One document matched: draft-ietf-mmusic-ice-02.txt
Differences from draft-ietf-mmusic-ice-01.txt
MMUSIC J. Rosenberg
Internet-Draft dynamicsoft
Expires: January 17, 2005 July 19, 2004
Interactive Connectivity Establishment (ICE): A Methodology for
Network Address Translator (NAT) Traversal for Multimedia Session
Establishment Protocols
draft-ietf-mmusic-ice-02
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Copyright Notice
Copyright (C) The Internet Society (2004). All Rights Reserved.
Abstract
This document describes a methodology for Network Address Translator
(NAT) traversal for multimedia session signaling protocols, such as
the Session Initiation Protocol (SIP). This methodology is called
Interactive Connectivity Establishment (ICE). ICE makes use of
existing protocols, such as Simple Traversal of UDP Through NAT
(STUN) and Traversal Using Relay NAT (TURN). ICE makes use of STUN
in peer-to-peer cooperative fashion, allowing participants to
discover, create and verify mutual connectivity.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Multimedia Signaling Protocol Abstraction . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. Overview of ICE . . . . . . . . . . . . . . . . . . . . . . . 8
5. Detailed ICE Algorithm . . . . . . . . . . . . . . . . . . . . 10
5.1 Initiator Processing . . . . . . . . . . . . . . . . . . . 10
5.1.1 Sending the Initiate Message . . . . . . . . . . . . . 10
5.1.2 Processing the Accept . . . . . . . . . . . . . . . . 10
5.2 Responder Processing . . . . . . . . . . . . . . . . . . . 11
5.2.1 Processing the Initiate Message . . . . . . . . . . . 11
5.3 Common Procedures . . . . . . . . . . . . . . . . . . . . 12
5.3.1 Gathering Transport Addresses . . . . . . . . . . . . 12
5.3.2 Enabling STUN on Each Local Transport Address . . . . 13
5.3.3 Prioritizing the Transport Addresses and Choosing
a Default . . . . . . . . . . . . . . . . . . . . . . 14
5.3.4 Sending STUN Connectivity Checks . . . . . . . . . . . 16
5.3.5 Receiving STUN Requests . . . . . . . . . . . . . . . 21
6. Running STUN on Derived Transport Addresses . . . . . . . . . 23
6.1 STUN on a TURN Derived Transport Address . . . . . . . . . 23
6.2 STUN on a STUN Derived Transport Address . . . . . . . . . 24
7. XML Schema for ICE Messages . . . . . . . . . . . . . . . . . 26
8. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
8.1 Port Restricted . . . . . . . . . . . . . . . . . . . . . 29
9. Mapping ICE into SIP . . . . . . . . . . . . . . . . . . . . . 32
10. Security Considerations . . . . . . . . . . . . . . . . . . 34
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . 35
12. IAB Considerations . . . . . . . . . . . . . . . . . . . . . 36
12.1 Problem Definition . . . . . . . . . . . . . . . . . . . . 36
12.2 Exit Strategy . . . . . . . . . . . . . . . . . . . . . . 36
12.3 Brittleness Introduced by ICE . . . . . . . . . . . . . . 37
12.4 Requirements for a Long Term Solution . . . . . . . . . . 38
12.5 Issues with Existing NAPT Boxes . . . . . . . . . . . . . 38
13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39
14. References . . . . . . . . . . . . . . . . . . . . . . . . . 40
14.1 Normative References . . . . . . . . . . . . . . . . . . . . 40
14.2 Informative References . . . . . . . . . . . . . . . . . . . 40
Author's Address . . . . . . . . . . . . . . . . . . . . . . . 41
Intellectual Property and Copyright Statements . . . . . . . . 42
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1. Introduction
A multimedia session signaling protocol is a protocol that exchanges
control messages between a pair of agents for the purposes of
establishing the flow of media traffic between them. This media flow
is distinct from the flow of control messages, and may take a
different path through the network. Examples of such protocols are
the Session Initiation Protocol (SIP) [3], the Real Time Streaming
Protocol (RTSP) [5] and the International Telecommunications Union
(ITU) H.323.
These protocols, by nature of their design, are difficult to operate
through Network Address Translators (NAT). Because their purpose in
life is to establish a flow of packets, they tend to carry IP
addresses within their messages, which is known to be problematic
through NAT [6]. The protocols also seek to create a media flow
directly between participants, so that there is no application layer
intermediary between them. This is done to reduce media latency,
decrease packet loss, and reduce the operational costs of deploying
the application. However, this is difficult to accomplish through
NAT. A full treatment of the reasons for this is beyond the scope of
this specification.
Numerous solutions have been proposed for allowing these protocols to
operate through NAT. These include Application Layer Gateways
(ALGs), the Middlebox Control Protocol [7], Simple Traversal of UDP
through NAT (STUN) [1], Traversal Using Relay NAT [16], Realm
Specific IP [8][9], symmetric RTP [10], along with session
description extensions needed to make them work, such as [2].
Unfortunately, these techniques all have pros and cons which make
each one optimal in some network topologies, but a poor choice in
others. The result is that administrators and implementors are
making assumptions about the topologies of the networks in which
their solutions will be deployed. This introduces a lot of
complexity and brittleness into the system. What is needed is a
single solution which is flexible enough to work well in all
situations.
This specification provides that solution. It is called Interactive
Connectivity Establishment, or ICE. ICE makes use of many of the
protocols above, but uses them in a specific methodology which avoids
many of the pitfalls of using any one alone. ICE uses STUN and TURN
without extension, and allows for other similar protocols to be used
as well. However, it does require additional signaling capabilities
to be introduced into the multimedia session signaling protocols.
For those protocols which make use of the Session Description
Protocol (SDP), this specification defines the necessary extensions
to it. Other protocols will need to define their own mechanisms.
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2. Multimedia Signaling Protocol Abstraction
This specification defines a general methodology that allows the
media streams of multimedia signaling protocols to successfully
traverse NAT. This methodology is independent of any particular
signaling protocol. In order to discuss the methodology, we need to
to define an abstraction of a multimedia signaling system, and define
terms that can be used throughout this specification. Figure 1 shows
the abstraction.
+-----------+
| |
| |
> | Signaling |\
/ | Relay | \
/ | | \
Initiate / | | \ Initiate
Message / / +-----------+ \ Message
/ / < \
/ / \ \
/ / \ \
/ / Accept Accept \ \
/ / Message Message \ >
/ / \
+-----------+ / \ +-----------+
| | < | |
| | Media Stream | |
| Session | ................................ | Session |
| Initiator | | Responder |
| | Media Stream | |
| | ................................ | |
+-----------+ +-----------+
Figure 1
Communications occur between two clients - the session initiator and
the session responder, also referred to as the initiator and
responder. The initiator is the one that decides to engage in
communications. To do so, it sends an initiate message. The
initiate message contains parameters that describe the capabilities
and configuration of media streams for the initiator. This message
may travel through signaling intermediaries, called a signaling
relay, before finally arriving at the session responder. Assuming
the session responder wishes to communicate, it generates an accept
message, which is relayed back to the initiator. This message
contains capabilities and configuration of media streams for the
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responder. As a result, media streams are established between the
initiator and responder. The signaling protocol may also support an
operation that allows for termination of the communications session.
We refer to this signaling message as a terminate message.
This abstraction is readily mapped to SIP, RTSP, and H.323, amongst
others. For SIP, the initiator is the User Agent Client (UAC), the
responder is the User Agent Server (UAS), the initiate message is a
SIP message containing an SDP offer (for example, an INVITE), the
accept message is a SIP message containing an SDP answer (for
example, a 200 OK), and the terminate message is a BYE. For RTSP,
the initiator is the RTSP client, the responder is the RTSP server,
the initiate message is a SETUP message, and the accept message is a
SETUP response.
This specification defines parameters that need to be included in
these various signaling messages in order to implement the
functionality described by ICE. Those parameters are represented in
XML for convenience. Any multimedia signaling protocol that uses ICE
will need to define how to map those parameters into its own protocol
messages. Section 9 provides such a mapping for SIP.
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3. Terminology
Several new terms are introduced in this specification:
Session Initiator: A software or hardware entity that, at the request
of a user, tries to establish communications with another entity,
called the session responder. A session initiator is also called
an initiator.
Initiator: Another term for a session initiator.
Session Responder: A software or hardware entity that receives a
request for establishment of communications from the session
initiator, and either accepts or declines the request. A session
responder is also called a responder.
Responder: Another term for a session responder.
Client: Either the initiator or responder.
Peer: From the perspective of one of the clients in a session, its
peer is the other client. Specifically, from the perspective of
the initiator, the peer is the responder. From the perspective of
the responder, the peer is the initiator.
Signaling Relay: An intermediary of signaling messages. Examples are
SIP proxies and H.323 Gatekeepers.
Initiate Message: The signaling message used by an initiator to
establish communications. It contains capabilities and other
information needed by the responder to send media to the
initiator.
Accept Message: The signaling message used by a responder to agree to
communications. It contains capabilities and other information
needed by the initiator to send media to the responder.
Terminate Message The signaling message used by a client to terminate
the session and associated media streams.
Transport Address: The combination of an IP address and port.
Local Transport Address: A local transport address is a transport
address that has been allocated from the operating system on the
host. This includes transport addresses obtained through Virtual
Private Networks (VPNs) and transport addresses obtained through
Realm Specific IP (RSIP) [8] (which lives at the operating system
level). Transport addresses are typically obtained by binding to
an interface.
Derived Transport Address: A derived transport address is a transport
address which is associated with, but different from, a local
transport address. The derived transport address is associated
with the local transport address in that packets sent to the
derived transport address are received on the socket bound to that
local transport address. Derived addresses are obtained using
protocols like STUN and TURN, and more generally, any UNSAF
protocol [11].
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Peer Derived Transport Address: A peer derived transport address is a
derived transport address learned from a STUN server running
within a peer in a media session.
TURN Derived Transport Address: A derived transport address obtained
from a TURN server.
STUN Derived Transport Address: A derived transport address obtained
from a STUN server whose address has been provisioned into the UA.
This, by definition, excludes Peer Derived Transport Addresses.
Unilateral Allocations: Queries made to a network server which
provides an UNSAF service.
Bilateral Allocations: Addresses obtained by using an UNSAF service
that actually runs on the peer of the communications session.
Peer derived transport addresses are synonymous with bilateral
allocations.
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4. Overview of ICE
ICE makes the fundamental assumption that clients exist in a network
of segmented connectivity. This segmentation is the result of a
number of addressing realms in which a client can simultaneously be
connected. We use "realms" here in the broadest sense. A realm is
defined purely by connectivity. Two clients are in the same realm
if, when they exchange the addresses each has in that realm, they are
able to send packets to each other. This includes IPv6 and IPv4
realms, which actually use different address spaces, in addition to
private networks connected to the public Internet through NAT.
The key assumption in ICE is that a client cannot know, apriori,
which address realms it shares with any peer it may wish to
communicate with. Therefore, in order to communicate, it has to try
connecting to addresses in all of the realms.
Before the initiator establishes a session, it obtains as many IP
address and port combinations in as many address realms as it can.
These adresses all represent potential points at which the initiator
will receive a specific media stream. Any protocol that provides a
client with an IP address and port on which it can receive traffic
can be used. These include STUN, TURN, RSIP, and even a VPN. The
client also uses any local interface addresses. A dual-stack v4/v6
client will obtain both a v6 and a v4 address/port. The only
requirement is that, across all of these addresses, the initiator can
be certain that at least one of them will work for any responder it
might communicate with. Unfortunately, if the initiator communicates
with a peer that doesn't support ICE, only one address can be
provided to that peer. As such, the client will need to choose one
default address, which will be used by non-ICE clients. This would
typically be a TURN derived transport address, as it is most likely
to work with unknown non-ICE peers.
The initiator then runs a STUN server on each of the local transport
addresses it has obtained. The initiator will need to be able to
demultiplex STUN messages and media messages received on that IP
address and port, and process them appropriately. All of these
addresses are placed into the initiate message, and they are ordered
in terms of preference. Preference is a matter of local policy, but
typically, lowest preference would be given to transport addresses
learned from a TURN server (i.e., TURN derived transport addresses).
The initiate message also conveys the STUN username and password
which are required to gain access to the STUN server on each address/
port combination.
The initiate message is sent to the responder. This specification
does not address the issue of how the signaling messages themselves
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traverse NAT. It is assumed that signaling protocol specific
mechanisms are used for that purpose. The responder follows a
similar process as the initiator followed; it obtains addresses from
local interfaces, STUN servers, TURN servers, etc., and it places all
of them into the accept message.
Once the responder receives the initiate message, it has a set of
potential addresses it can use to communicate with the initiator.
The initiator will be running a STUN server at each address. The
responder sends a STUN request to each address, in parallel. When
the initiator receives these, it sends a STUN response. If the
responder receives the STUN response, it knows that it can reach its
peer at that address. It can then begin to send media to that
address. As additional STUN responses arrive, the responder will
learn about additional transport addresses which work. If one of
those has a higher priority than the one currently in use, it starts
sending media to that one instead. No additional control messages
(i.e., SIP signaling) occur for this change.
The STUN messages described above happen while the accept message is
being sent to the intitiator. Once the intitiator receives the
accept message, it too will have a set of potential addresses with
which it can communicate to the responder. It follows exactly the
same process described above.
Furthermore, when a either the initiator or responder receives a STUN
request, it takes note of the source IP address and port of that
request. It compares that transport address to the existing set of
potential addresses. If it's not amongst them, it gets added as
another potential address. The incoming STUN message provides the
client with enough context to associate that transport address with a
STUN username, STUN password, and priority, just as if it had been
sent in an initiate or accept message. As such, the client begins
sending STUN messages to it as well, and if those succeed, the
address can be used if it has a higher priority.
After a successful STUN transaction, the client will re-perform the
STUN query periodically to revalidate connectivity. This allows for
recovery from NAT failures, or from route flaps which may cause
packets to suddenly traverse a different NAT. As such, the address
used as the destination for media is the highest priority address to
which connectivity currently exists.
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5. Detailed ICE Algorithm
This section describes the detailed processing needed for ICE.
5.1 Initiator Processing
5.1.1 Sending the Initiate Message
When the initiator wishes to begin communications, it starts by
gathering transport addresses, as described in Section 5.3.1, and
starting a STUN server on each local transport address, as described
in Section 5.3.2. This process can actually happen at any time
before sending an initiate message. A client can pre-gather
transport addresses, using a user interface cue (such as picking up
the phone, or entry into an address book) as a hint that
communications is imminent.
When it comes time to initiate communications, it determines a
priority for each one and identifies one as a default, as described
in Section 5.3.3.
The next step is to construct the initiate message. Section 7
provides the XML schema for the initiate message. The message
consists of a series of media streams. For each media stream, there
is a default address and a list of alternates. The default address
is the one that will be used by responders that don't understand ICE
(for SIP, this is accomplished by mapping the default address into
the m and c line in the SDP). The alternates represent addresses
that the responder should also try. In SIP, these are conveyed with
the new SDP alt parameter.
The client then encodes all of its available transport addresses
(including the default) as a series of alternate elements. Each
alternate element conveys a transport address for RTP, one for RTCP,
a STUN username fragment and STUN password. The client MUST assign
each alternate a unique identifier. These identifiers MUST be unique
across all alternates used within the session. This identifier is
encoded in the "id" attribute of the alternate element. The priority
for the transport address, as computed above, is included as an
attribute as well.
Once the initiate message is constructed, it is sent.
5.1.2 Processing the Accept
There are two possible cases for processing of the Accept message.
If the recipient of the Initiate message did not support ICE, the
Accept message will only contain the default address information. As
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a result, the initiator knows that it cannot perform its connectivity
checks. In this case, it SHOULD just send to the transport address
listed. However, if local configuration information tells the
initiator to try connectivity checks by sending them through the TURN
server, this means that packets sent directly to responder may be
dropped by a local firewall. To deal with this, the initiator SHOULD
issue a SEND command using this new transport address. The SEND
command contains the media packet to send to the responder. Once
this command has been accepted, the initiator SHOULD send all media
packets to the TURN server, which will then forward them towards the
responder.
If the Accept message contains alternates, it implies that the
responder supported ICE. In that case, the initiator takes each
transport address, STUN username, STUN password and priority, and
places them into a list, called the candidate list. It then begins
processing the candidate list as described in Section 5.3.4. That
processing associates a state with each transport address. As
described there, once a successful STUN query is made to the STUN
server at an address, the initiator can begin sending media to that
address.
5.2 Responder Processing
5.2.1 Processing the Initiate Message
Upon receipt of the initiate message, the client starts gathering
transport addresses, as described in Section 5.3.1, and starts a STUN
server on each local transport address, as described in Section
5.3.2. This processing is done immediately on receipt of the
request, to prepare for the case where the user should accept the
call, or early media needs to be generated.
At some point, the responder will decide to accept or reject the
communications. A rejection terminates ICE processing, of course.
In the case of acceptance, the accept message is constructed as
follows.
The client first determines a priority for each transport address it
has gathered, and identifies one as a default, as described in
Section 5.3.3.
Constructing the accept proceeds identically to the way in which the
initiate message is constructed (Section 5.1.1).
The accept is then sent.
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5.3 Common Procedures
This section discusses procedures that are common between initiator
and responder.
5.3.1 Gathering Transport Addresses
A client gathers addresses when it believes that communications is
imminent. For initiators, this occurs before sending an initiate
message (Section 5.1.1). For responders, it occurs before sending a
accept message (Section 5.2.1).
There are two types of addresses a client can gather - local
transport addresses, and derived transport addresses. Local
transport addresses are obtained by binding to an ephemeral port on
an interface (physical or virtual) on the host. A multi-homed host
SHOULD attempt to bind on all interfaces for all media streams it
wishes to receive. For media streams carried using the Real Time
Transport Protocol (RTP) [12], the client will need to bind to an
ephemeral port for both RTP and RTCP.
The result will be a set of local transport addresses. The client
may also have access to servers that provide unilateral self-address
fixing (UNSAF) [11]. Examples of such protocols include STUN, TURN,
and TEREDO [15]. All ICE implementations MUST implement STUN and
TURN, but MAY, through configuration, disable the use of STUN or TURN
for unilateral address allocation (STUN is mandatory for the
connectivity checks described below). When disabled, it MUST be
possible through user or administrator operation to re-enable. This
allows all implementations to have the breadth of protocol support
needed to work in all situations, with the flexibility to turn if off
if its not needed.
These protocols work by having the client send, from a specific local
transport address, some kind of message to a server. The server
provides to the client, in some kind of response, an additional
transport address, called a derived transport address. This derived
transport address is derived from the local transport address. Here,
derivation means that a request sent to the derived transport address
might (under good network conditions) reach the client on its local
transport address.
For each of these protocols, the client may have access to a
multiplicity of servers. For example, a user connected to a natted
cable access network might have access to a STUN server in the
private cable network and in the public Internet. For each local
transport address, the client SHOULD obtain an address from every
server for each protocol it supports. The result of this will be a
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set of derived transport addresses, with each derived address
associated with the local transport address it is derived from.
5.3.2 Enabling STUN on Each Local Transport Address
Once the client has obtained a set of transport addresses, it starts
a STUN server on each local transport address (including ones used
for RTCP). This, by definition, means that the STUN service will be
reached for requests sent to the derived addresses.
However, the client does not need to provide STUN service on any
other IP address or port, unlike the STUN usage described in [1].
The need to run the service on multiple ports is to support the
change flags. However, those flags are not needed with ICE, and the
server SHOULD reject, with a 400 response, any STUN requests with
these flags set.
Furthermore, there is no need to support TLS or to be prepared to
receive SharedSecret request messages. Those messages are used to
obtain shared secrets to be used with BindingRequests. However, with
ICE, usernames and passwords are exchanged in the signaling protocol.
The client will receive both STUN requests and media packets on each
local transport address. The client MUST be able to disambiguate
them. In the case of RTP/RTCP, this disambiguation is easy. RTP and
RTCP packets start with the bits 0b10 (v=2). The first two bits in
STUN are always 0b00. This disambiguation also works for packets
sent using Secure RTP [13], since the RTP header is in the clear.
Disambiguating STUN with other media stream protocols may be more
complicated. However, it can always be possible with arbitrarily
high probabilities by selecting an appropriately random username (see
below).
The need to run STUN on the same transport address as the media
stream represents the "ugliest" piece of ICE. However, it is an
essential part of the story. By sending STUN requests to the very
same place media is sent, any bindings learned through STUN will be
useful even when communicating through symmetric NATs. This results
in a substantial increase in the scope of applicability of STUN.
For each local transport address where a STUN server is running, the
client MUST choose a username fragment and a password. The username
fragment created by the client will be concatenated with the fragment
created by its peer. The result will serve as the username provided
by its peer in STUN requests. By creating the username as a
combination of information from each side of a call, it allows a
client to correlate the source of the request with a candidate
transport address. This is discussed further below.
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The username fragment MUST be globally unique, so that no other host
will select a username with the same value. This username fragment
and password will be passed to its peer in an initiate or accept
message. As such, the process described in this section will
associate, with each local transport address, a username fragment and
password. The client also associates this same username fragment and
password with any transport addresses derived from the local
transport address.
The global uniqueness requirement stems from the lack of uniquenes
afforded by IP addresses. Consider clients A, B, and C. A and B are
within private enterprise 1, which is using 10.0.0.0/8. C is within
private enterprise 2, which is also using 10.0.0.0/8. As it turns
out, B and C both have IP address 10.0.1.1. A initiates
communications to C. C, in its accept message, provides A with its
transport addresses. In this case, thats 10.0.1.1:8866 and 8877. As
it turns out, B is in a session at that same time, and is also using
10.0.1.1:8866 and 8877. This means that B has a STUN server running
on those ports, just as C does. A will send a STUN request to
10.0.1.1:8866 and 8877. However, these do not go to C as expected.
Instead, they go to B. If B just replied to them, A would believe it
has connectivity to C, when in fact it has connectivity to a
completely different user, B. To fix this, the STUN username takes
on the role of a unique identifier. C provides A with a unique
username. A uses this username in its STUN query to 10.0.1.1:8866.
This STUN query arrives at B. However, the username is unknown to B,
and so the request is rejected. A treats the rejected STUN request
as if there were no connectivity to C (which is actually true).
Therefore, the error is avoided.
Once the STUN server is started, it MUST run continuously until the
session is completed. While the server is running, it MUST act as a
normal STUN server, but MUST only accept STUN requests from clients
that authenticate, as discussed below in Section 5.3.5
5.3.3 Prioritizing the Transport Addresses and Choosing a Default
The prioritization process takes a list of transport addresses, and
associates each with a priority. This priority reflects the desire
that the UA has to receive media on that address, and is assigned as
a value from 0 to 1 (1 being most preferred). Priorities are
ordinal, so that their significance is only relative to other
transport address priorities in the same list.
This specification makes no normative recommendations on how the
prioritization is done. However, some useful guidelines are
suggested on how such a prioritization can be determined.
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One criteria for choosing one transport address over another is
whether or not that transport address involves the use of a relay.
That is, if media is sent to that transport address, will the media
first transit a relay before being received. TURN derived transport
addresses make use of relays (the TURN server), as to any local
transport addresses associated with a VPN server. When media is
transited through a relay, it can increase the latency between
transmission and reception. It can increase the packet losses,
because of the additional router hops that may be taken. It may
increase the cost of providing service, since media will be routed in
and right back out of a relay run by the provider. If these concerns
are important, transport addresses with this property can be listed
with lower priority.
Another criteria for choosing one address over another is IP address
family. ICE works with both IPv4 and IPv6. It therefore provides a
transition mechanism that allows dual-stack hosts to prefer
connectivity over IPv6, but to fall back to IPv4 in case the v6
networks are disconnected (due, for example, to a failure in a 6to4
relay) [14]. It can also help with hosts that have both a native
IPv6 address and a 6to4 address. In such a case, higher priority
could be afforded to the native v6 address, followed by the 6to4
address, followed by a native v4 address. This allows a site to
obtain and begin using native v6 addresss immediately, yet still
fallback to 6to4 addresses when communicating with clients in other
sites that do not yet have native v6 connectivity.
Another criteria for choosing one address over another is security.
If a user is a telecommuter, and therefore connected to their
corporate network and a local home network, they may prefer their
voice traffic to be routed over the VPN in order to keep it on the
local network when communicating within the enterprise, but use the
local network when communicating with users outside of the
enterprise.
Another criteria for choosing one address over another is topological
awareness. This is most useful for transport addresses which make
use of relays (including TURN and VPN). In those cases, if a client
has preconfigured or dynamically discovered knowledge of the
topological proximity of the relays to itself, it can use that to
select closer relays with higher priority.
Once the transport addresses have been prioritized, one is selected
as the default. This is the address that will be used by a peer that
doesn't understand ICE. The default has no relevance when
communicating with an ICE capable peer. As such, it is RECOMMENDED
that the default be chosen based on the likelihood of that address
being useful when communicating with a peer that doesn't support ICE.
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This will frequently be a TURN derived transport address from a TURN
server providing public IP addresses.
5.3.4 Sending STUN Connectivity Checks
Once a responder has received an initiate message, or an initiator
has received an accept message, the list of transport addresses is
extracted from the message. These transport addresses, called the
remote transport addresses, along with the username fragment,
password, and priority from the message are placed into a table,
called the candidate table. There is a candidate table for RTP for
each media stream, and for RTCP for each media stream. So, if a
session is established with audio and video, there would be four
tables - audio RTP, audio RTCP, video RTP and video RTCP.
The client then takes its own gathered addresses, and creates a
subset called the sourceable addresses. This subset is the set of
local transport addresses (including VPN and RSIP) and TURN derived
transport addresses. Thus, it excludes STUN derived transport
addresses. The formal definition of this subset is defined below.
Each row in this table is then replicated once for each sourceable
transport address. The table has a column for the sourceable
transport address value, and this is populated upon replication.
That table also has a column called "my username fragment", which is
the username fragment that the client created for sourceable
transport address in that row. Each row in this table is called a
candidate.
Each candidate is associated with a state. The state represents the
current understanding of connectivity to that remote transport
address when packets are sent from that sourceable address. There
are five possible states. These states are:
INIT: No STUN transaction has been completed towards this remote
transport address from this sourceable address.
HANDSHAKING: One or more STUN transactions have failed, but
insufficient time has passed since leaving the INIT state to be
certain that the remote transport address is unreachable from this
sourceable address. This state is important for connectivity
checks made to STUN derived transport addresses through port
restricted NAT or a TURN derived transport address.
BAD: All STUN transactions to this remote transport address from this
sourceable address have either timed out, or failed with a 600
response, and a sufficient amount of time has elapsed since the
INIT state to have high confidence that the remote transport
address cannot be reached from this sourceable address.
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GOOD: The last STUN transaction to this remote transport address from
this sourceable address was successful. However, it is not the
highest priority candidate, and therefore, is not in use for
media.
When the client first populates the tables from the initiate or
accept message, all of the transport addresses are set to the INIT
state.
Consider the the following example. An initiator sends an initiate
message with one media stream (audio), with two transport addresses,
A and B. A is a local transport address, and B is a STUN derived
transport address (although that fact is not signaled in the
message). Both of these will have the same username fragment and
password, but different priorities. The initiate message is sent to
the responder. The responder has a local transport address, a STUN
derived transport address, and a TURN derived transport address.
Call these X, Y and Z respectively. Thus, it has two sourceable
addresses, X and Z. The table created by the responder would have
four rows. Each of the two transport addresses in the initiate
message is present twice, once with the responder's local transport
address, and once with its TURN derived address. Such a table might
look like this:
Remote Srcable User Frag Passwd My-Usr-Frag Priority State
--------------------------------------------------------------------
A X asd9f8f8== siprulz x-frag 0.4 INIT
A Z asd9f8f8== siprulz z-frag 0.4 INIT
B X asd9f8f8== siprulz x-frag 0.2 INIT
B Z asd9f8f8== siprulz z-frag 0.2 INIT
The client begins a STUN BindingRequest transaction for each
candidate. This STUN transaction is sent to the IP address and port
from the Remote column. It sends the request from the IP address and
port in the sourceable column. For local transport addresses, that
means sending from the locally bound socket. For VPN addresses, that
means sending from the socket bound to the VPN interface. For TURN
derived transport addresses, this means using the TURN Send message
to send a request through the TURN server. This provides the
definition of the sourceable flag: they represent distinct transport
addresses that a client can send from. A STUN derived transport
address is not distinct from a local transport address, since a
client cannot send a packet to a particular IP address and port with
different source IP addresses and ports as seen by that recipient
[[REPHRASE]]
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The STUN USERNAME attribute MUST be present. It is set to the
concatenation of the user fragment from the table, with the "My User
Fragment" from the candidate. Thus, for the candidate with remote
transport address A and sourceable address X, the USERNAME would be
set to "asd9f8f8==x-frag". The BindingRequest SHOULD contain a
MESSAGE-INTEGRITY attribute, computed using the username in the
USERNAME attribute, and the password from the password field in the
row. The BindingRequest MUST NOT contain the CHANGE-REQUEST or
RESPONSE-ADDRESS attribute.
Each of these STUN transactions will generate either a timeout, or a
response. If the response is an error, but recoverable as described
in RFC 3489, the client SHOULD try again using the procedures
discussed there. Either initialy, or after retry, the STUN
transaction will produce a timeout result, a success result, or a
non-recoverable failure result (error codes 400, 431, or 600). These
correspond to "timeout", "success", and "error" events, respectively.
These events are fed into the state machine described in Figure 3.
This figure shows the transitions between states that occur on the
completion of the STUN BindingRequest transaction. After the
completion of each transaction, the client sets a timer that
determines when it will do another transaction for that candidate.
The result of that next transaction drives the next transition in the
state machine, and so on. Since timers are set at the entry to each
state, STUN BindingRequest tranasactions will be tried continuously
throughout a call. This is necessary to detect a variety of failure
cases, as discussed below.
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..........
. . timeout/
. . Set Rapid
+---------+ +---------+ . Retry Timer
| | | | .
| | | |<....
| INIT |......................>| HAND |
| | timeout/ | SHAKING |
| | Set Rapid | |
+---------+ Retry Timer, error/ +---------+
. . Giveup Timer Set . .
. . Retry . .
error/ . . Timer . .
Set . . ............................. . success/
Retry . . . . Set Refresh
Timer . ...C.............................. . Timer
. . success/ . .
. . Set Refresh . .
V V Timer V V
+---------+ +---------+
| | | |
| | | |
| BAD |......................>| GOOD |
...>| | success/ | |.......
. | | Set Refresh | | .
. +---------+ Timer +---------+ .
. . ^ . ^ .
. . . . . .
....... . . ..........
timeout or ................................ success/
error/ timeout or Set Refresh
Set error/ Timer
Retry Set
Timer Retry
Timer
Figure 3
Starting in the INIT state, if the transaction is successful, the
client has verified connectivity to that remote transport address
when sending from that sourceable transport address. This means that
media packets sent in exactly the same way will get through. As
such, the FSM transitions to the GOOD state, and the client sets the
Refresh Timer. This timer is used to continually check that a good
candidate remains good. It is possible for a candidate to cease
being good if a NAT should fail and recover, resulting in loss of any
bindings it holds, or if an IP route should flap, causing those
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packets to be delivered through a new NAT that allocates new
bindings, or a firewall with different policies. The Retry Timer
value SHOULD be configurable. In order to rapidly recover from
failures, it is RECOMMENDED that it default to five seconds. [[TODO:
Need to work this number as a function of codec rates as well,
perhaps apply the RTCP algorithm for its computation.]]
If, from the INIT state, the STUN transaction times out, the FSM
enters the HANDSHAKE state. At this point, there are two reasons
that the STUN request might have timed out. One reason is that the
candidate is simply unreachable. The other reason is that the peer
is behind a port restricted NAT, and so STUN requests from the client
cannot get through until its peer creates a permission by generating
its own STUN request. It may take some time to generate that STUN
request, as it may depend on a response message getting delivered.
As such, the HANDSHAKE state allows for rapid retry of the STUN
transaction until enough time has passed to be certain that the
remote transport address is actually unreachable. Thus, upon
entering the HANDSHAKE state, two timers are set. The first, called
the Rapid Retry timer, determines how long until the next attempt.
This timer SHOULD be configurable. It is RECOMMENDED that it default
to 1 second. The second timer, called the Giveup Timer, determines
how long the client will keep trying until it decides that the remote
transport address is unreachable. This timer SHOULD be configurable.
It is RECOMMENDED that it default to 50 seconds. This is a
reasonable approximation of the maximum SIP transaction duration.
If, from the INIT state, the STUN transaction generates an error, the
FSM moves into the BAD state. The retry timer is set. This retry
timer is used to periodically retry, and see if the candidate may now
be reachable. The value of this timer SHOULD be configurable. It is
RECOMMENDED that it default to 1 minute.
If, while in the HANDSHAKE state, the Giveup timer fires, or the STUN
transaction results in an error, the client moves into the BAD state,
and sets the retry timer. The default durations for ths timer are
identical for all entries into the BAD state, and thus it defaults to
1 minute here as well. If, while in the HANDSHAKE state, the Rapid
Retry timer fires, the timer is reset and the client remains in the
HANDSHAKE state.
If, while in the BAD state, the retried transaction is executed and
fails or results in a timeout, the client resets the timer and
remains in the BAD state. If the STUN transaction succeeds, it moves
into the GOOD state and sets the refresh timer. The default
durations for this timer are the same for all entries into the GOOD
state, and thus it defaults to 1 second.
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If while in the GOOD state, the transaction resulting from the
refresh timer times out or fails, the client moves into the BAD state
and sets the retry timer. If, however, that transaction succeeds,
the client stays in the GOOD state and resets the refresh timer.
As the FSM operates throughout the call, candidates will move their
states around. At any point in time, the client sends media packets
(including RTCP) using one of the candidates in the GOOD state. It
is RECOMMENDED that the one with highest priority be used. It
another candidate should change state such that it moves into the
GOOD state, and it has a higher priority, the client SHOULD switch to
that candidate, but SHOULD do so after waiting a small period of time
(10 seconds is RECOMMENDED) to prevent against flapping of candidates
during periods of route flaps in the network.
To send media to a candidate, the client sends media packets (whether
they are RTP or RTCP or something else) to the remote transport
address, from the sourceable transport address.
If, for some reason, there was at least one candidate in the GOOD
state, and due to an FSM transition, none of the candidates are in
the GOOD state, the client SHOULD forcefully transition all of the
candidates into the HANDSHAKE state in an attempt to rapidly
reconnect. If none of them succeed, and all of the candidates enter
the BAD state, the client SHOULD terminate the call and alert the
user to the failure [[TODO: Need to work in some good congestion
control here; in cases where timeouts happen due to network
congestion this is probably too agressive]].
5.3.5 Receiving STUN Requests
When a client receives a STUN request (presumably after
disambiguating it from a media packet), it follows the logic
described in this section.
The client MUST follow the procedures defined in RFC 3489 and verify
that the USERNAME attribute is known to the server. Here, this is
done by taking the USERNAME attribute, and doing a prefix match
against the "my user fragment" column in the candidate table. If it
doesn't match any rows, the client generates a 432 response. If it
matches multiple rows, the client checks the suffix of the username
against the "user fragment" column. If it doesn't match any rows,
the client generates a 432 response. If it does match rows, it will
match those rows corresponding to the transport addresses that the
peer could have sent this STUN request from.
Assuming the USERNAME is valid, the client MUST generate a STUN
response per RFC 3489.
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Once the response is sent, the client examines the source IP and port
where the request came from. It matches those against the remote
transport addresses in the candidate table. If there is no match,
this source address is itself another possible candidate. As with
other candidates, it must be associated with a STUN username
fragment, password and priority, all normally provided by the peer,
along with sourceable transport addresses and their username
fragments.
How does the client obtain this other information? The suffix of the
USERNAME is the key (literally). That suffix was already provided to
the client in an initiate or accept message, and was used to populate
the current candidate table. If it matches an existing value in the
table, it means that the STUN request came from the same transport
address as a previously advertised candidate; however, when it showed
up at the client, its source IP address was different than the peer
thought it would be. This will happen when a symmetric NAT exists
between the clients. In this case, the source IP address and port of
the STUN packet now become a viable candidate, since the client
should be able to send messages back to it and reach its peer.
However, this connectivity, like all other connectivity, needs to be
verified. So, the client needs to find out the user fragment and
password to use in STUN requests. To do that, it takes the suffix of
the USERNAME in the STUN request, and looks it up in the "user frag"
column of the table. If its a match, that is the user fragment
needed as part of the candidate. The password is the value from that
row. The sourceable transport address is also the value from that
row. The priority is also copied from that row.
This new candidate can then be verified by sending STUN requests to
it, as described in Section 5.3.4.
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6. Running STUN on Derived Transport Addresses
One of the seemingly bizarre operations done during the ICE
processing is the transmission of a STUN request to a transport
address which is obtained through TURN or STUN itself. This actually
does work, and in fact, has extremely useful properties. The
subsections below go through the detailed operations that would occur
at each point to demonstrate correctness and the properties derived
from it.
6.1 STUN on a TURN Derived Transport Address
Consider a client A that is behind a NAT. It connects to a TURN
server on the public side of the NAT. To do that, A binds to a local
transport address, say 10.0.1.1:8866, and then sends a TURN request
to the TURN server. The NAT translates the net-10 address to
192.0.2.88:5063. Assume that the TURN server is running on 192.0.2.1
and listening for TURN traffic on port 7764. The TURN server
allocates a derived transport address 192.0.2.1:26524 to the client,
and returns it in the TURN response. Remember that all traffic from
the TURN server to the client is sent from 192.0.2.1:7764 to
10.0.1.1:8866.
Now, the client runs a STUN server on 10.0.1.1:8866, and advertises
that its server actually runs on 192.0.2.1:26524. Another client, B,
sends a STUN request to this server. It sends it from a local
transport address, 192.0.2.77:1296. When it arrives at
192.0.2.1:26524, the TURN server "locks down" outgoing traffic, so
that data packets received from A are sent to 192.0.2.77:1296. The
STUN request is then forwarded to the client, sent with a source
address of 192.0.2.1:7764 and a destination address of
192.0.2.88:5063. This passes through the NAT, which rewrites the
source address to 10.0.1.1:8866. This arrives at A's STUN server.
The server observes the source address of 192.0.2.1:7764, and
generates a STUN response containing this value in the MAPPED-ADDRESS
attribute. The STUN response is sent with a source address fo
10.0.1.1:8866, and a destination of 192.0.2.1:7764. This arrives at
the TURN server, which, because of the lock-down, sends the STUN
response with a source address of 192.0.2.1:26524 and destination of
192.0.2.77:1296, which is B's STUN client.
Now, as far as B is concerned, it has obtained a new STUN derived
transport address of 192.0.2.1:7764. And indeed, it has! STUN
derived transport addresses are scoped to the session, so they can
only be used by the peer in the session. Furthermore, that peer has
to send requests from the socket on which the STUN server was
running. In this case, A is the peer, and its STUN server was on
10.0.1.1:8866. If it sends to 192.0.2.1:7764, the packet goes to the
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TURN server, and due to lock-down, is forwarded to B, and
specifically, is forwarded to the transport address B sent the STUN
request from. Therefore, the address is indeed a valid STUN derived
transport address.
The benefit of this is that it allows two clients to share the same
TURN server for media traffic in both directions. With "normal" TURN
usage, both clients would obtain a derived address from their own
TURN servers. The result is that, for a single call, there are two
bindings allocated by each side from their respective servers, and
all four are used. With ICE, that drops to two bindings allocated
from a single server. Of course, all four bindings are allocated
initially. However, once one of the clients begins receiving media
on its STUN derived address, it can deallocate its TURN resources.
[[TODO: Include a diagram that shows this pictorially.]]
6.2 STUN on a STUN Derived Transport Address
Consider a client A that is behind a NAT. It connects to a STUN
server on the public side of the NAT. To do that, A binds to a local
transport address, say 10.0.1.1:8866, and then sends a STUN request
to the STUN server. The NAT translates the net-10 address to
192.0.2.88:5063. Assume that the STUN server is running on 192.0.2.1
and listening for STUN traffic on port 3478, the default STUN port.
The STUN server sees a source IP address of 192.0.2.88:5063, and
returns that to the client in the STUN response. The NAT forwards
the response to the client.
Now, the client runs a STUN server on 10.0.1.1:8866, and advertises
that its server actually runs on 192.0.2.88:5063. Another client, B,
sends a STUN request to this address. It sends it from a local
transport address, 192.0.2.77:1296. When it arrives at
192.0.2.88:5063 (on the NAT), the NAT rewrites the source address to
10.0.1.1:8866, assuming that it is of the full-cone variety [1], or
is restricted, and the permission for 192.0.2.77:1296 is open. This
arrives at A's STUN server. The server observes the source address
of 192.0.2.77:1296, and generates a STUN response containing this
value in the MAPPED-ADDRESS attribute. The STUN response is sent
with a source address of 10.0.1.1:8866, and a destination of
192.0.2.77:1296. This arrives at B's STUN client.
Now, as far as B is concerned, it has obtained a new STUN derived
transport address of 192.0.2.77:1296. Of course, this is the same
address as the local transport address, and therefore this derived
address is not used. However, had there been additonal NATs between
B and A's NAT, B would end up seeing the binding allocated by that
outermost NAT. The net result is that STUN requests sent to a STUN
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derived address behave as normal STUN would. However, these STUN
requests have the side-effect of creating permissions in the NATs
which see those requests in the public to private direction. This
turns out to be very useful for traversing restricted NATs.
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7. XML Schema for ICE Messages
This section contains the XML schema used to define the initiate,
accept, and modify messages. Any protocol that uses ICE needs to map
the parameters defined here into its own messages.
Note that STUN allows both the username and password to contain the
space character. However, usernames and passwords used with ICE
cannot contain the space.
<?xml version="1.0" encoding="UTF-8"?>
<xs:schema targetNamespace="urn:ietf:params:xml:ns:ice"
xmlns:xs="http://www.w3.org/2001/XMLSchema"
xmlns:tns="urn:ietf:params:xml:ns:ice"
elementFormDefault="qualified" attributeFormDefault="unqualified">
<xs:import namespace="http://www.w3.org/XML/1998/namespace"
schemaLocation="http://www.w3.org/2001/xml.xsd"/>
<xs:element name="message" type="tns:message"/>
<xs:complexType name="message">
<xs:annotation>
<xs:documentation>This is the root element, which holds a
media-streams elements.</xs:documentation>
</xs:annotation>
<xs:sequence>
<xs:element name="media-streams" type="tns:media-streams"/>
</xs:sequence>
<xs:attribute name="type" type="tns:msg-type" use="required"/>
</xs:complexType>
<xs:complexType name="media-streams">
<xs:sequence>
<xs:element name="media-stream" minOccurs="0" maxOccurs="unbounded">
<xs:annotation>
<xs:documentation>There are zero or more media stream
elements. Each defines attributes for a specific media
stream.</xs:documentation>
</xs:annotation>
<xs:complexType>
<xs:sequence>
<xs:element name="default-address">
<xs:annotation>
<xs:documentation>The default address is used for
sending media before connectivity has been
verified.</xs:documentation>
</xs:annotation>
<xs:complexType>
<xs:complexContent>
<xs:extension base="tns:rtp-info"/>
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</xs:complexContent>
</xs:complexType>
</xs:element>
<xs:sequence>
<xs:element name="alternate" minOccurs="0" maxOccurs="unbounded">
<xs:annotation>
<xs:documentation>Each alternate is a
possible point of contact.
</xs:documentation>
</xs:annotation>
<xs:complexType>
<xs:complexContent>
<xs:extension base="tns:transport-data">
<xs:attribute name="preference" type="xs:double" use="required"/>
<xs:attribute name="id" type="xs:string" use="required"/>
</xs:extension>
</xs:complexContent>
</xs:complexType>
</xs:element>
</xs:sequence>
</xs:sequence>
</xs:complexType>
</xs:element>
</xs:sequence>
</xs:complexType>
<xs:simpleType name="msg-type">
<xs:restriction base="xs:string">
<xs:enumeration value="initiate"/>
<xs:enumeration value="accept"/>
<xs:enumeration value="modify"/>
</xs:restriction>
</xs:simpleType>
<xs:complexType name="transport-data">
<xs:sequence>
<xs:element name="stun-user-fragment" type="xs:string"/>
<xs:element name="stun-password" type="xs:string"/>
<xs:element name="rtp-address" type="tns:transport-address"/>
<xs:element name="rtcp-address" type="tns:transport-address"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="transport-address">
<xs:sequence>
<xs:element name="ip-address" type="xs:string"/>
<xs:element name="port">
<xs:simpleType>
<xs:restriction base="xs:integer">
<xs:minInclusive value="1"/>
<xs:maxInclusive value="65535"/>
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</xs:restriction>
</xs:simpleType>
</xs:element>
</xs:sequence>
</xs:complexType>
<xs:complexType name="rtp-info">
<xs:sequence>
<xs:element name="rtp-address" type="tns:transport-address"/>
<xs:element name="rtcp-address" type="tns:transport-address"/>
</xs:sequence>
</xs:complexType>
</xs:schema>
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8. Examples
In the examples that follow, messages are labeled with "message name
A,B" to mean a message from transport address A to B. For STUN
Requests, this is followed by curly brackets enclosing the username
and password. For STUN responses, this is followed by square
brackets and the value of MAPPED ADDRESS.
8.1 Port Restricted
This section shows a flow of two clients behind port restricted NAT
talking to each other.
A P.R. NAT STUN+TURN P.R. NAT B
|(1) STUN Req P1,S+T | | |
|----------->| | | |
| |(2) STUN Req U, S+T | |
| |----------->| | |
| |(3) STUN Res S+T,U [U] | |
| |<-----------| | |
|(4) STUN Res S+T,P1 [U] | | |
|<-----------| | | |
|(5) Intitiate {P1,unameA,passA,q=0.4} | |
|{U,unameA,passA,q=0.3} | | |
|-------------------------------------------------->|
| | | |(6) STUN Req P2,S+T
| | | |<-----------|
| | |(7) STUN Req V, S+T |
| | |<-----------| |
| | |(8) STUN Res S+T,V [V] |
| | |----------->| |
| | | |(9) STUN Res S+T,P2 [V]
| | | |----------->|
|(10) Accept {P2,unameB,passB,q=0.4} | |
|{V,unameB,passB,q=0.3} | | |
|<--------------------------------------------------|
|(11) STUN Req P1,P2 | | |
|(unameBunameA,passB) | | |
|----------->| | | |
| |Timeout | | |
|(12) STUN Req P1,V | | |
|(unameBunameA,passB) | | |
|----------->| | | |
| |(13) STUN Req U,V | |
| |(unameBunameA,passB) | |
| |------------------------>| |
| |Permission open V->U | |
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| | | |No success, Retries continue
| | | |(14) STUN Req P2,P1
| | | |(unameAunameB,passA)
| | | |<-----------|
| | | |Timeout |
| | | |(15) STUN Req P2,U
| | | |(unameAunameB,passA)
| | | |<-----------|
| |(16) STUN Req V,U | |
| |(unameAunameB,passA) | |
| |<------------------------| |
| | | |Permission open U->V
| |Passes NAT! | | |
|(17) STUN Req V,P1 | | |
|(unameAunameB,passA) | | |
|<-----------| | | |
|(18) STUN Res P1,V [V] | | |
|----------->| | | |
| |(19) STUN Res U,V [V] | |
| |------------------------>| |
| | | |(20) STUN Res U,P2 [V]
| | | |----------->|
| |Retries continue | |
|(21) STUN Req P1,V | | |
|(unameBunameA,passB) | | |
|----------->| | | |
| |(22) STUN Req U,V | |
| |(unameBunameA,passB) | |
| |------------------------>| |
| | | |Passes NAT! |
| | | |(23) STUN Req U,P2
| | | |(unameBunameA,passB)
| | | |----------->|
| | | |(24) STUN Res P2,U [U]
| | | |<-----------|
| |(25) STUN Res V,U [U] | |
| |<------------------------| |
|(26) STUN Res V,P1 [U] | | |
|<-----------| | | |
|(27) RTP P1,V | | |
|----------->| | | |
| |(28) RTP U,V| | |
| |------------------------>| |
| | | |Passes NAT! |
| | | |(29) RTP U,P2
| | | |----------->|
| | | |(30) RTP P2,U
| | | |<-----------|
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| |(31) RTP V,U| | |
| |<------------------------| |
| |Passes NAT! | | |
|(32) RTP V,P1 | | |
|<-----------| | | |
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9. Mapping ICE into SIP
In this section, we show how to map ICE into SIP. This requires
extensions to SDP.
A new SDP attribute is defined to support ICE. It is called "alt".
The alt attribute MUST be present within a media block of the SDP.
It contains an alternative IP address and port (or pair of IP
addresses and ports in the case of RTP) that the recipient of the SDP
can use instead of the ones indicated in the m and c lines. There
MAY be multiple alt attributes in a media block. In that case, each
of them MUST contain a different IP address and port (or a differing
pair of IP address and ports in the case of RTP).
The syntax of this attribute is:
alt-attribute = "alt" ":" id SP qvalue SP
username SP password SP
unicast-address SP port [unicast-address SP port]
;qvalue from RFC 3261
;unicast-address, port from RFC 2327
username = non-ws-string
password = non-ws-string
id = token
derived-from = ":" / id
With the addition of the alt attribute, the mapping of the ICE
messages to SIP/SDP is straightforward. The ICE initiate message
corresponds to a SIP message with an SDP offer. The ICE accept
message corresponds to a SIP message with a SDP answer. The ICE
modify message corresponds to a SIP INVITE or UPDATE with an offer,
and the ICE modify accept message corresponds to an INVITE or UPDATE
response with an answer.
Each media stream element in an ICE message maps to a media block in
the SDP. The default address maps to the m and c lines in the SDP.
If the ICE message indicates an RTCP address and port that are not
one higher than that of the RTP, the SDP RTCP attribute [2] MUST be
used to convey them.
Each alternate element in an ICE message maps either to an alt
attribute in the SDP, or a new media block, depending on the IP
version of the alternate. For the highest priority IPv6 alternate,
it is mapped into a separate media block, using the ANAT grouping
[4]. Any additional IPv6 addresses are placed as alternates within
this media block. For alternates that are IPv4 addresses, the alt
attribute is used. The rtp-address element maps to the first
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unicast-address and port components of the alt attribute. The
rtcp-address element maps to the second unicast-address and port
components of the alt attribute. Note that, if the RTCP address is
identical to the RTP address, and the port is one higher, the second
unicast-address and port MAY be omitted. The preference value from
the alternate element is mapped to the q-value component of the alt
attribute. The STUN user fragment and password elements map to the
user fragment and password components of the alt attribute.
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10. Security Considerations
ICE conveys the STUN username and password within its messages. If
an eavesdropper should see the username and password, the worst they
can do is send STUN requests to the host. Since STUN is a stateless
protocol, the attacker can not alter the processing of the call or
otherwise disrupt it. They could flood the server with
BindingRequest packets. However, this would be no worse than if the
attacker simply floods the host with any kind of packet.
However, integrity protection of the username and password are more
important. If an attacker is capable of intercepting the message and
modifying the username or password, they could prevent connectivity
from being established between peers, and therefore disrupt the call.
Of course, if the attacker can intercept the message, there are many
other ways in which they could do that, such as simply discarding the
message. Injecting fake messages with incorect usernames and
passwords can also disrupt a call, and does not require the
compromise of an intermediate server. A similar attack is possible
by modifying most of the ICE message attributes. To prevent these
kinds of attacks, it is RECOMMENDED that the actual protocols the ICE
maps to make use of security mechanisms that provide message
integrity protection.
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11. IANA Considerations
This specification defines one new media attribute: alt. Its syntax
is defined in Section 9.
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12. IAB Considerations
The IAB has studied the problem of "Unilateral Self Address Fixing",
which is the general process by which a client attempts to determine
its address in another realm on the other side of a NAT through a
collaborative protocol reflection mechanism [11]. ICE is an example
of a protocol that performs this type of function. Interestingly,
the process for ICE is not unilateral, but bilateral, and the
difference has a signficant impact on the issues raised by IAB. The
IAB has mandated that any protocols developed for this purpose
document a specific set of considerations. This section meets those
requirements.
12.1 Problem Definition
From RFC 3424 any UNSAF proposal must provide:
Precise definition of a specific, limited-scope problem that is to
be solved with the UNSAF proposal. A short term fix should not be
generalized to solve other problems; this is why "short term
fixes usually aren't".
The specific problems being solved by ICE are:
Provide a means for two peers to determine the set of transport
addresses which can be used for communication.
Provide a means for resolving many of the limitations of other
UNSAF mechanisms by wrapping them in an additional layer of
processing (the ICE methodology).
Provide a means for a client to determine an address that is
reachable by another peer with which it wishes to communicate.
12.2 Exit Strategy
From RFC 3424, any UNSAF proposal must provide:
Description of an exit strategy/transition plan. The better short
term fixes are the ones that will naturally see less and less use
as the appropriate technology is deployed.
ICE itself doesn't easily get phased out. However, it is useful even
in a globally connected Internet, to serve as a means for detecting
whether a router failure has temporarily disrupted connectivity, for
example. However, what ICE does is help phase out other UNSAF
mechanisms. ICE effectively selects amongst those mechanisms,
prioritizing ones that are better, and deprioritizing ones that are
worse. Local IPv6 addresses are always the most preferred. As NATs
begin to dissipate as IPv6 is introduced, derived transport addresses
from other UNSAF mechanisms simply never get used, because higher
priority connectivity exists. Therefore, the servers get used less
and less, and can eventually be remove when their usage goes to zero.
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Indeed, ICE can assist in the transition from IPv4 to IPv6. It can
be used to determine whether to use IPv6 or IPv4 when two dual-stack
hosts communicate with SIP (IPv6 gets used). It can also allow a
client in a v6 island to communicate with a v4 host on the other side
of a 6to4 NAT, by allowing the v6 host to address-fix against the v4
host, and in the process, obtain a v4 address which can be handed to
the v4 client.
12.3 Brittleness Introduced by ICE
From RFC3424, any UNSAF proposal must provide:
Discussion of specific issues that may render systems more
"brittle". For example, approaches that involve using data at
multiple network layers create more dependencies, increase
debugging challenges, and make it harder to transition.
ICE actually removes brittleness from existing UNSAF mechanisms. In
particular, traditional STUN (the usage described in RFC 3489) has
several points of brittleness. One of them is the discovery process
which requires a client to try and classify the type of NAT it is
behind. This process is error-prone. With ICE, that discovery
process is simply not used. Rather than unilaterally assessing the
validity of the address, its validity is dynamically determined by
measuring connectivity to a peer. The process of determining
connectivity is very robust. The only potential problem is that
bilaterally fixed addresses through STUN can expire if traffic does
not keep them alive. However, that is substantially less brittleness
than the STUN discovery mechanisms.
Another point of brittleness in STUN, TURN, and any other unilateral
mechanism is its absolute reliance on an additional server. ICE
makes use of a server for allocating unilateral addresses, but allows
clients to directly connect if possible. Therefore, in some cases,
the failure of a STUN or TURN server would still allow for a call to
progress when ICE is used.
Another point of brittleness in traditional STUN is that it assumes
that the STUN server is on the public Internet. Interestingly, with
ICE, that is not necessary. There can be a multitude of STUN servers
in a variety of address realms. ICE will discover the one that has
provided a usable address.
The most troubling point of brittleness in traditional STUN is that
it doesn't work in all network topologies. In cases where there is a
shared NAT between each client and the STUN server, traditional STUN
may not work. With ICE, that restriction can be lifted.
Traditional STUN also introduces some security considerations.
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Unfortunately, since ICE still uses network resident STUN servers,
those security considerations still exist.
12.4 Requirements for a Long Term Solution
From RFC 3424, any UNSAF proposal must provide:
Identify requirements for longer term, sound technical solutions
-- contribute to the process of finding the right longer term
solution.
Our conclusions from STUN remain unchanged. However, we feel ICE
actually helps because we believe it can be part of the long term
solution.
12.5 Issues with Existing NAPT Boxes
From RFC 3424, any UNSAF proposal must provide:
Discussion of the impact of the noted practical issues with
existing, deployed NA[P]Ts and experience reports.
A number of NAT boxes are now being deployed into the market which
try and provide "generic" ALG functionality. These generic ALGs hunt
for IP addresses, either in text or binary form within a packet, and
rewrite them if they match a binding. This will interfere with
proper operation of any UNSAF mechanism, including ICE.
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13. Acknowledgements
The authors would like to thank Douglas Otis and Francois Audet for
their comments and input.
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14. References
14.1 Normative References
[1] Rosenberg, J., Weinberger, J., Huitema, C. and R. Mahy, "STUN -
Simple Traversal of User Datagram Protocol (UDP) Through Network
Address Translators (NATs)", RFC 3489, March 2003.
[2] Huitema, C., "Real Time Control Protocol (RTCP) attribute in
Session Description Protocol (SDP)", RFC 3605, October 2003.
[3] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[4] Camarillo, G., "The Alternative Network Address Types Semantics
for the Session Description Protocol Grouping Framework",
draft-ietf-mmusic-anat-01 (work in progress), June 2004.
14.2 Informative References
[5] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
[6] Senie, D., "Network Address Translator (NAT)-Friendly
Application Design Guidelines", RFC 3235, January 2002.
[7] Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A. and A.
Rayhan, "Middlebox communication architecture and framework",
RFC 3303, August 2002.
[8] Borella, M., Lo, J., Grabelsky, D. and G. Montenegro, "Realm
Specific IP: Framework", RFC 3102, October 2001.
[9] Borella, M., Grabelsky, D., Lo, J. and K. Taniguchi, "Realm
Specific IP: Protocol Specification", RFC 3103, October 2001.
[10] Yon, D., "Connection-Oriented Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-comedia-07
(work in progress), June 2004.
[11] Daigle, L. and IAB, "IAB Considerations for UNilateral
Self-Address Fixing (UNSAF) Across Network Address
Translation", RFC 3424, November 2002.
[12] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", RFC
3550, July 2003.
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[13] Baugher, M., McGrew, D., Naslund, M., Carrara, E. and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
[14] Carpenter, B. and K. Moore, "Connection of IPv6 Domains via
IPv4 Clouds", RFC 3056, February 2001.
[15] Huitema, C., "Teredo: Tunneling IPv6 over UDP through NATs",
draft-huitema-v6ops-teredo-02 (work in progress), June 2004.
[16] Rosenberg, J., "Traversal Using Relay NAT (TURN)",
draft-rosenberg-midcom-turn-04 (work in progress), February
2004.
Author's Address
Jonathan Rosenberg
dynamicsoft
600 Lanidex Plaza
Parsippany, NJ 07054
US
Phone: +1 973 952-5000
EMail: jdrosen@dynamicsoft.com
URI: http://www.jdrosen.net
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