Internet DRAFT - draft-penno-rtcweb-pcp
draft-penno-rtcweb-pcp
RTCWEB R. Penno
Internet-Draft T. Reddy
Intended status: Standards Track D. Wing
Expires: November 26, 2013 Cisco
M. Boucadair
France Telecom
May 25, 2013
PCP Considerations for WebRTC Usage
draft-penno-rtcweb-pcp-00
Abstract
This document describes the motivations for WebRTC applications to be
PCP-aware and the benefits provided by PCP-capable NATs and
Firewalls.
Status of This Memo
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Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Notational Conventions . . . . . . . . . . . . . . . . . . . 3
3. Advantages of using PCP with WebRTC . . . . . . . . . . . . . 3
3.1. Firewalls Blocking UDP . . . . . . . . . . . . . . . . . 3
3.2. Firewalls permit specific WebRTC servers . . . . . . . . 5
3.3. ICE Lite . . . . . . . . . . . . . . . . . . . . . . . . 5
3.4. Reducing Call Set-Up Time . . . . . . . . . . . . . . . . 6
3.4.1. ICE Speedup . . . . . . . . . . . . . . . . . . . . . 6
3.4.2. Pre-allocating ports to speed call setup time . . . . 6
3.5. NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.6. Optimizing NAT and Firewall Keepalives . . . . . . . . . 7
3.7. Faster Flow Failure Detection . . . . . . . . . . . . . . 8
3.8. 3GPP Selective IP Traffic Offload (SIPTO) . . . . . . . . 8
3.9. Auditing . . . . . . . . . . . . . . . . . . . . . . . . 9
3.10. NAT64 . . . . . . . . . . . . . . . . . . . . . . . . . . 10
4. Usage of PCP with STUN and TURN . . . . . . . . . . . . . . . 10
4.1. STUN . . . . . . . . . . . . . . . . . . . . . . . . . . 10
4.2. TURN . . . . . . . . . . . . . . . . . . . . . . . . . . 11
5. Security Considerations . . . . . . . . . . . . . . . . . . . 11
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
7. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 12
8.1. Normative References . . . . . . . . . . . . . . . . . . 12
8.2. Informative References . . . . . . . . . . . . . . . . . 13
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15
1. Introduction
Port Control Protocol (PCP, [RFC6887]) provides a mechanism to
describe a flow to the network. The primary driver for PCP has been
creating port mappings on NAT and firewall devices. When doing this,
PCP pushes flow information from the host into the network
(specifically to the network's NAT or firewall device), and receives
information back from the network (from the NAT or firewall device).
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The Web Real-Time communication (WebRTC) framework
[I-D.ietf-rtcweb-overview] provides the protocol building blocks to
support direct, interactive, real-time communication using audio,
video, collaboration, games, etc., between peer web-browsers. WebRTC
application use Interactive Connectivity Establishment (ICE) protocol
[RFC5245] for gathering candidates, prioritizing them, choosing
default ones, exchanging them with the remote party, pairing them and
ordering them into check lists. Once all of the above steps have
been completed the participating ICE agents can begin a phase of
connectivity checks and eventually select a pair of candidates that
will be used for real-time communication.
This specification describes the reasons for WebRTC applications to
be PCP-aware and use PCP along side with STUN and TURN. It also
explains the benefits for a network that deploy PCP-controlled NATs
and Firewalls.
2. Notational Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
This document uses terms defined in [RFC5389] and [RFC5766].
eNodeB: The eNodeB is a base station entity that supports the Long-
Term Evolution (LTE) air interface [RFC6459].
3. Advantages of using PCP with WebRTC
The below sections explain the problems with NAT and Firewall,
current techniques used to solved them and the PCP solution in these
scenarios.
3.1. Firewalls Blocking UDP
Enterprise networks may deploy firewalls with restrictive policies
configured to block UDP traffic. These firewalls may be configured
to permit TCP or HTTP(s) traffic only. One of the reasons for
blocking UDP could be that there is no way for the firewall to
determine when the endpoints have terminated the call, in which case
the firewall has to close the dynamic mapping based on firewall UDP
mapping timer value. [RFC4787] mandates that the UDP mapping timer
for NAT must not expire in less than 2 minutes and recommends a
default value of five minutes or more. Firewalls are likely to
follow the same recommendation for their UDP mapping timer, which
would be applicable to both IPv4 and IPv6 firewalls. The behavioural
requirements for IPv6 firewalls is explained in section 3.2.3 of
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[RFC6092]. [I-D.hutton-rtcweb-nat-firewall-considerations] gives
details of other organization e.g. a public service agency or
university that deploy firewall which may have restrictive firewall
policy to block UDP traffic.
Modern firewalls may also have application-layer gateways (ALGs)
perform policy enforcement to permit peer-to-peer UDP media session.
Using the ALG, a firewall can determine when the call is terminated
and close any dynamic mappings created for the media session. But
the problem is the session signaling between the WebRTC application
running in the browser and the web server could be using TLS, in
which case the ALG no longer has access to the signaling. Moreover,
WebRTC does not enforce a particular session signaling protocol to be
used, so firewalls using ALGs would fail to inspect the signaling to
identify the 5-tuple used for each media stream. Furthermore, the
session signaling and the peer-to-peer media may traverse different
Firewalls.
Using TURN for all such communication to by-pass firewall causes the
following problems:
o TURN server could increase media latency as explained in section
4.1.2.2 of [RFC5245]. Using a reliable and ordered protocol like
TCP instead of UDP to transfer real-time media is problematic as
delays would be directly noticeable and may be unacceptable to the
user.
o High-end TURN server would be needed (For example when TLS-over-
TCP transport is used between the client and the server) to cater
to all such calls.
o TURN server could either be located in the DMZ of the enterprise
network or located in the public Internet. If the TURN server is
located in the public Internet it comes at a high cost to the
provider of the TURN server, since the server typically needs a
high-bandwidth connection to the Internet as explained in the
Introduction of [RFC5766]. As a consequence, it is best to use a
TURN server only when a direct communication path cannot be found.
When the client and a peer use ICE to determine the communication
path, ICE will use hole punching techniques to search for a direct
path first and only use a TURN server when a direct path cannot be
found.
o Some of the other limitations of TURN explained in section 2.6 of
[RFC5766] are, the value of the Diffserv field may not be
preserved, the Explicit Congestion Notification (ECN) field may be
reset etc.
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PCP resolves the above problems by restricting firewall traversal to
authorized PCP clients and communicating mapping lifetimes and call
termination between the PCP client and the PCP-controlled firewall.
A PCP Server can also enforce per-host quotas for mappings.
3.2. Firewalls permit specific WebRTC servers
When an enterprise uses a trusted WebRTC server deployed in a 3rd
party network for communication, the enterprise firewall could have
granular policies to permit peer-to-peer UDP media session only when
the call is initiated using the selected WebRTC server (Dr. Good) it
trusts and block others (Dr. Evil). Firewall policy has a white-
list of permitted outside applications/sites and can blacklist
HTTP(S) connections via various forms of detections (destination DNS
lookup, HTTP URL Filtering, DPI proxy that at least performs HTTPS
inspection of URL in certificate, Subject Name of TLS exchange and
validates SSL records etc). Firewall in this configuration would
also block TCP connection to arbitrary TURN servers in the Internet.
3GPP networks may also have a similar configuration where IMS
services of certain other operators are permitted and others are
blocked [[TR33.830].
With PCP, this problem is solved by associating the media session
with the signaling session. This is done by sending a cryptographic
token in the signaling which authorizes the firewall mapping for the
media session.
3.3. ICE Lite
For scenarios where the client is connected to the public Internet
and has public IP address at which it can receive packets from the
remote peer and uses ICE LITE implementation explained in section 2.7
of [RFC5245], the ICE Lite endpoint will not generate its own ICE
connectivity checks, by definition. Thus, if an ICE Lite endpoint is
behind a firewall that blocks unsolicited incoming traffic then ICE
Lite will fail.
This workaround for solving the problem is by using full ICE or by
changing the filtering policy on the firewall to permit unsolicited
incoming UDP traffic which would effectively disable the purpose of
firewall. Full ICE will take more time to be adapted especially with
legacy VoIP equipment which will initially start with ICE-Lite
implementation as discussed in section 6 of [I-D.cbran-rtcweb-nat].
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With PCP, a firewall can filter incoming UDP traffic and PCP client
can communicate exceptions to the firewall to permit specific
mappings when a call is active. In this way, the ICE Lite endpoint
and its network are protected from unsolicited incoming UDP traffic,
and can still operate using ICE Lite (rather than full ICE).
3.4. Reducing Call Set-Up Time
There are initiatives to speedup ICE processing in order to reduce
call setup time using techniques such as Trickle ICE
[I-D.rescorla-mmusic-ice-trickle] and RTP multiplexing Section 4.4 of
[I-D.ietf-rtcweb-rtp-usage]. Trickle ICE can begin connectivity
checks while the endpoint is still gathering candidates and can
considerably shorten the time necessary for ICE processing to
complete. RTP multiplexing suggests to bind interactive audio and
interactive video to the same 5-tuple {dest addr, source addr,
protocol, dest port, source port} to optimize NAT resource usage and
shorten the call setup time.
PCP can help reduce call set-up time by speeding up ICE and, if
appropriate, at the same time allowing each media for flow over a
different 5-tuple.
3.4.1. ICE Speedup
ICE requires time to perform its setup operations. This time grows
in proportion to the number of transport sessions which must be
opened in order to support the call. If using a different IP
addresses and/or ports for audio versus video streams, call setup
time will increase. The precise amount of this increase depends on
the type of NAT and other factors like packet loss. The use of RTP
Multiplexing technique introduces some QoS challenges in many
networks, e.g., In Mobile Networks the QoS considerations are
explained in Section 4.1 of [I-D.reddy-rtcweb-mobile].
Fast call setup time and QoS can both be retained by using PCP.
External IP addresses and ports can be learnt faster using PCP than
other techniques because the PCP client is communicating only with
PCP servers in the Home and Service Provider network. In contrast,
STUN and TURN servers may be located halfway around the world from
the endpoint adding delay to learn server-reflexive and relayed
candidates. Trickle ICE can begin connectivity checks using the
candidates learnt from PCP, while the endpoint is still gathering
other candidate types and thus can considerably shorten the time
necessary for ICE processing to complete.
3.4.2. Pre-allocating ports to speed call setup time
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The external IP:port allocated through PCP belong to the client for
duration of the lifetime of the mapping. This means that
connectivity checks for a new call can begin immediately using the
already allocated external IP:port and if necessary the client can
extend the lifetime of the mapping. TURN allocations can also be
extended using Refresh transaction to update the time-to-expiry of
existing allocation and thus can be used for a new call immediately.
Server Reflexive candidates learnt using STUN can also be maintained
for a new call but requires the endpoint to send frequent keepalives
to prevent the NAT and firewall mappings from expiring.
The PCP client for fast call setup can also use PORT_SET option
[I-D.ietf-pcp-port-set] requesting the PCP server to pre-allocate
contiguous ports with port parity preservation.
3.5. NAT
Direct peer-to-peer communication is not possible if both NATs are of
a certain type that changes the outside port number when connecting
to new hosts (NAT behaviour "address-dependent mapping" or "address
and-port-dependent mapping" as described in [RFC4787]).
When such NAT devices are encountered, communication can be
established using a media relay (TURN) server. But using TURN
servers is expensive as explained in section 4.1.1.2 of [RFC5245] and
other challenges of using TURN are discussed in Section 3.1 . Relayed
candidates should only be used as last-resort when connectivity
checks using other candidate types are not successful.
PCP improves this situation by creating explicit bindings on PCP-
controlled NATs and can adjust their mapping and filtering behavior
so that connections can be successfully created. PCP can also
recursively communicate with multiple layers of NATs using
[I-D.ietf-pcp-proxy]. Usage of STUN and PCP for learning candidates,
prioritization, encoding them in offer or answer is explained in
Section 4.1.
3.6. Optimizing NAT and Firewall Keepalives
Applications like WebRTC need to keep their Network Address
Translator (NAT) and firewall mappings alive for long periods of
time, even when they are otherwise not sending or receiving any
traffic. The signaling protocol used for WebRTC would want to keep
the client-server connection alive for as long as the application is
running. When the WebRTC application has otherwise no traffic to
send, specific keep-alive messages are sent periodically to ensure
that the NAT/Firewall state in the middle does not expire. The
endpoint would also have to send keepalives for the media session to
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keep NAT/Firewall bindings alive. As NAT/firewall mapping timers may
be short and unknown to the endpoint, the keepalive messages are sent
frequently.
In cellular mobile networks, frequent keepalive messages make the
radio transition between active and power-save states causing
signaling congestion. The excessive time spent on the active state
due to keepalives also greatly reduces the battery life of the
cellular connected devices such as smartphones or tablets.
PCP is useful to reduce NAT and firewall keepalive messages (e.g.,
Section 3.4 of [I-D.reddy-pcp-optimize-keepalives]) for both
signaling protocol and media session.
3.7. Faster Flow Failure Detection
If a NAT device has rebooted, lost its mappings or has its external
IP address changed then it may take few minutes before the endpoint
realizes that the connectivity is lost, that would result in
disruption of signaling and media traffic. Application can find that
the signaling session is broken by using TCP keepalive probes, the
time taken to detect that the connection is broken depends on the
frequency of keepalive probes. If the endpoint is using sendonly
media streams, it may take few minutes based on RTCP reports to
realize that the connectivity is lost. WebRTC client will then have
to re-establish connection with the WebRTC server and initiate ICE
restart.
Using the Rapid Recovery procedure explained in Section 14 of
[RFC6887], the PCP client upon receiving a PCP ANNOUNCE from a PCP
server, becomes aware that the PCP server has rebooted or lost its
mapping state. The PCP client issues new PCP requests to recreate
any lost mapping state and thus reconstructs lost mappings fast
enough that existing media streams do not break and re-establish
connectivity with its WebRTC server.
If for some reason PCP server determines that some or all of its
mappings have become unusable (e.g., when a home gateway is assigned
a different external IPv4 address by the upstream DHCP server) then
the PCP server automatically repairs its mappings and notifies its
clients about the new External IP address and port as part of the
Rapid Recovery techniques explained in Section 14.2 of [RFC6887].
The client based on this notification can use MICE
[I-D.wing-mmusic-ice-mobility] or ICE Restart to achieve RTP
Mobility.
3.8. 3GPP Selective IP Traffic Offload (SIPTO)
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Given the exponential growth in the mobile data traffic, Mobile
Operators are looking for ways to offload some of the IP traffic
flows at the nearest access edge that has an Internet peering point.
This approach results in efficient usage of the mobile packet core
and helps lower the transport cost. Since Release 10, 3GPP starts
supporting of Selected IP Traffic Offload (SIPTO) function defined in
[TS23.060][TS23.060], [TS23.401]. The SIPTO function allows an
operator to offload certain types of traffic at a network node close
to the UE's point of attachment to the access network. Limited
Mobility support available with SIPTO is explained in section 2.3.3
of [I-D.zuniga-dmm-gap-analysis].
If SIPTO is carried out in a Traffic offload Function (TOF) entity in
the path between the Radio stations and the Mobile Gateway (MGW) as
explained in [I-D.reddy-rtcweb-mobile] and the Mobile Node (MN) roams
from one eNodeB and changes its point of attachment to a new eNodeB
NAT changes. In this case host candidates for the MN will not change
but MN will be behind a new NAT after roaming. It may take few
minutes before the MN realizes that the connectivity is lost,
resulting in disruption of signalling and media traffic. Application
can find that the signaling session is broken by using TCP keepalive
probes, the time taken to detect that connection is broken depends on
the frequency of the keepalive probes. If the endpoint is using
sendonly media streams, it may take few minutes based on RTCP reports
to realize that the connectivity is lost. WebRTC client will then
have to re-establish connection with the WebRTC server and initiate
ICE restart.
The problem can be mitigated by the following mechanism using PCP:
When TOF receives the SIPTO rules for the MN, the PCP-controlled NAT
at TOF sends unicast PCP ANNOUNCE response to the MN informing it
that the NAT has changed. WebRTC application using PCP can verify
that external IP addresses and ports have changed for the media
streams and proceed accordingly (e.g., MICE
[I-D.wing-mmusic-ice-mobility] or ICE Restart to achieve RTP
Mobility).
3.9. Auditing
On certain networks, it is necessary to audit communications across
the network firewall and attribute those communications to certain
users or users running certain applications. The use case for
auditing is also explained in Section 4.2.5.1 of
[I-D.ietf-rtcweb-use-cases-and-requirements].
Today, this is done by tracking IP address assignment on the network
and auditing lots of mappings created by firewalls.
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PCP improves that auditing by PCP Authentication
[I-D.ietf-pcp-authentication]. A PCP server can audit all traffic
including media sessions from inside an enterprise premises to any
external peer. An enterprise that uses an WebRTC based web
application for communication and desires to audit all WebRTC based
application sessions used from inside the company towards any
external peer can deploy a PCP-controlled firewall and enforce a
policy on the PCP-controlled firewall to mandate PCP client
authentication. Only after successful authentication, PCP client
will be permitted to create dynamic mappings on the firewalls and
NATs.
3.10. NAT64
For the IPv6-only WebRTC client to establish media session with
IPv4-only WebRTC client it must learn prefix64(s).
The workaround for solving the problem is by using heuristics is
explained in [I-D.ietf-behave-nat64-discovery-heuristic]. Various
other solutions including STUN for discovery based on heuristics are
discussed in [I-D.ietf-behave-nat64-learn-analysis].
PCP allows to learn PREFIX64 when a NAT64 is in the path
[I-D.ietf-pcp-nat64-prefix64]. PCP client can directly communicate
with PCP-controlled NAT64 device to learn the Prefix64(s). This
feature is useful to help establishing successful media session
between an IPv6-only WebRTC client and an IPv6-only WebRTC client.
The other advantages of using PCP is that endpoint will be notified
whenever the Network Specific Prefix (NSP) is changed and endpoint
will also learn multiple NSPs configured in the network.
Experimental results related to the use of this feature for SIP-based
applications in general are provided in Section 4.2 of
[I-D.boucadair-pcp-nat64-experiments].
4. Usage of PCP with STUN and TURN
4.1. STUN
This section explains the procedure to use STUN and PCP with ICE
[RFC5245]:
The ICE agent learns external IP addresses and ports using the PCP
MAP opcode. If server reflexive candidates and external IP addresses
learnt using PCP are different than the candidates learnt through
STUN, the PCP discovered candidates are encoded in the ICE offer and
answer just like the server reflexive candidates learnt using STUN
[RFC5389]. When using the recommended formula explained in
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Section 4.1.2.1 of [RFC5245] to compute priority for the candidate
learnt through PCP, the ICE agent should use a preference value
greater than the server reflexive candidate and hence they are tested
before the server reflexive candidates.
The recommended type preference value is 105 for candidates
discovered using PCP and is explained in section 4.2 of [RFC6544].
During connectivity checks the ICE agent SHOULD check if the XOR-
MAPPED-ADDRESS from the STUN Binding response matches the external
address and port provided by PCP MAP response.
o If the match is successful, then it indicates that only PCP-aware
NATs exist between the peers. PCP can further be used to keep the
NAT bindings alive and close the mappings.
o If the match is not successful then it indicates PCP unaware NATs
exist between the peers.
4.2. TURN
TURN server may be used for the following reasons even if PCP capable
Firewalls and NATs exist:
o Users of WebRTC based web application may choose to use TURN so as
to not expose the host candidate addresses to the remote peer for
privacy reasons.
o IPv6 support in TURN includes IPv4-to-IPv6 and IPv6-to-IPv4
relaying [RFC6156].
o ICE connectivity checks using the candidates provided by STUN and
PCP could fail because the endpoint is behind PCP-unaware NAT that
performs address-dependent mapping and thus only relayed candidate
allocated from the TURN server gets selected for media.
o TURN server could also be used for RTP Mobility
[I-D.wing-mmusic-ice-mobility], etc.
5. Security Considerations
Security considerations discussed in [RFC6887] are to be taken into
account. PCP authentication [I-D.ietf-pcp-authentication] MAY also
be used.
6. IANA Considerations
This document does not require any action from IANA.
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7. Acknowledgments
The authors would like to thank Charles Eckel for review and
comments.
8. References
8.1. Normative References
[I-D.ietf-pcp-authentication]
Wasserman, M., Hartman, S., and D. Zhang, "Port Control
Protocol (PCP) Authentication Mechanism", draft-ietf-pcp-
authentication-01 (work in progress), October 2012.
[I-D.ietf-pcp-proxy]
Boucadair, M., Penno, R., and D. Wing, "Port Control
Protocol (PCP) Proxy Function", draft-ietf-pcp-proxy-02
(work in progress), February 2013.
[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-06 (work in progress),
February 2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[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.
[RFC6092] Woodyatt, J., "Recommended Simple Security Capabilities in
Customer Premises Equipment (CPE) for Providing
Residential IPv6 Internet Service", RFC 6092, January
2011.
[RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal
Using Relays around NAT (TURN) Extension for IPv6", RFC
6156, April 2011.
[RFC6887] Wing, D., Cheshire, S., Boucadair, M., Penno, R., and P.
Selkirk, "Port Control Protocol (PCP)", RFC 6887, April
2013.
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8.2. Informative References
[I-D.boucadair-mmusic-altc]
Boucadair, M., Kaplan, H., Gilman, R., and S.
Veikkolainen, "Session Description Protocol (SDP)
Alternate Connectivity (ALTC) Attribute", draft-boucadair-
mmusic-altc-09 (work in progress), January 2013.
[I-D.boucadair-pcp-nat64-experiments]
Abdesselam, M., Boucadair, M., Hasnaoui, A., and J.
Queiroz, "PCP NAT64 Experiments", draft-boucadair-pcp-
nat64-experiments-00 (work in progress), September 2012.
[I-D.cbran-rtcweb-nat]
Bran, C., Kaufman, M., Jennings, C., and J. Rosenberg,
"WebRTC Network Address Translation", draft-cbran-rtcweb-
nat-02 (work in progress), October 2011.
[I-D.hutton-rtcweb-nat-firewall-considerations]
Stach, T., Hutton, A., and J. Uberti, "RTCWEB
Considerations for NATs, Firewalls and HTTP proxies",
draft-hutton-rtcweb-nat-firewall-considerations-00 (work
in progress), March 2013.
[I-D.ietf-behave-nat64-discovery-heuristic]
Savolainen, T., Korhonen, J., and D. Wing, "Discovery of
the IPv6 Prefix Used for IPv6 Address Synthesis", draft-
ietf-behave-nat64-discovery-heuristic-17 (work in
progress), April 2013.
[I-D.ietf-behave-nat64-learn-analysis]
Korhonen, J. and T. Savolainen, "Analysis of solution
proposals for hosts to learn NAT64 prefix", draft-ietf-
behave-nat64-learn-analysis-03 (work in progress), March
2012.
[I-D.ietf-pcp-nat64-prefix64]
Boucadair, M., "Learn NAT64 PREFIX64s using PCP", draft-
ietf-pcp-nat64-prefix64-02 (work in progress), May 2013.
[I-D.ietf-pcp-port-set]
Sun, Q., Boucadair, M., Sivakumar, S., Zhou, C., Tsou, T.,
and S. Perreault, "Port Control Protocol (PCP) Extension
for Port Set Allocation", draft-ietf-pcp-port-set-01 (work
in progress), May 2013.
[I-D.ietf-rtcweb-overview]
Penno, et al. Expires November 26, 2013 [Page 13]
Internet-Draft PCP with WebRTC May 2013
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-06 (work
in progress), February 2013.
[I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-10 (work in
progress), December 2012.
[I-D.reddy-pcp-optimize-keepalives]
Reddy, T., Isomaki, M., Wing, D., and P. Patil,
"Optimizing NAT and Firewall Keepalives Using Port Control
Protocol (PCP)", draft-reddy-pcp-optimize-keepalives-01
(work in progress), January 2013.
[I-D.reddy-rtcweb-mobile]
Reddy, T., Kaippallimalil, J., R, R., and R. Ejzak,
"Considerations with WebRTC in Mobile Networks", draft-
reddy-rtcweb-mobile-03 (work in progress), May 2013.
[I-D.rescorla-mmusic-ice-trickle]
Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol", draft-
rescorla-mmusic-ice-trickle-01 (work in progress), October
2012.
[I-D.wing-mmusic-ice-mobility]
Wing, D., Patil, P., Reddy, T., and P. Martinsen,
"Mobility with ICE (MICE)", draft-wing-mmusic-ice-
mobility-03 (work in progress), January 2013.
[I-D.zuniga-dmm-gap-analysis]
Zuniga, J., Bernardos, C., Melia, T., and C. Perkins,
"Mobility Practices and DMM Gap Analysis", draft-zuniga-
dmm-gap-analysis-03 (work in progress), December 2012.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
Penno, et al. Expires November 26, 2013 [Page 14]
Internet-Draft PCP with WebRTC May 2013
[RFC6459] Korhonen, J., Soininen, J., Patil, B., Savolainen, T.,
Bajko, G., and K. Iisakkila, "IPv6 in 3rd Generation
Partnership Project (3GPP) Evolved Packet System (EPS)",
RFC 6459, January 2012.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B.B., and A.B.
Roach, "TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, March 2012.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B.B., and A.B.
Roach, "TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, March 2012.
[TR33.830]
3GPP, , "3rd Generation Partnership Project; Technical
Specification Group Services and System Aspects;
Feasibility study on IMS firewall traversal (Release
12).", September 2012.
[TS23.060]
3GPP, , ""General Packet Radio Service (GPRS); Service
description; Stage 2", June 2012.", September 2012.
[TS23.401]
3GPP, , "General Packet Radio Service (GPRS) enhancements
for Evolved Universal Terrestrial Radio Access Network (E-
UTRAN) access (Release 11), 3GPP TS 23.401, V11.2.0 (2012-
06).", September 2012.
Authors' Addresses
Reinaldo Penno
Cisco Systems, Inc.
170 West Tasman Drive
San Jose 95134
USA
Email: repenno@cisco.com
Tirumaleswar Reddy
Cisco Systems, Inc.
Cessna Business Park, Varthur Hobli
Sarjapur Marathalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: tireddy@cisco.com
Penno, et al. Expires November 26, 2013 [Page 15]
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Dan Wing
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, California 95134
USA
Email: dwing@cisco.com
Mohamed Boucadair
France Telecom
Rennes 35000
France
Email: mohamed.boucadair@orange.com
Penno, et al. Expires November 26, 2013 [Page 16]
