTLS over DCCP over UDP with some appropriate congestion
   control scheme chosen for DCCP.  Lastly there has been a proposal to
   define a data codec to carry the data in RTP.

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3.5.1.  Non-Media Data In RTP

   This section will answer the question regarding the addition of non-
   media data types into an RTC-WEB client application initiated RTP

   RTP by design adheres to the application level framing architectural
   principle.  This principle requires that RTP payload formats be
   specified.  By requiring specific payload formats RTP provides a
   mechanism to optimize the transmission of encoded media.  Other than
   this optimization there is no congestion control mechanisms for RTP.

   This principle also implies that if a payload format cannot be
   specified, as the case is with generic data, it breaks one of the
   fundamental architectural principles of RTP and makes optimization
   impossible.  Given that the ability to optimize the transmission of
   non-media data types is lost and there are no capabilities for
   congestion control within RTP, it follows that there is no benefit to
   using RTP instead of a more general data transport such as UDP.
   Until non-media data payload formats are created, the use of RTP as a
   non-media data transport SHALL NOT be used in conjunction with any
   RTC-WEB client application implementation.

   [Open issue: There has been mention of actually creating new payload
   formats for non-media data types.  If new payload formats are
   actually created for specific types of non-media data, the
   requirement above would still stand as the application level framing
   principle would be preserved and the new formats would have to adhere
   to the principle.  Any new formats would be specified outside of this
   document but referred to]

3.5.2.  Transporting Non-Media Data

   [OPEN issue: need further discussion around this area]

   There have been some ideas proposed but nothing has emerged as the
   dominant paradigm.  The current thinking is that, for RTC-WEB client
   applications, RTP is not an option for non-media data types that do
   not have a payload format specification.  Without a payload format
   specification a workable solution would resemble something that
   allows datagrams to be transmitted via a secure, congestion-
   controlled, unreliable transport mechanism.  Proposed Solutions for Non-Media Data Transport

   One of the current proposed solutions could meet the requirements for
   a non-media data type transport for RTC-WEB client application is to
   use a DCCP via the following specifications:

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   o  DCCP [RFC4340] using TCP [RFC4341] or TFRC [RFC4342] for
      congestion control

   o  DTLS for DCCP [RFC5238]

   o  DCCP over UDP [draft-ietf-dccp-udpencap]

   The maturity of available implementations of DCCP is of concern along
   with the partiality of this proposed solution.  Another way of
   tackling the problem of non-media data transport is to push the
   requirements into the RTC-WEB client application implementation.

   The following is a proposed set of REQUIRED RTC-WEB client
   application non-media data transport requirements.

   o  Support for a broad congestion control constraints are REQUIRED,
      it is RECOMMENDED that implementors support either TFRC [RFC5348]
      or TFRC-SP [RFC4828].

   o  Support for full congestion control is RECOMMENDED, the specific
      implementation is left to the RTC-WEB client application

   o  Support for DTLS over UDP is REQUIRED

   As an example of how these proposed requirements could be implemented
   within an RTC-WEB client application, lets explore a web browser-
   based implementation.  In this specific implementation, the web
   browser would provide DTLS over UDP and implement a broad congestion
   control solution such as TFRC or TFRC-SP.  This implementation will
   yield a coarse-grained congestion controlled non-media data transport
   solution that is accessible via JavaScript API calls.  These non-
   media data transport capabilities would provide a flexible solution
   for web developers to build a full congestion control solution into
   their WEB-RTC client application.

   [Open Issue: Given that there is no consensus with regards to a
   transport solution, this topic needs further discussion.

   Open Issue: Areas for further discussion:

   o  Unreliable datagram transport - should this be UDP, SCTP or DCCP?

   o  Congestion control mechanisms - DCCP and SCTP something else?

   o  Protection of data confidentiality and integrity - DTLS

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   o  Receiver consent mechanisms for data transmission


4.  API Requirements

   NOT Ready - need to decide on protocols first, API comes after that


   The API needs to allow the DSCP REF for each RTP or media stream to
   be set.

   The API needs to allow the browser app to observer and control the
   SSRC values in the RTP.


   The API needs to support the following OPTIONAL codecs: H263-2000,
   H264, H264-SVC, raw and VP8.

   The API needs to support the following OPTIONAL codecs: G729, G722,
   G7221, G723, AMR, AMR-WB, iLBC, L16 and opus.

5.  Legacy VoIP Interoperability

   There is no way to meet all the security requirements and maintain
   comparability with all legacy VoIP equipment.  This draft tries to
   minimize the impedance mismatch.  The requirements here would allow
   interoperability with legacy VoIP equipment as long as that equipment
   either directly supported, or was fronted by an SBC that supported,
   the following: a CORS [W3C.WD-cors-20090317] extension for SIP, ICE
   or ICE-Lite, the mandatory to implement codecs in [SECTION],
   supported SIP invites containing an offer, and supported DTMF over
   RTP with telephone events.

   Of the items listed above, support for ICE-Lite has historically been
   lacking in VoIP equipment, this is changing and ICE-Lite becoming
   increasingly prevalent, particularly on devices designed to sit on
   the edge of a domain and connect to remote user agents that may be
   behind NATs.  Given the increasing adoption of ICE-Lite, it could be
   conjectured that a substantial fraction of VoIP equipment meets the
   RTC-WEB interoperability list except for the CORS extensions.

   For an edge device that was willing to receive SIP call from others,
   implementing the CORS is pretty trivial.  When the UAS receives a SIP

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   options request with an Origin header, it checks whether the header
   field value is on the white list, and if it is then the UAS copies
   the value to the Access-Control-Allow-Origin header field value in
   the response.  For many situations the white list would be
   everything, while for others it would be just the list of websites
   that are expected to originate calls to this SIP device.

6.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an

7.  Security Considerations

   Because there are a number of security issues, considerations and
   requirements for RTC-WEB client applications there is a draft that
   specifically addresses the RTC-WEB application security
   considerations.  This draft defers it's security considerations and
   requirements to the security considerations for RTC-Web draft

8.  Acknowledgements

   Many thanks to Harald Alvestrand, Magnus Westerlund, Colin Perkins,
   Joerg Ott for a signifcant amount of text and contributed ideas on
   this topic.

9.  Normative References

              Rescorla, E., "Security Considerations for RTC-Web",
              May 2011.

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

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation

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              Protocol (SIP): Locating SIP Servers", RFC 3263,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, October 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3581]  Rosenberg, J. and H. Schulzrinne, "An Extension to the
              Session Initiation Protocol (SIP) for Symmetric Response
              Routing", RFC 3581, August 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340, March 2006.

   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion Control ID 2: TCP-like
              Congestion Control", RFC 4341, March 2006.

   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for
              Datagram Congestion Control Protocol (DCCP) Congestion
              Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
              March 2006.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,

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              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              July 2006.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [RFC4734]  Schulzrinne, H. and T. Taylor, "Definition of Events for
              Modem, Fax, and Text Telephony Signals", RFC 4734,
              December 2006.

   [RFC4828]  Floyd, S. and E. Kohler, "TCP Friendly Rate Control
              (TFRC): The Small-Packet (SP) Variant", RFC 4828,
              April 2007.

   [RFC4916]  Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", RFC 4916, June 2007.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, July 2007.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
              January 2008.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [RFC5238]  Phelan, T., "Datagram Transport Layer Security (DTLS) over
              the Datagram Congestion Control Protocol (DCCP)",
              RFC 5238, May 2008.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 5348, September 2008.

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   [RFC5450]  Singer, D. and H. Desineni, "Transmission Time Offsets in
              RTP Streams", RFC 5450, March 2009.

   [RFC5484]  Singer, D., "Associating Time-Codes with RTP Streams",
              RFC 5484, March 2009.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC5621]  Camarillo, G., "Message Body Handling in the Session
              Initiation Protocol (SIP)", RFC 5621, September 2009.

   [RFC5626]  Jennings, C., Mahy, R., and F. Audet, "Managing Client-
              Initiated Connections in the Session Initiation Protocol
              (SIP)", RFC 5626, October 2009.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, November 2010.

   [RFC6222]  Begen, A., Perkins, C., and D. Wing, "Guidelines for
              Choosing RTP Control Protocol (RTCP) Canonical Names
              (CNAMEs)", RFC 6222, April 2011.

Authors' Addresses

   Cary Bran
   170 West Tasman Drive
   San Jose, CA  95134

   Phone: +1 206 256-3502
   Email: cbran@cisco.com

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   Cullen Jennings
   170 West Tasman Drive
   San Jose, CA  95134

   Phone: +1 408 421-9990
   Email: fluffy@cisco.com

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