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
TUM
July 3, 2017
RTP over QUIC
draft-rtpfolks-quic-rtp-over-quic-00
Abstract
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
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This Internet-Draft will expire on January 4, 2018.
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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
[I-D.ietf-mmusic-sdp-bundle-negotiation]
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
decisions.
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
stream.
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
messages.
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
API
Multiparty operation
5. SDP Extensions for Negotiating RTP-over-QUIC
TBD
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,
<http://www.rfc-editor.org/info/rfc2736>.
[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,
<http://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<http://www.rfc-editor.org/info/rfc3264>.
[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,
<http://www.rfc-editor.org/info/rfc3711>.
[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,
<http://www.rfc-editor.org/info/rfc4588>.
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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,
<http://www.rfc-editor.org/info/rfc5245>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>.
[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,
<http://www.rfc-editor.org/info/rfc5764>.
[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,
<http://www.rfc-editor.org/info/rfc6464>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<http://www.rfc-editor.org/info/rfc7667>.
[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017,
<http://www.rfc-editor.org/info/rfc8083>.
9.2. Informative References
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[Delay-TCP]
Brosh, E., Baset, S., Rubinstein, D., and H. Schulzrinne,
"The Delay-Friendliness of TCP", Proceedings of ACM
SIGMETRICS, 2008.
[I-D.ietf-avtcore-rfc5285-bis]
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.
[I-D.ietf-avtext-framemarking]
Berger, E., Nandakumar, S., and M. Zanaty, "Frame Marking
RTP Header Extension", draft-ietf-avtext-framemarking-04
(work in progress), March 2017.
[I-D.ietf-avtext-splicing-notification]
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
2016.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
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.
[I-D.ietf-quic-applicability]
Kuehlewind, M. and B. Trammell, "Applicability of the QUIC
Transport Protocol", draft-ietf-quic-applicability-00
(work in progress), July 2017.
[I-D.ietf-quic-http]
Bishop, M., "Hypertext Transfer Protocol (HTTP) over
QUIC", draft-ietf-quic-http-04 (work in progress), June
2017.
[I-D.ietf-quic-manageability]
Kuehlewind, M., Trammell, B., and D. Druta, "Manageability
of the QUIC Transport Protocol", draft-ietf-quic-
manageability-00 (work in progress), July 2017.
[I-D.ietf-quic-recovery]
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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[I-D.ietf-quic-tls]
Thomson, M. and S. Turner, "Using Transport Layer Security
(TLS) to Secure QUIC", draft-ietf-quic-tls-04 (work in
progress), June 2017.
[I-D.ietf-quic-transport]
Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
and Secure Transport", draft-ietf-quic-transport-04 (work
in progress), June 2017.
[I-D.ietf-rmcat-coupled-cc]
Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion
control for RTP media", draft-ietf-rmcat-coupled-cc-06
(work in progress), March 2017.
[I-D.ietf-rmcat-gcc]
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.
[I-D.ietf-rmcat-nada]
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.
[I-D.ietf-rmcat-scream-cc]
Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation
for Multimedia", draft-ietf-rmcat-scream-cc-09 (work in
progress), May 2017.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-18
(work in progress), March 2017.
[I-D.ietf-rtcweb-rtp-usage]
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
2016.
[I-D.singh-rmcat-adaptive-fec]
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
Germany
Email: ott@in.tum.de
Roni Even
Huawei
Israel
Email: Even.roni@huawei.com
Colin Perkins
University of Glasgow
UK
Email: csp@csperkins.org
Varun Singh
Callstats I/O
Finland
Email: varun@callstats.io
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