One document matched: draft-penno-rtcweb-pcp-00.xml
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<?rfc tocompact="yes"?>
<?rfc tocdepth="3"?>
<?rfc tocindent="yes"?>
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<rfc category="std" docName="draft-penno-rtcweb-pcp-00" ipr="trust200902">
<front>
<title abbrev="PCP with WebRTC">PCP Considerations for WebRTC
Usage</title>
<author fullname="Reinaldo Penno" initials="R." surname="Penno">
<organization abbrev="Cisco">Cisco Systems, Inc.</organization>
<address>
<postal>
<street>170 West Tasman Drive</street>
<city>San Jose</city>
<region></region>
<code>95134</code>
<country>USA</country>
</postal>
<phone></phone>
<email>repenno@cisco.com</email>
<uri></uri>
</address>
</author>
<author fullname="Tirumaleswar Reddy" initials="T." surname="Reddy">
<organization abbrev="Cisco">Cisco Systems, Inc.</organization>
<address>
<postal>
<street>Cessna Business Park, Varthur Hobli</street>
<street>Sarjapur Marathalli Outer Ring Road</street>
<city>Bangalore</city>
<region>Karnataka</region>
<code>560103</code>
<country>India</country>
</postal>
<email>tireddy@cisco.com</email>
</address>
</author>
<author fullname="Dan Wing" initials="D." surname="Wing">
<organization abbrev="Cisco">Cisco Systems, Inc.</organization>
<address>
<postal>
<street>170 West Tasman Drive</street>
<city>San Jose</city>
<region>California</region>
<code>95134</code>
<country>USA</country>
</postal>
<email>dwing@cisco.com</email>
</address>
</author>
<author fullname="Mohamed Boucadair" initials="M." surname="Boucadair">
<organization>France Telecom</organization>
<address>
<postal>
<street></street>
<city>Rennes</city>
<region></region>
<code>35000</code>
<country>France</country>
</postal>
<email>mohamed.boucadair@orange.com</email>
</address>
</author>
<date />
<workgroup>RTCWEB</workgroup>
<abstract>
<t>This document describes the motivations for WebRTC applications to be
PCP-aware and the benefits provided by PCP-capable NATs and
Firewalls.</t>
</abstract>
</front>
<middle>
<section anchor="introduction" title="Introduction">
<t>Port Control Protocol (PCP, <xref target="RFC6887"></xref>) 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).</t>
<t>The Web Real-Time communication (WebRTC) framework <xref
target="I-D.ietf-rtcweb-overview"></xref> 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 <xref target="RFC5245"></xref> 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.</t>
<t>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.</t>
<!--
<t>Details related to the specific case where WebRTC is enabled in a
controlled network environment are discussed in <xref
target="no_checks"></xref>.</t>
-->
</section>
<section anchor="notation" title="Notational Conventions">
<t>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 <xref
target="RFC2119"></xref>.</t>
<t>This document uses terms defined in <xref target="RFC5389"></xref>
and <xref target="RFC5766"> </xref>.</t>
<t>eNodeB: The eNodeB is a base station entity that supports the
Long-Term Evolution (LTE) air interface <xref
target="RFC6459"></xref>.</t>
</section>
<section anchor="PCP" title="Advantages of using PCP with WebRTC">
<t>The below sections explain the problems with NAT and Firewall,
current techniques used to solved them and the PCP solution in these
scenarios.</t>
<section anchor="FW" title="Firewalls Blocking UDP">
<t>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. <xref target="RFC4787"></xref> 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 <xref
target="RFC6092"></xref>. <xref
target="I-D.hutton-rtcweb-nat-firewall-considerations"></xref> 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.</t>
<t>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.</t>
<t>Using TURN for all such communication to by-pass firewall causes
the following problems:</t>
<t><list style="symbols">
<t>TURN server could increase media latency as explained in
section 4.1.2.2 of <xref target="RFC5245"></xref>. 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.</t>
<t>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.</t>
<t>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 <xref target="RFC5766"></xref>. 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.</t>
<t>Some of the other limitations of TURN explained in section 2.6
of <xref target="RFC5766"></xref> are, the value of the Diffserv
field may not be preserved, the Explicit Congestion Notification
(ECN) field may be reset etc.</t>
</list></t>
<t>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.</t>
</section>
<section title="Firewalls permit specific WebRTC servers">
<t>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 [<xref
target="TR33.830"></xref>.</t>
<t>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.</t>
</section>
<section title="ICE Lite">
<t>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 <xref target="RFC5245"></xref>, 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.</t>
<t>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 <xref
target="I-D.cbran-rtcweb-nat"></xref>.</t>
<t>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).</t>
</section>
<section title="Reducing Call Set-Up Time">
<t>There are initiatives to speedup ICE processing in order to reduce
call setup time using techniques such as Trickle ICE <xref
target="I-D.rescorla-mmusic-ice-trickle"></xref> and RTP multiplexing
<xref target="I-D.ietf-rtcweb-rtp-usage">Section 4.4 of</xref>.
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.</t>
<t>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.</t>
<section title="ICE Speedup">
<t>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 <xref
target="I-D.reddy-rtcweb-mobile"></xref>.</t>
<t>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.</t>
</section>
<section title="Pre-allocating ports to speed call setup time">
<t>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.</t>
<t>The PCP client for fast call setup can also use PORT_SET option
<xref target="I-D.ietf-pcp-port-set"></xref> requesting the PCP
server to pre-allocate contiguous ports with port parity
preservation.</t>
</section>
</section>
<section title="NAT">
<t>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 <xref
target="RFC4787"></xref>).</t>
<t>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 <xref
target="RFC5245"></xref> and other challenges of using TURN are
discussed in <xref target="FW"></xref> . Relayed candidates should
only be used as last-resort when connectivity checks using other
candidate types are not successful.</t>
<t>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 <xref
target="I-D.ietf-pcp-proxy"></xref>. Usage of STUN and PCP for
learning candidates, prioritization, encoding them in offer or answer
is explained in <xref target="STUN"></xref>.</t>
</section>
<section title="Optimizing NAT and Firewall Keepalives">
<t>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 keep
NAT/Firewall bindings alive. As NAT/firewall mapping timers may be
short and unknown to the endpoint, the keepalive messages are sent
frequently.</t>
<t>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.</t>
<t>PCP is useful to reduce NAT and firewall keepalive messages (e.g.,
Section 3.4 of <xref
target="I-D.reddy-pcp-optimize-keepalives"></xref>) for both signaling
protocol and media session.</t>
</section>
<section title="Faster Flow Failure Detection">
<t>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.</t>
<t>Using the Rapid Recovery procedure explained in Section 14 of <xref
target="RFC6887"></xref>, 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.</t>
<t>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 <xref
target="RFC6887"></xref>. The client based on this notification can
use MICE <xref target="I-D.wing-mmusic-ice-mobility"></xref> or ICE
Restart to achieve RTP Mobility.</t>
</section>
<section title="3GPP Selective IP Traffic Offload (SIPTO)">
<t>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
<xref target="TS23.060"></xref><xref target="TS23.060"></xref><xref
target="TS23.401">,</xref>. 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 <xref
target="I-D.zuniga-dmm-gap-analysis"></xref>.</t>
<t>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 <xref target="I-D.reddy-rtcweb-mobile"></xref> 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.</t>
<t>The problem can be mitigated by the following mechanism using
PCP:</t>
<t>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 <xref
target="I-D.wing-mmusic-ice-mobility"></xref> or ICE Restart to
achieve RTP Mobility).</t>
</section>
<section title="Auditing">
<t>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 <xref
target="I-D.ietf-rtcweb-use-cases-and-requirements"></xref>.</t>
<t>Today, this is done by tracking IP address assignment on the
network and auditing lots of mappings created by firewalls.</t>
<t>PCP improves that auditing by PCP Authentication <xref
target="I-D.ietf-pcp-authentication"></xref>. 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.</t>
</section>
<section title="NAT64">
<t>For the IPv6-only WebRTC client to establish media session with
IPv4-only WebRTC client it must learn prefix64(s).</t>
<t>The workaround for solving the problem is by using heuristics is
explained in <xref
target="I-D.ietf-behave-nat64-discovery-heuristic"></xref>. Various
other solutions including STUN for discovery based on heuristics are
discussed in <xref
target="I-D.ietf-behave-nat64-learn-analysis"></xref>.</t>
<t>PCP allows to learn PREFIX64 when a NAT64 is in the path <xref
target="I-D.ietf-pcp-nat64-prefix64"></xref>. 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.</t>
<t>Experimental results related to the use of this feature for
SIP-based applications in general are provided in <xref
target="I-D.boucadair-pcp-nat64-experiments">Section 4.2
of</xref>.</t>
</section>
</section>
<section anchor="usage" title="Usage of PCP with STUN and TURN">
<t></t>
<section anchor="STUN" title="STUN">
<t>This section explains the procedure to use STUN and PCP with ICE
<xref target="RFC5245"></xref>:</t>
<t>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
<xref target="RFC5389"></xref>. When using the recommended formula
explained in Section 4.1.2.1 of <xref target="RFC5245"></xref> 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.</t>
<t>The recommended type preference value is 105 for candidates
discovered using PCP and is explained in section 4.2 of <xref
target="RFC6544"></xref>.</t>
<t>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.</t>
<t><list style="symbols">
<t>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.</t>
<t>If the match is not successful then it indicates PCP unaware
NATs exist between the peers.</t>
</list></t>
</section>
<section title="TURN">
<t>TURN server may be used for the following reasons even if PCP
capable Firewalls and NATs exist:</t>
<t><list style="symbols">
<t>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.</t>
<t>IPv6 support in TURN includes IPv4-to-IPv6 and IPv6-to-IPv4
relaying <xref target="RFC6156"></xref>.</t>
<t>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.</t>
<t>TURN server could also be used for RTP Mobility <xref
target="I-D.wing-mmusic-ice-mobility"></xref>, etc.</t>
</list></t>
</section>
</section>
<!--
<section anchor="no_checks"
title="Sample Use Case: WebRTC in Controlled Environments">
<t>This section focuses on the sample use case where WebRTC is deployed
in a controlled environment such as within a mobile network or among
sites belonging to the same enterprise network. Within this section, a
controlled environment denotes any network which does not require
connectivity checks. Such controlled environments may allow:<list
style="numbers">
<t>Communications between two WebRTC clients both connected to the
same underlying managed infrastructure.</t>
<t>Communications between a WebRTC client and a legacy SIP UA
(through a WebRTC-IMS Gateway for instance).</t>
<t>Communications from/to a WebRTC client to/from any destination
located in PSTN or PLMN (through a PSTN Gateway for instance).</t>
</list></t>
<t>In such controlled networks, a WebRTC agent can be packaged with the
required PCP tuning parameters (e.g., identity of PCP server(s),
explicit activation of PCP service, etc.).</t>
<t>The following procedure depicts the typical generic steps to be
followed by a WebRTC agent:</t>
<t><list style="symbols">
<t>To detect whether PCP service is available, the WebRTC client
proceeds to the following on its bootstrapping:<list style="symbols">
<t>Check if PCP service is explicitly enabled. This can be made
available by setting a dedicated configuration parameter or
accessing to a dedicated environment variable.</t>
<t>If the PCP Server is explicitly configured, the WebRTC will
use that server to install PCP mappings whenever needed.</t>
<t>If the PCP service is explicitly disabled, PCP must not be
used to install mappings.</t>
<t>If the PCP service is not explicitly disabled or it is
explicitly enabled but no PCP server is provisioned, the WebRTC
must initiate a procedure to discover its PCP server(s):<list
style="symbols">
<t>This can be achieved by sending a MAP request to discover
the external IP address (see Section 11.6 of <xref
target="RFC6887"></xref>). This request is first sent to the
default router and then to the PCP IP anycast address in
case of failure.</t>
<t>If no answer is received, the WebRTC agent concludes no
PCP server is available.</t>
<t>If an answer is received, the WebRTC stores the identity
of discovered PCP server(s).</t>
</list></t>
<t>If a PCP server is detected on a network, the WebRTC agent
caches this information in a dedicated parameter called
STALE_PCP_SERVER. On reboot of WebRTC agent, the content of this
variable will be used by the WebRTC agent to contact its PCP
server when connected to the same network. If this variable is
not set, the discovery procedure detailed in the above bullet
must be followed.</t>
</list></t>
<t>Upon detection of a PCP Server, the WebRTC agent uses PCP to
install a mapping for the signaling message. It may also prereserve
a pair of ports to be used for media sessions.</t>
<t>If the WebRTC is dual-stack, ALTC attribute <xref
target="I-D.boucadair-mmusic-altc"></xref> is used to signal one
IPv4 address and one IPv6 address.</t>
</list></t>
<t>Implementing a proof of concept of this procedure is ongoing.
Experiment results will be published once available.</t>
</section>
-->
<section anchor="security" title="Security Considerations">
<t>Security considerations discussed in <xref target="RFC6887"></xref>
are to be taken into account. PCP authentication <xref
target="I-D.ietf-pcp-authentication"></xref> MAY also be used.</t>
</section>
<section anchor="iana" title="IANA Considerations">
<t>This document does not require any action from IANA.</t>
</section>
<section title="Acknowledgments">
<t>The authors would like to thank Charles Eckel for review and
comments.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.5766'?>
<?rfc include='reference.RFC.5389'?>
<?rfc include='reference.RFC.6156'?>
<?rfc include='reference.I-D.ietf-pcp-proxy'?>
<?rfc include='reference.RFC.6887'?>
<?rfc include='reference.RFC.6092'?>
<?rfc include='reference.I-D.ietf-pcp-authentication'?>
<?rfc include='reference.I-D.ietf-rtcweb-rtp-usage'?>
<?rfc ?>
</references>
<references title="Informative References">
<?rfc include="reference.RFC.5245"?>
<?rfc include="reference.RFC.6544"
?>
<?rfc include="reference.RFC.4787"?>
<?rfc include='reference.RFC.6459'?>
<?rfc include='reference.I-D.ietf-rtcweb-use-cases-and-requirements'?>
<?rfc include='reference.I-D.ietf-pcp-port-set'?>
<?rfc include='reference.I-D.ietf-rtcweb-overview'?>
<?rfc include='reference.I-D.wing-mmusic-ice-mobility'
?>
<?rfc include='reference.I-D.rescorla-mmusic-ice-trickle'
?>
<?rfc include='reference.I-D.reddy-rtcweb-mobile'?>
<?rfc include='reference.I-D.reddy-pcp-optimize-keepalives'?>
<?rfc include='reference.I-D.boucadair-pcp-nat64-experiments'?>
<?rfc include='reference.I-D.boucadair-mmusic-altc'?>
<?rfc include='reference.I-D.ietf-pcp-nat64-prefix64'
?>
<?rfc include='reference.I-D.zuniga-dmm-gap-analysis'
?>
<?rfc include='reference.I-D.hutton-rtcweb-nat-firewall-considerations'
?>
<?rfc include='reference.RFC.6544'
?>
<?rfc include='reference.I-D.cbran-rtcweb-nat' ?>
<?rfc include='reference.I-D.ietf-behave-nat64-discovery-heuristic'?>
<?rfc include='reference.I-D.ietf-behave-nat64-learn-analysis'?>
<reference anchor="TS23.060" target="">
<front>
<title>"General Packet Radio Service (GPRS); Service description;
Stage 2", June 2012.</title>
<author fullname="3GPP" surname="3GPP">
<organization></organization>
</author>
<date day="0" month="September" year="2012" />
</front>
</reference>
<reference anchor="TS23.401" target="">
<front>
<title>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).</title>
<author fullname="3GPP" surname="3GPP">
<organization></organization>
</author>
<date day="0" month="September" year="2012" />
</front>
</reference>
<reference anchor="TR33.830" target="">
<front>
<title>3rd Generation Partnership Project; Technical Specification
Group Services and System Aspects; Feasibility study on IMS firewall
traversal (Release 12).</title>
<author fullname="3GPP" surname="3GPP">
<organization></organization>
</author>
<date day="0" month="September" year="2012" />
</front>
</reference>
<!---->
</references>
</back>
</rfc>
| PAFTECH AB 2003-2026 | 2026-04-24 04:21:32 |