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Versions: 00 01                                                         
Network Working Group                                             J. Ott
Internet-Draft                                                       TUM
Intended status: Standards Track                                 R. Even
Expires: January 4, 2018                                          Huawei
                                                              C. Perkins
                                                   University of Glasgow
                                                                V. Singh
                                                            July 3, 2017

                             RTP over QUIC


   QUIC is a UDP-based protocol for congestion controlled reliable data
   transfer, while RTP serves carrying (conversational) real-time media
   over UDP.  This draft discusses design aspects and issues of carrying
   RTP over QUIC.

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 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 January 4, 2018.

Copyright Notice

   Copyright (c) 2017 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
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   (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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  RTP-to-Transport Interface  . . . . . . . . . . . . . . . . .   3
   3.  RTP-to-QUIC Mapping . . . . . . . . . . . . . . . . . . . . .   5
     3.1.  Mapping Semantic Units  . . . . . . . . . . . . . . . . .   5
     3.2.  Encapsulating Media Units . . . . . . . . . . . . . . . .   6
     3.3.  Mapping Media to Streams  . . . . . . . . . . . . . . . .   6
     3.4.  Mapping RTCP packets  . . . . . . . . . . . . . . . . . .   7
     3.5.  Mapping of RTP header extensions  . . . . . . . . . . . .   8
   4.  Design considerations for QUIC  . . . . . . . . . . . . . . .   8
   5.  SDP Extensions for Negotiating RTP-over-QUIC  . . . . . . . .   8
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   8.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .   9
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   9
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .   9
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  10
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  13

1.  Introduction

   The Real-time Transport Protocol (RTP) [RFC3550] is a protocol for
   carrying media with real-time properties.  It is usually mapped to
   UDP, possibly with DTLS [RFC5763] [RFC5764] in-between, as UDP allows
   RTP full control over packet transmission timing and congestion
   control.  A number of media-specific and media-independent error
   control mechanisms have been developed in the AVTCORE and AVTEXT WGs
   to cope with the unreliability of UDP (e.g., [RFC4588]), and several
   congestion control mechanisms are presently being explored in the
   RMCAT WG (e.g., [I-D.ietf-rmcat-scream-cc] [I-D.ietf-rmcat-gcc]
   [I-D.ietf-rmcat-nada] [I-D.ietf-rmcat-coupled-cc]
   [I-D.singh-rmcat-adaptive-fec]), in addition to the basic circuit
   breaker mechanism [RFC8083]).  RTP could also run over TCP or DCCP,
   but experiments have shown that the operational range in terms of
   underlying network conditions is fairly limited [Delay-TCP].

   How to use of RTP is usually agreed upon between two endpoints using
   a signaling channel (e.g., SIP [RFC3261]) or WebRTC
   [I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-rtp-usage], both with the
   offer/answer exchange [RFC3264] using the Session Description
   Protocol (SDP) [RFC4566].  RTP can run on top connectionless as well

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   as connection-oriented transport protocols.  The signaling channel is
   also exploited to support NAT traversal RTP using ICE [RFC5245].

   The QUIC transport protocol [I-D.ietf-quic-transport]
   [I-D.ietf-quic-tls] [I-D.ietf-quic-recovery]
   [I-D.ietf-quic-manageability] [I-D.ietf-quic-applicability]
   [I-D.ietf-quic-http] is being developed as a secure, reliable,
   congestion controlled UDP-based transport protocol with web
   applications as the primary target.  In particular, QUIC allows for
   low latency establishment of secure connections and supports
   extensive multiplexing of many independent streams within a single
   connection (over a single UDP port), making it attractive for
   bundling of multiple media streams currently specified in SDP using

   The document discusses the possible use of RTP over QUIC with three
   main purposes:

   o  Understanding and defining a sensible mapping of RTP sessions onto
      one (or more) QUIC connections (section 3);
   o  Deriving a wishlist for QUIC functionality to be fed into the QUIC
      WG (section 4); and
   o  Defining a profile of the QUIC protocol with the necessary
      signaling extensions to enable RTP over QUIC (section 5).

   Editor's note: Section 4 is intended to document requirements for now
   and may disappear later if those are met or formally folded into a
   separate document.  Also sections 3 and 5 may ultimately become
   separate Internet drafts for considerations by different working
   groups (e.g., AVTCORE and MMUSIC).

2.  RTP-to-Transport Interface

   The Real-time Transport Protocol defines the notion of RTP sessions
   to describe an elementary communication relationship between two or
   more parties.  An RTP session comprises a uni-, bi-, or
   multidirectional flow of RTP packets carrying media as well as flows
   of RTCP packets providing feed forward from RTP senders to receivers
   and feedback from RTP receivers to senders.

   Each media source is identified by a 32-bit Synchronization Source
   (SSRC) identifier, unique within an RTP session.  An RTP session
   comprise the set of media sources that have the same view of the SSRC
   space.  A single endpoint may use multiple SSRC identifiers (e.g.,
   one for audio and one for video).  Multiple media streams of a single
   endpoint are tied together by means of a common Canonical Name
   (CNAME) carried as part of the RTCP Source Description (SDES)

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   packets.  This allows receivers to, e.g., determine which media
   streams to synchronize.

   Originally, in an RTP session the RTP and RTCP streams each used
   different port numbers, so that a single RTP session would use two
   port numbers (historically, when used with multicast conferencing,
   these were adjacent port numbers, RTP on the even and RTCP on the
   next higher odd port number).  However, the use of unicast RTP has,
   (not just) due to the presence of NATs, motivated the multiplexing of
   both RTP and RTCP on a single port number [RFC5761].  The payload
   structure and number spaces used for RTP and RTCP packets were
   designed to support this easily.

   The bundle framework [I-D.ietf-mmusic-sdp-bundle-negotiation] allows
   multiplexing of multiple RTP streams on a single address:port
   combination.  All the RTP streams in a bundled group are part of a
   single RTP session sharing a single SSRC number space [RFC3550].

   These two efforts also reduce the number of ICE candidates to be
   validated as part of a multimedia call or conference setup procedure.
   They are particularly required in conjunction with WebRTC to reduce
   the signaling and resource requirements, which would affect NATs as
   well as STUN and TURN servers.  We note, however, that ICE is not
   currently usable with QUIC, since QUIC and STUN packets are not
   readily distinguished on a single UDP port, due to poor choice of
   packet formats.

   WebRTC deserves particular consideration because its potential close
   relationship to QUIC: WebRTC uses HTTP/1.1 (possibly using
   WebSockets), or HTTP/2 to connect to web servers, and thus will
   likely use QUIC in the future as a signaling transport.  Moreover,
   WebRTC supports peer-to-peer data channels, which currently target
   using SCTP over UDP over DTLS: SCTP for stream multiplexing within a
   connection and UDP for better NAT traversal properties.  Since QUIC
   would seem to support these two functions, it could be a natural
   choice to be used for the data channel as well - although this would
   require changes to the QUIC packet formats to allow demultiplexing
   with STUN for NAT traversal.

   For the actual media transmission, RTP use codec-specific payload
   formats that define how a piece of encoded media is broken down into
   data units that can fit into an MTU-sized packet for transmission.
   One important goal of RTP payload format design is allowing decoding
   packets as much as possible independent of each other as some may be
   lost due to the best-effort nature of the underlying UDP [RFC2736].
   This implies, on the one hand, that RTP senders have to perform
   codec-level fragmentation in a semantically meaningful manner and, on
   the other hand, that are in control of packet boundaries and

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   transmission scheduling and timing as well as retransmission

   On the receiving side, RTP expects a detailed understanding of packet
   reception timing, possible reordering, and losses, as this
   information is used for the feedback statistics.

3.  RTP-to-QUIC Mapping

   This section address the necessary considerations to realize _one_
   possible way of carrying RTP-over-QUIC.

   Editor's note: At this point, this section is intended to explore the
   design space and briefly describe a number of different options
   without making specific recommendations about which option(s) to
   choose.  Future revisions of this document move towards taking
   concrete decisions.

3.1.  Mapping Semantic Units

   RTP payload formats define a mapping of media data units (e.g., video
   or audio frames, audio samples, etc.) to packets.  Assuming that we
   will preserve the structure of RTP header, optional header extension,
   and payload, there are two obvious options:

      Preserve the previous RTP assumptions about semantic fragmentation
      at MTU size boundaries; i.e., use the same packetization mechanism
      as before, just then drop the resulting RTP packet into a QUIC
      payload.  Note that the MTU size may be smaller since QUIC packet
      headers are larger than plain UDP headers.
      Operate solely on semantic units such as video frames, and map
      each semantic unit to a QUIC payload.  This approach leaves the
      final packetization decision to QUIC.  In this case, our "MTU
      size" would not be defined by the IP layer but by QUIC.  It is
      possible in this case for video frame composed of multiple RTP
      packets to use one RTP header for the whole video frame; no need
      to break the video frame to multiple RTP packet, put all payload
      as one RTP packet whose size may be bigger than MTU and send it as
      QUIC payload.

   If we assume that semantic units are to be received and processed
   atomically for best performance results, then option 2) would be
   preferred.  If we consider that subunits are meaningful (e.g., slices
   in case of video), then option 1) may be preferred.  In any case,
   however, it would be up to the payload definition to determine what a
   semantic unit.

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3.2.  Encapsulating Media Units

   QUIC streams do not preserve packet boundaries but rather offer the
   same abstraction as TCP does.  Therefore, if multiple identifiable
   media units are to be transmitted on the the same stream, the
   encapsulation mechanisms MUST provide boundaries for media data
   units, e.g., similar to the approach chosen for carrying RTP in TCP.

   The exception would be if only a single frame is ever transmitted
   across a single stream (see option 3 in section 3.3) so that stream
   termination signifies the end of the respective packet.

3.3.  Mapping Media to Streams

   There are three basic distinct options for mapping media to streams:

      Map an RTP session to a QUIC stream.  In this case, all media
      packets of the RTP session would be carried within a single QUIC
      Map an RTP stream to a QUIC stream.  In case, as presently
      discussed in the QUIC WG, the QUIC stream would be unidirectional
      and we will have one QUIC stream per transmission direction.

   Note that both options would map, e.g., FEC or retransmission
   sessions to different QUIC streams.  Note also that both 1. and 2.
   implicitly create the problem of head-of-line blocking since QUIC
   streams are reliable and order preserving.  This would thus not serve
   the real-time nature of RTP packets well.

      Map each independently decodable groups of frames, video frame, or
      even packet, depending on the encapsulation chosen to an
      individual QUIC stream.  This is independent of whether streams,
      would be uni- or bi-directional.

   Option 3 eliminates the head of line blocking problem of options 1.
   and 2.  because QUIC does not provide any ordering across different
   streams.  Using larger semantic units (e.g., GOPs) for stream
   mapping, would provide for more efficient stream number usage.
   However, all stream frames are still transmitted reliably.  This
   implies that QUIC will perform retransmissions even for packets that
   would be too late already.

   Mapping each video frame or packet to a different stream would raise
   an issue with stream numbering unless all RTP sessions are
   multiplexed on a single UDP socket anyway and then all RTP packets
   would simply be mapped to different streams.

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   An open question here would be how to deal with additional data
   channels that don't use RTP.  Ideally, it should be possible that
   those be within the same QUIC connection (if QUIC is used as
   transport) to avoid consuming again more port numbers.  Since, on the
   one hand, data channels can be set up and torn down at any time and,
   on the other hand, media packets are transmitted continuously, a need
   arises to set aside streams for data channels.  One option would be
   "reserving" those streams in some form.  But then, how many to
   reserve?  Moreover, this would be incompatible with the slides stream
   number window being used by QUIC.  Alternatively, one would need to
   synchronize the use of QUIC streams in real-time between the
   signaling and application channels and the media packet transmission.
   This may be hard to achieve and also suffers from the problem of the
   stream id window moving fast with frame transmissions.  A third
   option would be adding another demultiplexing structure (e.g., to
   different RTP headers from data packets) and use a similar scheme of
   one application data unit (ADU) per stream for other applications.
   While feasible, this appears somewhat cumbersome in the mapping.

   We finally need to consider inter RTP stream synchronisation and how/
   if this would be affected by use of multiple QUIC streams.

   None of the above schemes appear truly satisfactory from a system
   design perspective.  This may call for some refined design
   considerations for QUIC, which we will begin discussing in section 4.

3.4.  Mapping RTCP packets

   RTCP is a bi-directional stream unlike RTP streams which are
   unidirectional.  There can be for example a video stream receiver
   that only receives video content but will send and receive RTCP

   The current discussion on uni-directional streams direction will
   allow both uni- and bi- dirctional QUIC streams in the same QUIC
   connection.  Such a solution will allow multiplexing of RTP and RTCP
   streams in the same QUIC connection.

   An issue to consider is the encryption of RTCP messages.  The RTP
   secure profiles SAVP [RFC3711] and SAVPF [RFC5124] allow NULL cipher
   for RTCP with message integrity.  Using a NULL cipher allow RTP
   middleboxes to monitor the RTP delivery quality.

   Whether to use a single stream for forward RTCP and another for
   reverse could be a function of the streams being uni- or
   bidirectional in the end.  Another question to answer is if there
   should be one stream per SSRC per direction for RTCP.  Finally, RTCP

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   packets may also be lost and they contain timing information.
   Avoiding HoL blocking may thus also be important.

3.5.  Mapping of RTP header extensions

   QUIC provides a reliable protocol which addresses the requirement in
   [I-D.ietf-avtcore-rfc5285-bis] to transmit the RTP header extension
   in a couple of RTP packets to provide better reliability.  Still if
   we use mapping option 3 we will still need to transmit the RTP header
   extensions more than once.  Using QUIC as a transport for RTP will
   have all RTP header extensions encrypted allowing only entities that
   terminate a QUIC connection to decode them.  RTP header extension as
   defined in [I-D.ietf-avtcore-rfc5285-bis] can be sent in the clear
   and provide information to RTP middleboxes enabling them to route
   encrypted RTP packets.  Currently the following header extensions are
   used for routing of encrypted RTP streams.  Client to mixer audio
   level [RFC6464].  Frame marking [I-D.ietf-avtext-framemarking] and
   splicing interval [I-D.ietf-avtext-splicing-notification].

4.  Design considerations for QUIC

   This section will address design implications for QUIC and the
   interaction with QUIC of both RTP and RTCP.  We expect to discuss the
   following aspects in the future:

      Reliability (or restransmission) control for stream frames
      Congestion control adaptation
      RTCP mapping
      Priming QUIC 0-RTT
      Multiparty operation

5.  SDP Extensions for Negotiating RTP-over-QUIC


6.  Security Considerations

   RTP is used as a plain payload for QUIC, exploiting its multiplexing
   capabilities.  To this end, the RTP packets are protected
   (confidentiality) by the QUIC security mechanisms.  Hence, the
   security considerations pertinent to QUIC apply.

   QUIC is by its very nature a transport layer security mechanisms.
   RTP traffic will thus be protected on a single transport hop only.
   As soon RTP topologies more complex than a point-to-point connection
   are used (e.g., [RFC7667]), RTP traffic will lose its end-to-end

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   protection as transport connections are terminated at the
   intermediary, even if this acts just as a relay.

7.  IANA Considerations

   There are no IANA considerations at this point.

8.  Acknowledgments

9.  References

9.1.  Normative References

   [RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP
              Payload Format Specifications", BCP 36, RFC 2736,
              DOI 10.17487/RFC2736, December 1999,

   [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,

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,

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   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <http://www.rfc-editor.org/info/rfc5124>.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
              2010, <http://www.rfc-editor.org/info/rfc5763>.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,

   [RFC6464]  Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication", RFC 6464,
              DOI 10.17487/RFC6464, December 2011,

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,

   [RFC8083]  Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", RFC 8083,
              DOI 10.17487/RFC8083, March 2017,

9.2.  Informative References

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              Brosh, E., Baset, S., Rubinstein, D., and H. Schulzrinne,
              "The Delay-Friendliness of TCP", Proceedings of ACM
              SIGMETRICS, 2008.

              Singer, D., Desineni, H., and R. Even, "A General
              Mechanism for RTP Header Extensions", draft-ietf-avtcore-
              rfc5285-bis-12 (work in progress), June 2017.

              Berger, E., Nandakumar, S., and M. Zanaty, "Frame Marking
              RTP Header Extension", draft-ietf-avtext-framemarking-04
              (work in progress), March 2017.

              Xia, J., Even, R., Huang, R., and D. Lingli, "RTP/RTCP
              extension for RTP Splicing Notification", draft-ietf-
              avtext-splicing-notification-09 (work in progress), August

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-38 (work in progress), April 2017.

              Kuehlewind, M. and B. Trammell, "Applicability of the QUIC
              Transport Protocol", draft-ietf-quic-applicability-00
              (work in progress), July 2017.

              Bishop, M., "Hypertext Transfer Protocol (HTTP) over
              QUIC", draft-ietf-quic-http-04 (work in progress), June

              Kuehlewind, M., Trammell, B., and D. Druta, "Manageability
              of the QUIC Transport Protocol", draft-ietf-quic-
              manageability-00 (work in progress), July 2017.

              Iyengar, J. and I. Swett, "QUIC Loss Detection and
              Congestion Control", draft-ietf-quic-recovery-04 (work in
              progress), June 2017.

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              Thomson, M. and S. Turner, "Using Transport Layer Security
              (TLS) to Secure QUIC", draft-ietf-quic-tls-04 (work in
              progress), June 2017.

              Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
              and Secure Transport", draft-ietf-quic-transport-04 (work
              in progress), June 2017.

              Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
              control for RTP media", draft-ietf-rmcat-coupled-cc-06
              (work in progress), March 2017.

              Holmer, S., Lundin, H., Carlucci, G., Cicco, L., and S.
              Mascolo, "A Google Congestion Control Algorithm for Real-
              Time Communication", draft-ietf-rmcat-gcc-02 (work in
              progress), July 2016.

              Zhu, X., Pan, R., Ramalho, M., Cruz, S., Jones, P., Fu,
              J., and S. D'Aronco, "NADA: A Unified Congestion Control
              Scheme for Real-Time Media", draft-ietf-rmcat-nada-04
              (work in progress), March 2017.

              Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
              for Multimedia", draft-ietf-rmcat-scream-cc-09 (work in
              progress), May 2017.

              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-18
              (work in progress), March 2017.

              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-26 (work in progress), March

              Singh, V., Nagy, M., Ott, J., and L. Eggert, "Congestion
              Control Using FEC for Conversational Media", draft-singh-
              rmcat-adaptive-fec-03 (work in progress), March 2016.

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Authors' Addresses

   Joerg Ott
   TU Munich
   Boltzmannstrasse 3
   Garching bei Muenchen

   Email: ott@in.tum.de

   Roni Even

   Email: Even.roni@huawei.com

   Colin Perkins
   University of Glasgow

   Email: csp@csperkins.org

   Varun Singh
   Callstats I/O

   Email: varun@callstats.io

Ott, et al.              Expires January 4, 2018               [Page 13]