dispatch J. Rosenberg
Internet-Draft jdrosen.net
Intended status: Standards Track C. Jennings
Expires: May 13, 2010 Cisco
November 9, 2009
Verification Involving PSTN Reachability: Requirements and Architecture
Overview
draft-rosenberg-dispatch-vipr-overview-01
Abstract
The Session Initiation Protocol (SIP) has seen widespread deployment
within individual domains, typically supporting voice and video
communications. Though it was designed from the outset to support
inter-domain federation over the public Internet, such federation has
not materialized. The primary reasons for this are the complexities
of inter-domain phone number routing and concerns over security.
This document reviews this problem space, outlines requirements, and
then describes a new model and technique for inter-domain federation
with SIP, called Verification Involving PSTN Reachability (ViPR).
ViPR addresses the problems that have prevented inter-domain
federation over the Internet. It provides fully distributed inter-
domain routing for phone numbers, authorized mappings from phone
numbers to domains, a new technique for automated VOIP anti-spam, and
privacy of number ownership, all while preserving the trapezoidal
model of SIP.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 5
2.1. The Phone Number Routing Problem . . . . . . . . . . . . . 6
2.2. The Open Pinhole Problem . . . . . . . . . . . . . . . . . 7
2.3. Quality of Service Problem . . . . . . . . . . . . . . . . 7
2.4. Troubleshooting Problem . . . . . . . . . . . . . . . . . 8
3. Summary of Existing Solutions . . . . . . . . . . . . . . . . 8
3.1. Domain Routing . . . . . . . . . . . . . . . . . . . . . . 8
3.2. Public ENUM . . . . . . . . . . . . . . . . . . . . . . . 9
3.3. Private Federations . . . . . . . . . . . . . . . . . . . 9
4. Key Requirements . . . . . . . . . . . . . . . . . . . . . . . 10
5. Executive Overview . . . . . . . . . . . . . . . . . . . . . . 11
5.1. Key Properties . . . . . . . . . . . . . . . . . . . . . . 11
5.2. Challenging Past Assumptions . . . . . . . . . . . . . . . 13
5.3. Technical Overview . . . . . . . . . . . . . . . . . . . . 14
5.3.1. Storage of Phone Numbers . . . . . . . . . . . . . . . 16
5.3.2. PSTN First Call . . . . . . . . . . . . . . . . . . . 17
5.3.3. Validation and Caching . . . . . . . . . . . . . . . . 18
5.3.4. SIP Call . . . . . . . . . . . . . . . . . . . . . . . 19
6. Architecture Components and Functions . . . . . . . . . . . . 24
6.1. ViPR Server . . . . . . . . . . . . . . . . . . . . . . . 25
6.2. Call Agent . . . . . . . . . . . . . . . . . . . . . . . . 26
6.3. Border Element . . . . . . . . . . . . . . . . . . . . . . 27
6.4. Enrollment Server . . . . . . . . . . . . . . . . . . . . 28
6.5. P2P Network . . . . . . . . . . . . . . . . . . . . . . . 28
7. Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . 28
7.1. P2P: RELOAD . . . . . . . . . . . . . . . . . . . . . . . 28
7.1.1. ViPR Usage . . . . . . . . . . . . . . . . . . . . . . 29
7.1.2. Certificate Usage . . . . . . . . . . . . . . . . . . 30
7.2. ViPR Access Protocol (VAP) . . . . . . . . . . . . . . . . 30
7.3. Validation Protocol . . . . . . . . . . . . . . . . . . . 31
7.4. SIP Extensions . . . . . . . . . . . . . . . . . . . . . . 32
8. Example Call Flows . . . . . . . . . . . . . . . . . . . . . . 32
8.1. PSTN Call and VCR Upload . . . . . . . . . . . . . . . . . 32
8.2. DHT Query and Validation . . . . . . . . . . . . . . . . . 34
8.3. DHT Query and No Match . . . . . . . . . . . . . . . . . . 35
8.4. SIP Call . . . . . . . . . . . . . . . . . . . . . . . . . 35
9. Security Considerations . . . . . . . . . . . . . . . . . . . 36
9.1. Theft of Phone Numbers . . . . . . . . . . . . . . . . . . 37
9.2. Eavesdropping . . . . . . . . . . . . . . . . . . . . . . 37
9.3. Attacks on the DHT . . . . . . . . . . . . . . . . . . . . 37
9.4. Spam . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 38
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 39
11.1. Normative References . . . . . . . . . . . . . . . . . . . 39
11.2. Informative References . . . . . . . . . . . . . . . . . . 39
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Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 40
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1. Introduction
The Session Initiation Protocol (SIP) was originally published as RFC
2543 [RFC2543] in May of 1999. This was followed by subsequent
publication of RFC 3261 [RFC3261], which brought the protocol to
sufficient maturity to enable large scale market adoption.
And indeed, it has seen large scale market adoption. SIP has seen
hundreds of implementations, spanning consumer products, enterprise
servers, and large scale carrier equipment. It carries billions and
billions of minutes of calls, and has become the lingua franca of
interconnection between products from different vendors. If one
measures success in deployment, then clearly SIP is a success.
However, in other ways, it has failed. SIP was designed from the
ground up to enable communications between users in different
domains, all over the public Internet. The intention was that real-
time communications should be no different than email or the web,
with the same any-to-any connectivity that has fueled the successes
of those technologies. Though SIP is used between domains, it is
typically through private federation agreements. The any-to-any
Internet federation model envisioned by SIP has not materialized at
scale.
This document introduces a new technology, called Verification
Involving PSTN Reachability (ViPR), that enables us to break down the
barriers that have prevented inter-domain VoIP. By stepping back and
changing some of the most fundamental assumptions about federation,
ViPR is able to address the key problems preventing its deployment.
ViPR focuses on incremental deployability over the unrealizable
nirvana. At the same time, ViPR ensures that SIP's trapezoidal model
- direct federation between domains without any intermediate
processing beyond IP transport - is realized. That model is required
in order to allow innovative new services to be deployed.
2. Problem Statement
The first question that must be asked is this - why haven't we seen
widespread adoption of inter-domain SIP federation?
There are many reasons for it. They are - in order of importance -
the phone number routing problem, the open pinhole problem, the
quality of service problem, and the troubleshooting problem. The two
former ones are the most significant.
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2.1. The Phone Number Routing Problem
Inter-domain federation requires that the sending domain determine
the address of the receiving domain, in the form of a DNS name
(example.com) or one or more IP addresses that can be used to reach
the domain. In email and in the web, this is easy. The identifiers
used by those services - the email address and web URL respectively -
embed the address of the receiving domain. A simple DNS lookup is
all that is required to route the connection. SIP was designed to
use the same email-style identifiers.
However, most SIP deployments utilize phone numbers, and not email-
style SIP URI. This is due to the huge installed base of users that
continue to exist solely on the public switched telephone network
(PSTN). In order to be reached by users on the PSTN, and in order to
reach them, users in SIP deployments need to be assigned a regular
PSTN number. Users in SIP deployments need to place that PSTN number
on business cards, use it in their email signatures, and in general,
give it out to their friends and colleagues, in order to be reached.
While those users could additionally have an email style SIP URI, the
PSTN number serves as a single, global identifier that works for
receiving calls from users on the PSTN as well as users within the
same SIP domain. Why have two identifiers when one will suffice?
The universality of PSTN numbers is the reason why most SIP
deployments continue to use them - often exclusively.
Another reason is that many SIP deployments utilize hardphones or
telephony adaptors, and the user interfaces on these devices -
patterned after existing phones - only allow phone-number based
dialing. Consequently, these users are only allocated PSTN numbers,
and not email-style SIP URI.
Finally, a large number of SIP deployments are in domains where the
endpoints are not IP. Rather, they are circuit based devices,
connected to a SIP network through a gateway. SIP is used within the
core of the network, providing lower cost transit, or providing
add-on services. Clearly, in these deployments, only phone numbers
are used.
Consequently, to make inter-domain federation incrementally
deployable and widely applicable, it needs to work with PSTN numbers
rather than email-style SIP URI. Telephone numbers, unlike email
addresses, do not provide any indication of the address of the domain
which "owns" the phone number. Indeed, the notion of phone number
ownership is somewhat cloudy. Numbers can be ported between
carriers. They can be assigned to a user or enterprise, and then
later re-assigned to someone else. Numbers are granted to users and
enterprises through a complex delegation process involving the ITU,
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governments, and telecommunications carriers, often involving local
regulations that vary from country to country.
Therefore, in order to deploy inter-domain federation, domains are
required to utilize some kind of mechanism to map phone numbers to
the address of the domain to which calls should be routed. Though
several techniques have been developed to address this issue, none
have achieved large-scale Internet deployments.
2.2. The Open Pinhole Problem
The inter-domain federation mechanism built into SIP borrows heavily
from email. Each domain runs a SIP server on an open port. When one
domain wishes to contact another, it looks up the domain name in the
DNS, and connects to the that server on the open port. Here, "open"
means that the server is reachable from anywhere on the public
Internet, and is not blocked by firewalls.
This simple design worked well in the early days of email. However,
the email system has now become plagued with spam, to the point of
becoming useless. Administrators of SIP domains fear - rightfully so
- that if they make a SIP server available for anyone on the Internet
to contact, it will open the floodgates for VoIP spam, which is far
more disruptive than email-based spam [RFC5039]. Administrators also
worry - rightfully so - that an open server will create a back-door
for denial-of-service and other attacks that can potentially disrupt
their voice service. Administrators are simply not willing to take
that risk; rightly or wrongly, voice deployments demand higher
uptimes and better levels of reliability than email, especially for
enterprises.
Fears around spam and denial-of-service attacks, when put together,
form the "open pinhole problem" - that domains are not willing to
enable SIP on an open port facing the Internet.
To fix this, a new model for federation is needed - a model where
these problems are addressed as part of the fundamental design, and
not as an after-thought.
2.3. Quality of Service Problem
The Internet does not provide any QoS guarantees. All traffic is
best effort. This is not an issue for data transaction services,
like web and email. It is, however, a concern when using real-time
services, such as voice and video.
That said, there are a large number of existing VoIP deployments that
run over the Internet. Though the lack of QoS is a concern, it has
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not proven a barrier to deployment. We believe that, if the more
fundamental issues - the phone number routing and open pinhole
problems - can be addressed, the QoS problem will sort itself out.
As such, we do not discuss this issue further here.
2.4. Troubleshooting Problem
The final problem that is stopping large scale inter-domain
federation is the troubleshooting problem. When connecting calls
between domains, problems will happen. Calls will get blocked.
Calls will get misdelivered. Features won't work. There will be
one-way media or no media at all. The video won't start. Call
quality will be poor.
These problems are common in VoIP deployments, and they are tough to
troubleshoot even within a single administrative domain. When real-
time services extend inter-domain, the problem becomes worse. A new
angle is introduced: the first step is identifying who is at fault.
Fortunately, work is underway to improve the ability for network
administrators to diagnose VoIP problems. Common log formats
[I-D.roach-sipping-clf-syntax] and consistent session IDs
[I-D.kaplan-sip-session-id], for example, can help troubleshoot
interdomain calls.
In addition to these, any new technology that facilitates inter-
domain federation needs to have troubleshooting built-in, so that it
is not a barrier to deployment.
3. Summary of Existing Solutions
Given the value that inter-domain SIP federation brings, it is no
suprise that many attempts have been made at solving it. Indeed,
these have all been deployed to varying degrees. However, all of
them have fundamental limitations that have inhibited widespread
deployment.
3.1. Domain Routing
The first solution that has been proposed for SIP inter-domain
federation is built into SIP itself - domain routing. In this
technique, users utilize email-style SIP URI as identifiers. By
utilizing the DNS lookup mechanism defined in [RFC3263], SIP enables
calls to be routed between domains in much the same way email is
routed between domains.
This technique works well in theory, but it has two limitations which
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have limited its deployment:
1. The majority of SIP deployments utilize phone numbers, often
exclusively. In such a case, domain routing cannot be used.
2. Domain federation brings with it the possibility (and strong
likelihood) of the same levels of spam and DoS attacks that have
plauged the email system.
These issues have already been discussed above.
3.2. Public ENUM
Public ENUM, defined in [RFC3761], tries to address the phone number
routing problem by cleverly placing phone numbers into the public
DNS. Clients can then perform a simple DNS lookup on a phone number,
and retrieve a SIP URI which can be used to route to that phone
number.
Unfortunately, public ENUM requires that the entries placed into the
DNS be populated following a chain of responsibility that mirrors the
ownership of the numbers themselves. This means that, in order for a
number to be placed into the DNS, authorization to do so must start
with the ITU, and from there, move to the country, telecom regulator,
and ultimately the end user. The number of layers of beaurocracy
required to accomplish this is non-trivial. In addition, the telecom
operators - which would be partly responsible for populating the
numbers into the DNS - have little incentive to do so. As a
consequence, public ENUM is largely empty, and is likely to remain so
for the forseeable future.
Instead, ENUM has morphed into a technique for federation amongst
closed peering partners, called private ENUM or infrastructure ENUM
[RFC5067]. While there is value in this technology, it does not
enable the open federation that public ENUM was designed to solve.
It is clear from the legacy of ENUM deployments, that any kind of
phone number routing solution should not rely on government or
telecom processes for population of the databases.
3.3. Private Federations
Private federations are a cooperative formed amongst a small number
of participating domains. The cooperative agrees to use a common
technique for federation, and through it, is able to connect to each
other. There are many such federations in use today.
Some of these federations rely on a central database, typically run
by the federation provider, that can be queried by participating
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domains. The database contains mappings from phone numbers to
domains, and is populated by each of the participating domains, often
manually. Each domain implements an agreed-upon query interface that
can be used to access the database when a number is called.
Sometimes ENUM is used for this interface (called private ENUM),
other times, a SIP redirection is used. Some federations also
utilize private IP networks in order to address QoS problems. "SIP
trunking" - a service being offered by many telecom operators as a
SIP-based PRI replacement - is a form of private federation.
Private federations work, but they have one major limitation: scale.
As the number of participating domains grows, several problems arise.
Firstly, the size of the databases become unruly. Secondly, the
correctness of the database becomes an issue, since the odds of
misconfigured numbers (either intentionally or accidentally)
increases. As the membership grows further, the odds increase that
"bad" domains will be let in, introducing a source of spam and
further problems. The owner of the federation can - and often does -
assume responsibility for this, and can attempt to identify and shut
down misbehaving participants. Indeed, as the size of the
federations grow, the owner of the federation needs to spend
increasing levels of capital on maintaining it. This, in turn,
requires them to charge money for membership, and this can be a
barrier to entry.
4. Key Requirements
From the discussion on the problems of inter-domain federation and
the solutions that have been attempted so far, several key
requirements emerge:
REQ-1: The solution should allow for federation between any number
of domains.
REQ-2: The solution must enable users in one domain to identify
users in another domain through the use of their existing E.164
based phone numbers.
REQ-3: The solution must work with deployments that utilize any kind
of endpoint, including non-IP phones connected through gateways,
IP softphones and hardphones.
REQ-4: The solution should not require any change in user behavior.
The devices and techniques that users have been using previously
to make inter-domain calls should continue to work, but now result
in inter-domain IP federation.
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REQ-5: The solution should work worldwide, for any domain anywhere.
REQ-6: The solution should not require any new services from any
kind of centralized provider. A domain should be able, of its own
free-will and accord, to deploy equipment and connect to the
federation.
REQ-7: The solution should not require any prior arrangement between
domains in order to facilitate federation between those domains.
Federation must occur opportunistically - connections established
when they can be.
REQ-8: The solution must work for domains of any size - starting at
a single phone to the largest telecom operator with tens of
millions of numbers.
REQ-9: The solution must have built-in mechanisms for preventing
spam and DoS attacks. This mechanism must be fully automated.
REQ-10: The solution must not require any processing whatsoever by
SIP or RTP intermediaries. It must be possible for a direct SIP
connection to be established between participating domains.
These requirements, when put together, appear to be mutually
unsolvable. And indeed, they have been - until now.
5. Executive Overview
Verification Involving PSTN Reachability (ViPR) is a new technology
that is aimed at solving the problems that have prevented large-scale
Internet-based SIP federation of voice and video. ViPR solves these
problems by creating a hybrid of three technologies - the PSTN
itself, a P2P network, and SIP. By combining all three, ViPR enables
an incrementally deployable solution to federation.
5.1. Key Properties
ViPR has several important properties that enable it to solve the
federation problem:
Works With Numbers: ViPR enables federation for existing PSTN
numbers. It does not require users or administrators to know or
configure email-style identifiers. It does not require the
allocation of new numbers. It does not require a change in user
behaviors. Whatever way users were dialing numbers yesterday,
works with ViPR tomorrow.
Works with Existing Endpoints: ViPR does not require any changes to
endpoints. Consequently, it works with existing SIP endpoints, or
with non-IP endpoints connected through gateways.
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Fully Distributed: ViPR does not require any kind of central
authority or provider. A domain wishing to utilize ViPR just
deploys it on their own. ViPR utilizes the existing PSTN and
existing Internet connectivity the domain already has, and by
combining them, achieves inter-domain federation. Domains do not
need to wait for their service providers to roll out any kind of
new features, databases, or functionality.
Verified Mappings: The biggest issue in mapping from a phone number
to a domain or IP address, is determining whether the mapping is
correct. Does that domain really own the given phone number?
While solutions like ENUM have solved this problem by relying on
centralized delegations of authorization, ViPR provides a secure
mapping in a fully distributed way. ViPR guarantees that phone
calls cannot be misrouted or numbers stolen.
Worldwide: ViPR works worldwide. Any domain that is connected to
both the PSTN and the Internet can participate. It doesn't matter
whether the domain is in Africa, the Americas, or Australia.
Since ViPR does not depend on availability of any regional
services beyond IP and PSTN access - both of which are already
available globally - ViPR itself is globally available.
Unlimited Scale: ViPR has nearly infinite scale. Any number of
domains can participate.
Self-Scale: ViPR self-scales. This means that the amount of
computation, memory, and bandwith that a domain must deploy scales
in direct proportion to the size of their own user base.
Self-Learning: ViPR is completely automated. A domain never, ever
has to configure any information about another domain. It never
has to provision IP addresses, domain names, certificates, phone
number prefixes or routing rules. Without any prior coordination,
ViPR enables one domain to connect to a different domain.
Automated Anti-Spam ViPR comes with a built-in mechanism for
preventing VoIP spam. This mechanism is new, and specific to
VoIP. In this way, it is fundamentally different from existing
VoIP anti-spam technqiues which borrow from email [RFC5039]. This
new technique is fully automated, and requires no configuration by
administrators and no participation from end users. Though it is
not a 100% solution to the problem, it brings substantial economic
and legal ammunition to the table to act as a good deterrent for a
long while.
Feature Velocity: ViPR enables direct SIP connections between two
domains seeking to federate. There are no SIP intermediaries of
any sort between the two. This means that domains have no
dependencies on intermediaries for deployment of new features.
Designed for the Modern Internet: ViPR is built to run on the modern
Internet. It assumes the worst from everyone. It assumes limited
connectivity. It assumes network failures. It assumes there are
attackers seeking to eavesdrop calls. Security is built-in and
cannot be disabled.
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Reliable: ViPR is reliable. Through its hybridization of the PSTN
and the Internet, it makes sure that calls always go through.
Indeed, to route a call between domains A and B, ViPR never
depends on a server or service anywhere outside of domains A and B
(besides vanilla PSTN and IP access) being operational.
At first glance, these properties seem impossible to realize. And
indeed, given the assumptions that have traditionally been made about
how federation has to work, these properties are impossible to
realize. It is only by stepping back, and rethinking these
fundamental assumptions, that a solution can be found.
5.2. Challenging Past Assumptions
Two unstated assumptions of SIP federation are challenged by ViPR.
The first assumption that federation solutions have made is this:
The purpose of SIP federation is to eliminate the PSTN, and
consequently, we cannot assume the PSTN itself as part of the
solution.
Though unstated, this assumption has clearly been part of the design
of existing solutions. SIP federation based on email-style URIs, as
defined in RFC 3261, doesn't utilize or make mention of the PSTN.
Solutions like ENUM, or private registries, do not utilize or make
mention of the PSTN. In one sense, it's obvious that they shouldn't
- after all, the purpose is to replace the PSTN. However, such an
approach ignores an incremental solution - a solution which utilizes
the PSTN itself to solve the hard problems in SIP federation.
After all, the PSTN has accomplished a great deal. It reaches
worldwide. It provides a global numbering translation service that
maps phone numbers to circuits. It is highly reliable, and provides
QoS. It has been built up over decades to achieve these goals. This
begs the question - can we build upon the capabilities already
provided by the PSTN, and use them to solve the problems that plague
SIP federation?
Indeed, the answer is yes once another assumption is challenged.
This second assumption is:
A federation solution must be the same as the final target
federation architecture, and not just a step towards it.
Though unstated, this assumption has also been true. SIP's email-
style federation was a pure 'target architecture' - the place we want
to get to. ENUM was the same - a worldwide global DNS database with
everyone's phone numbers - an unrealizable nirvana of open
connectivity.
Historically, technologies are more successful when they are
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incrementally deployable. Indeed, in many cases, the target
architecture is unrealizable because there is no obvious way to get
there. As such, the focus needs to be on the next incremental step
that we can take, and that step in turn creates the technological and
market pressures that will drive the next step. In the end, the
target may not be the perfect nirvana we all imagined, but we've at
least arrived.
As such, ViPR is very much focused on incremental deployability. It
is not the end of the federation story, it is the beginning. It
discards the nirvana of perfect IP federation for a solution that
federates most, but not all calls, by relying on the PSTN to fill in
the gaps. ViPR's philosophy is not to let the perfect be the enemy
of the good.
5.3. Technical Overview
A high level view of the architecture is shown in Figure 1. The
figure shows four different domains, a.com, b.com, c.com and d.com,
federating using ViPR technology. Each domain is connected to both
the public Internet and to the traditional PSTN.
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//\\
\/
|
|
|
+-------+
| Call |
| Agent |
| |
| | d.com
+-------+
|
//-------\\
|// \\|
| Internet |
+-------+ |\\ //| +-------+
| Call | \\-------// | Call |
//\\ | Agent |-- --| Agent | //\\
\/ ---| | //-------\\ | |---- \/
| | |// \\| | |
+-------+ | PSTN | +-------+
|\\ //|
a.com \\-------// b.com
|
+-------+
| Call |
| Agent |
| |
| |
+-------+
| c.com
|
//\\
\/
Figure 1: High Level Architecture
For purposes of explanation, it is easiest to think of each domain as
having a single call agent which participates in the federation
solution. In actuality, the functionality is decomposed into several
sub-components, and this is discussed in more detail below. The call
agent is connected to one or more phones in the domain, and is
responsible for routing calls, handling features, and processing call
state. The call agent is stateful, and is aware of when calls start
and stop.
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Assume that all four domains have a 'fresh' installation of ViPR, and
that domain b.com 'owns' +1 (408) 902-5xxx, a block of 1000 numbers
allocated by its PSTN provider.
The ViPR mechanism can be broken into four basic steps: storage of
phone numbers, PSTN first call, validation and caching, and SIP call.
5.3.1. Storage of Phone Numbers
The first step is that the call agents form a single, worldwide P2P
network, using RELOAD [I-D.ietf-p2psip-base] with the Chord
algorithm. This P2P network forms a distributed hash table (DHT)
running amongst all participating domains. A distributed hash table
is like a simple database, allowing storage of key-value pairs, and
lookup of objects by key. Unlike a normal hash table, which resides
in the memory of a single computer, a distributed hash table is
spread across all of the servers which make up the P2P network. In
this case, it is spread across all of the domains participating in
the ViPR federation.
The neat trick solved by Chord (and by other DHT algorithms), is an
answer to the following: given that the desired operation is to read
or write an object with key K, which node in the DHT is the box that
currently stores the object with that key? Chord provides a clever
algorithm which routes read and write operations through nodes in the
DHT until they eventually arrive at the right place. With Chord,
this will take no more than log2N hops, where N is the number of
nodes in the DHT. Consequently, for a DHT with 1024 nodes, 10 hops
are required in the worst case. For 2048, 11 hops. And so on. The
logarithmic factor allows DHTs to achieve incredible scale and to
provide enormous storage summed across all of the nodes that make up
the DHT.
This logarithmic hopping behavior also means that each node in the
DHT does not need to establish a TCP/TLS connection to every other
node. Rather, connections are established to a smaller subset - just
log(N) of the nodes.
In DHTs, each participating entity is identified by a node-ID. The
node-ID is a 128 bit number, assigned randomly to each entity. They
have no inherent semantic meaning; they are not like domain names or
IP addresses.
In the case of ViPR, each call agent is identifed by one or more
node-IDs. For purposes of discussion, consider the case where the
call agent has just one. Each participating domain, including b.com
in our example, uses the DHT to store a mapping from each phone
number that it owns, to its own Node-ID. In the case of b.com, it
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would store 1000 entries into the DHT, each one being a mapping from
one of its phone numbers, to its own nodeID. Furthermore, when the
mappings are stored, the mapping is actually from the SHA-1 hash of
the phone number, to the nodeID of the call agent which claims
ownership of that number.
Pretending that the node-ID of the call agent in domain b.com is
0x1234 (a shorter 16 bit value to simplify discussion), the entries
stored into the DHT by b.com would be:
Key | Value
----------------------------------
SHA1(+14089025000) | 0x1234
SHA1(+14089025001) | 0x1234
SHA1(+14089025002) | 0x1234
.....
SHA1(+14089025999) | 0x1234
Figure 2: DHT Contents
It is important to note that the DHT does not contain phone numbers
(it contains hashes of them), nor does it contain IP addresses or
domain names. Instead, it is a mapping from the hash of a phone
number (in E.164 format) to a node-ID.
b.com will store this mapping when it starts up, or when a new number
is provisioned. The information is refreshed periodically by b.com.
The actual server on which these mappings are stored depends on the
Chord algorithm. Typically, the entries will be uniformly
distributed amongst all of the call agents participating in the
network.
5.3.2. PSTN First Call
At some point, a user (Alice) in a.com makes a call to +1 (408) 952-
5432, which is her colleague Bob. Even though both sides have ViPR,
the call takes place over the plain old PSTN. Alice talks to Bob for
a bit, and they hang up.
At a random point of time after the call has completed, the call
agent in a.com "wakes up" and says to itself, "that's interesting,
someone in my domain called +1 (408) 952-5432, and it went over the
PSTN. I wonder if that number is reachable over IP instead?". To
make this determination, it hashes the called phone number, and looks
it up in the DHT. It is important to note that this lookup is not at
the time of an actual phone call - this lookup process happens
outside of any phone call, and is a background process.
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The query for +1 (408) 952-5432 will traverse the DHT, and eventually
arrive at the node that is reponsible for storing the mapping for
that number. Typically, that node will not be b.com, but rather one
of the other nodes in the network (for example. c.com). In many
cases, the called number will not find a matching mapping in the DHT.
This happens when the number that was dialed is not owned by a domain
participating in ViPR. When that happens, a.com takes no further
action. Next time there is another call to the same number, it will
repeat the process and check once more whether the dialed number is
in the DHT.
In this case, there is a match in the DHT, and a.com learns the
node-ID of b.com. It then proceeds to the validation step. It is
also possible that there are multiple matches in the DHT. This can
happen if another domain - d.com for example - also claims ownership
of that number. When there are multiple matching results, a.com
learns all of them, and performs the validation step with each.
5.3.3. Validation and Caching
Why not just store the domain in the DHT, instead of the node-ID? In
that case, once a.com performed the lookup, it would immediately
learn that the number maps to b.com, and could then make a direct SIP
call next time.
The main reason this doesn't work is security. The information in
the DHT is completely untrusted. There is nothing so far that
enables a.com to know that b.com does, in fact, own the phone number
in question. Indeed, if multiple domains make a claim on the number,
it has no way to know which one (if any) actually owns it.
To address this critical problem, ViPR utilizes a technique called
phone number validation. Phone number validation is the key concept
in ViPR. The essential idea is that a.com will connect to the b.com
server, by asking the DHT to form a connection to b.com's nodeID.
Once connected, a.com demands proof of ownership of the phone number.
This proof comes in the form of demonstrated knowledge of the
previous PSTN call. When a call was placed from a.com to +1 (408)
952-5432, the details of that call - including its caller ID, start
time, and stop time, create a form of shared secret - information
that is only known to entities that participated in the call. Thus,
to obtain proof that b.com really owns the number in question, a.com
will demand a knowledge proof - that b.com is aware of the details of
the call. The only way that b.com could know these details is if it
had received the call, and the only way it could have received the
call is if it owned the phone number.
There are a great many details required for this validation protocol
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to be secured. It needs to handle the fact that call start and stop
times won't exactly match on both sides. It needs to deal with the
fact that many calls start on the top of the hour. It needs to deal
with the fact that caller ID is not often delivered, and when it is
delivered, is not reliable. It needs to deal with the fact that
a.com may in fact be the attacker, trying to use the validation
protocol to extract the shared secret from b.com. All of this is, in
fact, handled by the protocol. The protocol is based on the Secure
Remote Password for TLS Authentication (SRP-TLS) [RFC5054], and is
described more fully in [I-D.rosenberg-dispatch-vipr-pvp].
At the end of the validation process, both a.com and b.com have been
able to ascertain that the other side did in fact participate in the
previous PSTN call. At that point, a.com sends its domain name to
b.com (this is described in more detail below), and b.com sends to
a.com - all over a secured channel - a SIP URL to use for routing
calls to this number, and a ticket. The ticket is is a cryptographic
object, opaque to a.com, but used by b.com to allow incoming SIP
calls. It is similar in concept to kerberos tickets - it is a grant
of access. In this case, it is a grant of access for a.com to call
+1 (408) 952-5432, and only +1 (408) 952-5432.
The a.com call agent receives the SIP URI and ticket, and stores both
of them in an internal cache. This cache builds up slowly over time,
containing the phone number, SIP URI, and ticket, for those numbers
which are called by a.com and validated using ViPR. Because the
cache entries are only built for numbers which have actually been
called by users in the enterprise, the size of the cache self-scales.
A call agent supporting only ten users will build up a cache
proportional to the volume of numbers called by ten people, whereas a
call agent supporting ten thousand users will build up a cache which
is typically a thousand times larger.
5.3.4. SIP Call
At some point in the future, another call is made to +1 (408) 952-
5432. The caller could be Alice, or it could be any other user
attached to the same call agent. This time, the call agent notes
that it has a cached route for the number in question, along with a
SIP URI that can be used to reach that route. It also has a ticket.
The a.com call agent attempts to contact the SIP URI by establishing
a TCP/TLS connection to the SIP URI it learned. If this connection
cannot be made, it proceeds with the call over the PSTN. This
ensures that, in the event of an Internet failure or server failure,
the call can still proceed. Assuming the connection is established,
the a.com call agent sends a traditional SIP INVITE to the
terminating call agent, over this newly formed secure connection.
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The SIP call setup request also contains the ticket, placed into a
new SIP header in the message.
When this call setup request arrives at the b.com call agent, it
extracts the ticket from the new SIP header. This ticket is an
object, opaque to a.com, that was previously generated by the b.com
call agent. Figure 3 illustrates how this ticket is generated and
used.
//\\
\/
+----------------------+ |
| | |
| Hi, I am a.com. | |
| How do I reach you? | +-------+
| | | Call |
+--------------\-------+ | Agent |
\ | |
\ | | d.com
\ +-------+
\ |
\
\ //-------\\
+===================================+
^ | Internet | V
+-------+ |\\ //| +-------+
| Call | \\-------// | Call |
//\\ | Agent |-- --| Agent | //\\
\/ ---| | //-------\\ | |---- \/
| | |// \\| | |
+-------+ | PSTN | +-------+
|\\ //|
a.com \\-------// b.com
|
+-------+
| Call |
| Agent |
| |
| |
+-------+
| c.com
|
//\\
\/
Figure 3: Ticket Validation Step 1
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Towards the end of the validation process, domains a.com and b.com
had determined that each was, in fact in possession of the shared
secret information about the prior PSTN call. However, neither side
has any information about the domain names of the other side. The
originating domain - a.com - tells b.com that its domain name is
a.com. It offers no proof of this assertion at this time.
Next, the b.com domain generates the ticket. The ticket has three
fundamental parts to it:
1. The phone number that was just validated - in this case, +1 (408)
952-5432.
2. The domain name that the originating side claims it has - a.com
in this case.
3. A signature generated by b.com, using a key known to itself only,
over the other two pieces of information.
This ticket is then sent back to a.com at the end of the validation
process, as shown in Figure 4.
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//\\
\/
|
|
| +--------------------+
+-------+ | Here is your ticket|
| Call | |and a SIP URI to |
| Agent | | reach this number. |
| | | |
| | d.com +--------------------+
+-------+ /
| /
/
//-------\\ /
+===================================+
V | Internet | ^
+-------+ |\\ //| +-------+
| Call | \\-------// | Call |
//\\ | Agent |-- --| Agent | //\\
\/ ---| | //-------\\ | |---- \/
| | |// \\| | |
+-------+ | PSTN | +-------+
|\\ //|
a.com \\-------// b.com
|
+-------+
| Call |
| Agent |
| |
| |
+-------+
| c.com
|
//\\
\/
Figure 4: Ticket Validation Step 2
When a.com generates a SIP INVITE, it will contain this ticket. The
INVITE arrives at the b.com call agent over the mutually
authenticated TLS connection established between the domains.
The b.com call agent looks for the SIP header field in the INVITE
that contains the ticket. First, it verifies the signature over the
ticket. Remember that the b.com agent is the one that generated the
ticket in the first place; as such, it is in possession of the key
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required to validate the signature. Once validated, it performs two
checks:
1. It compares the phone number in the call setup request (the
Request URI) against the phone number stored in the ticket.
2. It compares the domain name of the calling domain, learned from
the certificates in the mutual TLS exchange, against the domain
name stored in the ticket.
If both match, the b.com call agent knows that the calling party is
in fact the domain they claimed previously, and that they had in fact
gone through the validation process successfully for the number in
question. A consequence of this is that the following property is
maintained:
A domain can only call a specific number over SIP, if it had
previously called that exact same number over the PSTN.
This property is key in fighting spam and denial-of-service attacks.
Because calling numbers on the PSTN costs money - especially
international calls - ViPR creates a financial disincentive for
spammers. For a spammer to ring every phone in a domain with a SIP
call, it must have previously called every number in the domain with
a PSTN call, and had a successfully completed call to each and every
one of them. Of course, once that PSTN call had been placed, the
spammer would have already achieved their goals, and at cost. The
additional VoIP call is not so exciting.
This property also means that, in order for an attacker to spam call
numbers on VoIP, it must have already spam-called those same numbers
on the PSTN. This means that the attacker would clearly be subject
to regulations and laws governing usage of the PSTN for calling. As
an example, a spammer in the United States would have already
violated U.S. do-not-call rules by initiating the spam calls to the
PSTN numbers.
It is important to note that ViPR does not completely address the
spam problem. A large spamming clearing house organization could
actually incur the costs of launching the PSTN calls to numbers, and
then, in turn, act as a conduit allowing other spammers to launch
their calls to those numbers for a fee. The clearinghouse would
actually need to transit the signaling traffic (or, divulge the
private keys to their domain name), which would incur some cost. As
such, while this is not an impossible situation, the barrier is set
reasonably high to start with - high enough that it is likely to
deter spammers until it becomes a highly attractive target, at which
point other mechanisms can be brought to bear. This is, again, an
example of the incremental deployability philosophy that ViPR takes -
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let not the perfect be the enemy of the good.
6. Architecture Components and Functions
The architecture in Figure 1 is overly simplistic. ViPR allows the
functionality embedded within the call agent can be split up into
three components, as shown in Figure 5:
+-+ +-+
| | | | +------+
| | +-----| |---|Enroll|
| | | | | +------+
|I| | |I|
|n| +-----+ |n|
VAP |t| | ViPR| |t|
+----------|r|---|Srvr |--|e|-----------------
| |a| | | |r| P2P-Validation
| |n| +-----+ |n|
| |e| |e|
| |t| |t|
+-----+ SIP | | +-----+ | |
| CA |-------|F|---| |--|F| ---------------
+-----+ |i| | | |i| SIP/TLS
. |r| | . | |r|
SIP/ . |e| | | |e|
MGCP/ . |w| | BE | |w|
TDM . |a| | | |a|
. |l| | | |l|
+-----+ |l| | | |l|
| UA |-------| |---| |--| |-----------------
+-----+ | | +-----+ | | SRTP
| | | |
+-+ +-+
| |
+--------------------+-----------------+
|
Single administrative domain
Figure 5: Architecture
Within each domain, there are three components that are ViPR-aware.
These are the ViPR server, the call agent (CA), and the border
element (BE). Outside of the domain, there is a P2P network and an
enrollment server. A domain will typically have firewalls - an
Internet firewall and an intranet firewall.
The sections which follow describe the roles and responsibilities of
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each component in more detail.
6.1. ViPR Server
The ViPR server is the heart of the system. It performs several key
functions:
1. It implements the P2P protocol, acting as one or more nodes in
the DHT. By placing this function separate from the call agent,
it allows the call agent to be isolated from the traffic and
security concerns that are often associated with a P2P network.
2. It implements the validation mechanism. It is informed of call
events by the call agent, and sometime after the call, looks up
the number in the DHT, and if found, attempts to connect to the
node claiming ownership of the number, and then validates it.
3. It pushes newly learned routes to the call agent once validation
has occurred. The ViPR server does not hold the call routes;
this eliminates the need for an off-box query to perform call
routing logic.
4. It stores numbers into the DHT. The call agent informs the ViPR
servers of numbers to be published, and the ViPR server places
them into the P2P network. Refreshing the stored numbers (by
asking the ViPR server to restore them) is the responsibility of
the call agent.
5. It implements a distributed quota enforcement algorithm, ensuring
that malicious ViPR servers cannot store excessive data into the
network.
6. It implements a policing function, pacing its store and fetch
requests into the DHT to ensure that the network is not
overwhelmed.
In order to join the P2P network and be able to receive incoming
validation requests, the ViPR server must have open access to the
public Internet. For this reason, it is typically placed into the
DMZ. The Internet firewall will require two pinholes to be opened
towards the ViPR server: one for the P2P protocol, and one for the
validation protocol.
It is important to understand that the ViPR server does not perform
any call processing. It does not process SIP or RTP traffic. It is
a non-real-time server that performs validation processing in the
background, outside of actual call attempts.
The ViPR server needs to connect with the call agent. This is done
through the ViPR Access Protocol (VAP). VAP is described in more
detail below.
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6.2. Call Agent
The call agent is a box within the domain which performs call
processing on behalf of one or more phones within the domain. ViPR
can work with a wide variety of call agents, as long as they meet
some specific criteria:
o The call agent must be know of the start time, stop time, caller
ID, and called numbers of calls placed from phones towards the
PSTN.
o The call agent must be capable of making routing decisions for
outbound calls from phones that would otherwise go to the PSTN,
directing them towards the PSTN or towards other domains (based on
ViPR routing rules).
Based on this definition, many different types of products typically
found within a domain could act as the call agent. An IP PBX or TDM
PBX with a SIP interface can be the call agent. A Session Border
Controller (SBC) that connects calls from a PBX to the PSTN, can act
as the call agent. An IMS application server can act as the call
agent. A PSTN gateway, used for all calls egressing a domain from a
set of phones, can act as a call agent.
A SIP proxy can act as a call agent; as long as it is capable of
stashing the relevant call information into Record-Route headers for
usage at the end of the call, it can even operate without retaining
call state.
A single phone can also act as the call agent, representing itself
and its own phone number.
In ViPR, the call agent performs several key functions specific to
ViPR:
o It informs the ViPR server of the phone numbers to be stored in
the DHT for its domain.
o It refreshes those numbers in the DHT, redoing the storage
operation periodically.
o At the end of a call, the call agent sends a ViPR Call Record
(VCR) to the ViPR server, containing the start time, stop time,
caller ID and called party number.
o It learns validated routes from the ViPR server. These routes
consist of a phone number, a SIP URI to utilize when contacting
that phone number, and a corresponding ticket. The call agent is
responsible for storing those routes.
o When a call is to be made towards a PSTN number, the call agent is
responsible for checking whether there is a route for that number,
learned via a prior notification from the ViPR server. If so, it
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is responsible for sending the INVITE towards the learned SIP URI,
and for including the ticket the X-Cisco-ViPR-Ticket header field.
Those functions which require communications with the ViPR server are
done by implementing VAP. VAP is a client-server protocol, with the
call agent acting as the client, and the ViPR server acting as the
server. For this reason, the call agent is sometimes called the VAP
client or ViPR client.
6.3. Border Element
The border element is responsible for the SIP layer perimeter
security functions. In particular:
o The border element ensures that all egress SIP traffic is carried
over TLS. Border elements must reject any incoming SIP requests
which are not over TLS. SIP over TLS is mandatory-to-use in ViPR,
and it must be performed using mutual TLS.
o The border element ensures that all egress RTP traffic is actually
carried using SRTP. If the traffic originated by the UA in the
domain is inherently SRTP, the criteria is met. However, many
domains do not utilize SRTP internally, and if it is not used
internally, the border element must convert to SRTP. Similarly,
the border element is responsible for rejecting any incoming SIP
calls that are not set up with SRTP. SRTP is mandatory in ViPR.
o The border element ensures that ingress and egress SIP traffic is
'fixed up' so that it can pass through the Internet firewall
succesfully. Typically, this is done using a traditional SBC/ALG
function.
o The border element inspects all incoming SIP INVITEs, and performs
ticket verification. In this process, it looks for the X-Cisco-
ViPR-Ticket header field in the INVITE. If not present, it
discards the request. If present, it verifies the signature, and
then compares the called number and remote TLS domain against the
contents of the ticket. If they do not match, the border element
discards the INVITE.
The border element can perform other, non-ViPR tasks, as is common
for border elements. These include header inspection and validation,
anti-virus checks on embedded content, SIP state machine conformance,
policy checks on various services, and so on.
The role of the border element can be fulfilled by any number of
products typically found within domains. These include Session
Border Controllers and firewalls. Indeed, the border element
function can be embedded directly in the Internet firewall.
The border element is connected to the call agent via SIP, and to the
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user agent (UA) via RTP. The border element has no direct connection
to the ViPR server. However, in order for ticket processing to work
in this model, the ViPR server and border element must share a secret
that is used to create the tickets. This is discussed in more detail
below.
6.4. Enrollment Server
P2P protocols - including RELOAD - require the usage of an enrollment
server in order to obtain the certificates that are used to secure
the network. ViPR uses, and indeed requires, that all RELOAD traffic
be over TCP/TLS with mutual authentication. The certificates used
are obtained through an enrollment process. The details on how P2P
enrollment are done are beyond the scope of this document.
6.5. P2P Network
The collection of ViPR servers form a single, worldwide, P2P network
utiizing RELOAD and the Chord algorithm.
It is very important to understand that the DHT is never accessed in
real-time. It is not queried at call setup time. This is because
the DHT is slow, involving many hops. Queries could take seconds.
Furthermore, we don't want to rely on proper operation of the DHT to
actually make calls.
7. Protocols
The overall ViPR solution utilizes several protocols, each peforming
a different function.
7.1. P2P: RELOAD
ViPR utilizes the RELOAD protocol [I-D.ietf-p2psip-base] to run
amongst each of the ViPR servers. Each ViPR server acts as one or
more nodes in the DHT. The number of nodes that the ViPR server
implements directly determines the quota allocated to that ViPR
server, and in turn, the amount of work it must perform storing data.
ViPR, however, does not implement the SIP usage that has been defined
for RELOAD [I-D.ietf-p2psip-sip]. That is because the DHT is not
used as a traditional distributed registrar. Instead, it implements
a new usage - the ViPR usage - which stores phone numbers. It also
utilizes the DHT for storage of certificates, using a certificate
usage.
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7.1.1. ViPR Usage
The ViPR usage is described in detail in
[I-D.rosenberg-dispatch-vipr-reload-usage]. This section provides a
brief overview.
The ViPR usage makes use of the dictionary type. Each resource ID is
a key, computed by taking the SHA1 hash of an E.164 formatted phone
number. The value stored at this resource-ID is a dictionary. The
dictionary entries are the set of virtual ViPR servers which claim
ownership of those numbers.
Since a ViPR server might support a multiplicity of call agents from
different domains, it is necessary to logically segment a ViPR server
so that - from a security perspetive - it operates logically like
different virtual ViPR servers, one for each call agent. Each
virtual instance of a ViPR server is called a VService. Thus, the
entries in the dictionary are key value pairs whose key is the
concatenation of the node-ID and an identifier for the VService
within that node. The value at each key is the node-ID to contact
for validation.
When a node in the DHT receives a Store request, and it is the
responsible node for the resource ID, it will verify that the node-ID
in both the key and value of the dictionary entry match the node-ID
in the certificate it presents. This ensures that one ViPR server
can never overwrite data from another ViPR server.
The ViPR usage also specifies a quota mechanism. Unlike the SIP
usage, where there are very specific rules about what resource-IDs a
node may store into the DHT, with ViPR, there is no way to restrict
what resource IDs may be stored by a ViPR server. This is because,
in ViPR, the resource IDs are derived from phone numbers, and at the
time of storage, there is no way to know whether the node performing
the store actually owns this phone number. Consequently, a
responsible node will accept stores from any node for any resource
ID. However, to limit malicious users from consuming all of the
resources of the DHT, the ViPR usage imposes a quota on storage.
Each node performing a store is allocated a fixed quota on the number
of records it can place into the DHT. A probabilistic enforcement
model is utilized at each responsible node based on the fraction of
the hashspace owned by that responsible node. Roughly speaking, if
the system quota is 10,000 phone numbers per node-ID, if a
responsible node owns 10% of the DHT, it will accept an average of
1000 phone numbers from any one single node-ID.
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7.1.2. Certificate Usage
Further details pending.
7.2. ViPR Access Protocol (VAP)
The ViPR Access Protocol (VAP) is documented in
[I-D.draft-rosenberg-dispatch-vipr-vap].
VAP is a client-server protocol that runs between the call agent and
the ViPR server. VAP is a simple, binary based, request/response
protocol. It utilizes the same syntactic structure and transaction
state machinery as STUN [RFC5389], but otherwise is totally distinct
from it. VAP clients initiate TCP/TLS connections towards the ViPR
server. The ViPR server never opens connections towards the call
agent. This allows the ViPR servers to run on the public side of
NATs and firewalls.
Once the connections are established, the call agent sends a Register
message to the ViPR server. This register message primarily provides
authentication and connects the client to the ViPR server. VAP
provides several messages for different purposes:
o Publish: The Publish message informs the ViPR server of service
information. There are two types of Publishes supported in ViPR.
The first is the ViPR Service (VService). This informs the ViPR
server of the SIP URIs on the call agent and black and white lists
used by the ViPR server to block validations. The ViPR server
stores that information locally and uses it during the validation
process, as described above. The second Publish is the ViPR
number service. The ViPR server, upon receiving this message,
performs a Store operation into the DHT.
o UploadVCR: This message comes in two flavors - an originating and
terminating message. An originating UploadVCR comes from a call
agent upon completion of a non-VIPR call to the PSTN. A
terminating UploadVCR comes from an agent upon completion of a
call received FROM the PSTN. The ViPR server behavior for both
messages is very different. For Originating UploadVCR, the ViPR
server will store these, and at a random time later, query the DHT
for the called number and attempt validation against the ViPR
servers that are found. For a terminating UploadVCR, the ViPR
server will store these, awaiting receipt of a validation against
them.
o Subscribe: Call agents can subscribe for information from the
VipR server. There is one service that UCM can subscribe for:
number Service. When a new number is validated, the ViPR server
will send a Notify to the call agent, containing the validated
number, the ticket, and a set of SIP trunk URIs.
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o Notify: The ViPR server sends this message to the call agent when
it has an event to report for a particular subscription.
The VAP protocol provides authentication by including an integrity
object in each message. This integrity message is the hash of the
contents of the message and a shared secret between the ViPR server
and the client. VAP can also be run over TLS, which enhances
security further.
The P2P network introduces rate limits for the purposes of
performance management and limiting denial of service attacks. Each
node in the DHT comes with it a limit on the amount of stores per
second, reads per second, and total amount of data it can store in
the DHT. The ViPR server rigorously follows those limits.
As a consequence, when numbers are stored into the DHT, they are
written in slowly based on the rate limits. The call agent will send
a Publish operation for each individual number. The ViPR server will
perform the store in a rate-limited fashion. When the store is
complete, the ViPR server responds to the Publish, and the call agent
can move to the next DID to publish. Thus, it may take hours or even
days to fully store the set of numbers into the DHT. The process
then repeats several days later in order to refresh the data in the
DHT.
7.3. Validation Protocol
The core of ViPR is the validation protocol. The validation protocol
is used by one ViPR server to connect to another, demand proof-of-
knowledge of a previous PSTN call, and once proven, securely learn a
SIP URI and ticket for usage in future SIP calls between domains.
The validation protocol is documented in
[I-D.rosenberg-dispatch-vipr-pvp].
The validation protocol is built using TLS-SRP [RFC5054]. TLS-SRP
creates a secure TLS connection, but instead of using certificates,
utilizes a password. TLS-SRP was designed for cases where the
passwords are relatively weak. In the case of the validation
protocol, the passwords are formed from parameters of a previous PSTN
call. Once a secure TLS connection is formed, a simple request/
response protocol is run over it. The request contains the domain
name of the originating ViPR server, and the response contains the
SIP URI and ticket for that number.
The validation protocol properly handles time offsets between the two
domains for the start and stop times of the calls, the relatively
weak entropy of a single phone call, the grand chessmaster attack,
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and non-delivery or inaccurate delivery of caller-ID, amongst other
issues. The validation protocol can be tuned by administrators to
allow for arbitrary levels of security, measured in terms of
equivalent entropy. The equivalent entropy is the number of bits of
entropy that must be demonstrated, as if the domains were
authenticating each other using a password with that amount of
entropy. This gives domains a 'nerd knob' they can turn to trade off
security for performance.
Because the validation protocol utilizes TLS-SRP, it does not run
directly through the DHT. This is why a ViPR server requires a
separate pinhole to be opened for the validation protocol.
7.4. SIP Extensions
The connection between the call agents in different domains is SIP.
ViPR requires that the inter-domain connections run over TLS, and
furthermore, utilize SRTP keyed with Sdescriptions.
ViPR extends SIP with its anti-spam mechanism. This takes the form
of a ticket, present in a SIP header field.
[I-D.rosenberg-dispatch-vipr-sip-antispam] defines this header field
and the format of the ticket it contains.
8. Example Call Flows
This section provides call flows for the key use cases.
8.1. PSTN Call and VCR Upload
A call flow for the initial PSTN call and VCR upload is shown in
Figure 6.
Alice CA+O GW+O VIPR+O GW+T CA+T VIPR+T Bob
|(1) Call NumX | | | | | |
|------->| | | | | | |
| |(2) INVITE NumX | | | | |
| |------->| | | | | |
| | |(3) setup NumX | | | |
| | |---------------->| | | |
| | | | |(4) INVITE NumX | |
| | | | |------->| | |
| | | | | |(5) Call NumX |
| | | | | |---------------->|
| | | | | | | |Answers
| | | | | |(6) answer |
| | | | | |<----------------|
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| | | | |(7) 200 OK | |
| | | | |<-------| | |
| | | | |(8) ACK | | |
| | | | |------->| | |
| | |(9) answer | | | |
| | |<----------------| | | |
| |(10) 200 OK | | | | |
| |<-------| | | | | |
| |(11) ACK| | | | | |
| |------->| | | | | |
|(12) accept | | | | | |
|<-------| | | | | | |
|hangs up| | | | | | |
|(13) hangup | | | | | |
|------->| | | | | | |
| |(14) BYE| | | | | |
| |------->| | | | | |
| |(15) 200 OK | | | | |
| |<-------| | | | | |
| | |(16) hangup | | | |
| | |---------------->| | | |
| | | | |(17) BYE| | |
| | | | |------->| | |
| | | | |(18) 200 OK | |
| | | | |<-------| | |
| | | | | |(19) hangup |
| | | | | |---------------->|
| |(20) Orig UploadVCR | | | |
| |---------------->| | | | |
| |(21) Success | | | | |
| |<----------------| | | | |
| | | |Set timer | | |
| | | | | |(22) Term UploadVCR
| | | | | |------->| |
| | | | | |(23) Success |
| | | | | |<-------| |
Figure 6: PSTN Call and Upload
In message 1, Alice calls the number of her colleauge, Bob. This is
NuMX. This call is routed over the PSTN, through the terminating
call agent, and rings Bob's phone (messages 1-5). Bob answer the
phone, and this is propagated back to Alice (messages 6-12). Bob and
Alice talk for a while, and then Alice hangs up. This hangup is
propagated to Bob, and the call is terminated (messages 13-19).
The originating call agent notes that this call went to the PSTN, and
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might be a candidate for a future SIP call. It sends an UploadVCR
message to its ViPR server (message 20), containing the start time,
stop time, callerID and called party number. The ViPR server
acknowledges this (message 21), and then sets a timer for a random
time into the future, at which point it will attempt validation. The
terminating side is similar; it sends an UploadVCR to its ViPR server
(message 22), which is acknowledged (message 23). The terminating
side does not set a timer; it waits for a possible validation attempt
which may or may not arrive in the future.
8.2. DHT Query and Validation
This section provides the call flow for what happens on the
originating ViPR server when the timer fires, in Figure 7.
CA+O VIPR+O DHT VIPR+T
| |timer fires | |
| |(1) Query NumX | |
| |------------------->| |
| |(2) nodeID-T | |
| |<-------------------| |
| |(3) Connect nodeID-T| |
| |------------------->| |
| | |(4) Connect nodeID-T|
| | |------------------->|
| | |(5) Connect resp. |
| | |<-------------------|
| |(6) Connect resp. | |
| |<-------------------| |
| |(7) TCP Connect | |
| |---------------------------------------->|
| |(8) TLS-SRP | |
| |---------------------------------------->|
| |(9) ValExchange(a.com) |
| |---------------------------------------->|
| |(10) ValResponse(URI, ticket) |
| |<----------------------------------------|
|(11) Notify(NumX,URI,ticket) | |
|<-------------------| | |
|Store route | | |
Figure 7: Validation Flow
First, the timer that was set by the originating ViPR server in
Figure 6 fires. When it fires, the ViPR server examines the called
party number from the VCR. It performs a query into the DHT, to see
if this number has been stored by any domain (message 1). In this
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case, it has, and the DHT returns with a successful query response
(message 2). This response indicates that the terminating ViPR
server, with nodeID T, claims ownership of the number.
The originating ViPR server asks the DHT to form a connection between
itself and the terminating ViPR server. This message exchanges IP
addresses and ports through which a TCP connection can be attempted;
details are omitted (messages 3-6). Now, the originating ViPR server
can establish a TCP connection to the terminating ViPR server
(message 7). Next, the originating ViPR server begins negotiation of
a TLS-SRP connection. The TLS-SRP uses the caller ID and called
number as a "username" for this exchange, and the start time and stop
time of the call as a password. As both sides share the same values
for this secret, the secure connection is established. This is now a
TLS connection between the two ViPR servers.
Over this secure connection, the originating ViPR server sends a
ValExchange request. This request contains the domain name that is
claimed by the originating ViPR server (this claim is not verified at
this time) (message 9). This is received by the terminating ViPR
server, which then creates a ticket for that domain and numX, and
passes the ticket and the SIP URI back to the originating VipR server
(message 10). The originating VipR server sends this information to
its call agent (message 11), which then stores it for usage in a
future call.
8.3. DHT Query and No Match
In this case, after the PSTN call of Figure 6, the timer fires, but
the originating ViPR server finds no match in the DHT. This is an
alternative case to the flow in Figure 7.
CA+O VIPR+O DHT VIPR+T
| |timer fires | |
| |(1) Query NumX | |
| |------------------->| |
| |(2) noMatch | |
| |<-------------------| |
Figure 8: DHT No-Match
8.4. SIP Call
In this case, shown in Figure 9, a user makes a call to a number
which has been learned via ViPR.
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Alfred CA+O BE+O BE+T CA+T Bob
|(1) Call NumX | | | | |
|------------->| | | | |
| |(2) INVITE NumX | | |
| |Ticket | | | |
| |------------->| | | |
| | |(3) TCP and TLS | |
| | |w domain certs| | |
| | |------------->| | |
| | |(4) INVITE NumX | |
| | |Ticket | | |
| | |------------->| | |
| | | |Validate Ticket |
| | | |(5) INVITE NumX |
| | | |Ticket | |
| | | |------------->| |
| | | | |(6) INVITE NumX
| | | | |------------->|
Figure 9: SIP Call
First, a user in the originating domain - Alfred - calls Bob's number
(message 1). The originating call agent notes that it has a cached
route for that number. It extracts the SIP URI, using it as the
topmost Route header field, and then attaches the ticket to the
X-Cisco-ViPR-Ticket header field. This INVITE is sent to a default
next hop border element (message 2). The border element establishes
a TCP/TLS connection with the domain in the Route header. It uses a
traditional domain certification for this TLS connection (message 3).
Once established, it sends the INVITE over the connection (message
4).
This arrives at the terminating call agent, which extracts the ticket
and verifies it. To verify it, it checks the signature using the key
that was used to create the ticket. Then, it compares the domain
name in the ticket with the domain name from the TLS connection
handshake. Finally, it compares the called party number in the
Request-URI with the value from the ticket. Assuming they all match,
the call is forwarded to the terminating call agent (message 5),
where it is finally delivered to Bob (message 6).
9. Security Considerations
Security is incredibly important for ViPR. This section provides an
overview of some of the key threats and how they are handled.
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9.1. Theft of Phone Numbers
The key security threat that ViPR is trying to address is the theft
of phone numbers. In particular, a malicious domain could store,
into the DHT, phone numbers that it does not own, in an attempt to
steal calls targeted to those numbers. This attack is prevented by
the core validation mechanism, which performs a proof of knowledge
check to verify ownership of numbers.
An attacker could try to claim numbers it doesn't own, which are
claimed legitimately by other domains in the ViPR network. This
attack is prevented as well. Each domain storing information into
the DHT can never overwrite information stored by another domain. As
a consequence, if two domains claim the same number, two records are
stored in the DHT. An originating domain will validate against both,
and only one will validate - the real owner.
An attacker could actually own a phone number, use it for a while,
validate with it, and build up a cache of routes at other domains.
Then, it gives back the phone number to the PSTN provider, who
allocates it to someone else. However, the attacker still claims
ownership of the number, even though they no longer have it. This
attack is prevented by expiring the learned routes after a while.
Typically, operators do not re-assign a number for a few months, to
allow out-of-service messages to be played to people that still have
the old number. Thus, the TTL for cached routes is set to match the
duration that carriers typically hold numbers.
An attacker could advertise a lot of numbers, most of which are
correct, some of which are not. ViPR prevents this by requiring each
number to be validated individually.
9.2. Eavesdropping
Another class of attacks involves outsiders attempting to listen in
on the calls that run over the Internet, or obtain information about
the call through observation of signaling.
All of these attacks are prevented by requiring the usage of SIP over
TLS and SRTP. These are mandatory to use.
9.3. Attacks on the DHT
Attackers could attempt to disrupt service through a variety of
attacks on the DHT.
Firstly, it must be noted that the DHT is never used at call setup
time. It is accessed as a background task, solely to learn NEW
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numbers and routes that are not already known. If, by some tragedy,
an attacker destroyed the P2P network completely, it would not cause
a single call to fail. Furthermore, it would not cause calls to
revert to the PSTN - calls to routes learned previously would still
go over the IP network. The only impact to such a devastating
attack, is that a domain could not learn *new* routes to new numbers,
until the DHT is restored to service. This service failure is hard
for users and administrators to even notice.
That said, VIPR prevents many of these attacks. The DHT itself is
secured using TLS - its usage is mandatory. Quota mechanisms are put
into place that prevent an attacker from storign large amounts of
data in the DHT. Other attacks are prevented by mechanisms defined
by RELOAD itself, and are not ViPR specific.
9.4. Spam
Another serious concern is that attackers may try to launch VoIP spam
(also known as SPIT) calls into a domain. ViPR prevents this by
requiring that a domain make a PSTN call to a number before it will
allow a SIP call to be accepted to that same number. This provides a
financial disincentive to spammers. The current relatively high cost
of international calling, and the presence of national do-not-call
regulations, have prevented spam on the PSTN to a large degree. ViPR
applies those same protections to SIP connections.
As noted above, ViPR still lowers the cost of communications, but it
does so by amortizing that savings over a large number of calls. The
costs of communications remain high for infrequent calls to many
numbers, and become low for frequent calls to a smaller set of
numbers. Since the former is more interesting to spammers, ViPR
gears its cost incentives away from the spammers, and towards domains
which collaborate frequently.
Of course, ViPR's built-in mechanism is not a guarantee. A SPIT
clearinghouse could shoulder the costs of the PSTN calls, and then
re-sell its access for a fee. However, this still causes the
clearinghouse to utilize non-trivial resources in its attack. Though
these costs are less than the PSTN, they are more than zero, and
should act as a deterrent for a long while.
10. IANA Considerations
This specification does not require any actions from IANA.
11. References
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11.1. Normative References
[I-D.ietf-p2psip-base]
Jennings, C., Lowekamp, B., Rescorla, E., Baset, S., and
H. Schulzrinne, "REsource LOcation And Discovery (RELOAD)
Base Protocol", draft-ietf-p2psip-base-05 (work in
progress), October 2009.
[I-D.ietf-p2psip-sip]
Jennings, C., Lowekamp, B., Rescorla, E., Baset, S., and
H. Schulzrinne, "A SIP Usage for RELOAD",
draft-ietf-p2psip-sip-03 (work in progress), October 2009.
[I-D.rosenberg-dispatch-vipr-reload-usage]
Rosenberg, J. and C. Jennings, "A Usage of Resource
Location and Discovery (RELOAD) for Public Switched
Telephone Network (PSTN) Verification", November 2009.
[I-D.rosenberg-dispatch-vipr-sip-antispam]
Rosenberg, J. and C. Jennings, "Session Initiation
Protocol (SIP) Extensions for Blocking VoIP Spam Using
PSTN Validation", November 2009.
[I-D.draft-rosenberg-dispatch-vipr-vap]
Rosenberg, J. and C. Jennings, "Validation Access Protocol
(VAP)", November 2009.
[I-D.rosenberg-dispatch-vipr-pvp]
Rosenberg, J. and C. Jennings, "The Public Switched
Telephone Network (PSTN) Validation Protocol (PVP)",
November 2009.
11.2. Informative References
[RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J.
Rosenberg, "SIP: Session Initiation Protocol", RFC 2543,
March 1999.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263,
June 2002.
[RFC5039] Rosenberg, J. and C. Jennings, "The Session Initiation
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Protocol (SIP) and Spam", RFC 5039, January 2008.
[RFC3761] Faltstrom, P. and M. Mealling, "The E.164 to Uniform
Resource Identifiers (URI) Dynamic Delegation Discovery
System (DDDS) Application (ENUM)", RFC 3761, April 2004.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[RFC5067] Lind, S. and P. Pfautz, "Infrastructure ENUM
Requirements", RFC 5067, November 2007.
[RFC5054] Taylor, D., Wu, T., Mavrogiannopoulos, N., and T. Perrin,
"Using the Secure Remote Password (SRP) Protocol for TLS
Authentication", RFC 5054, November 2007.
[I-D.roach-sipping-clf-syntax]
Roach, A., "Binary Syntax for SIP Common Log Format",
draft-roach-sipping-clf-syntax-01 (work in progress),
May 2009.
[I-D.kaplan-sip-session-id]
Kaplan, H., "A Session Identifier for the Session
Initiation Protocol (SIP)", draft-kaplan-sip-session-id-02
(work in progress), March 2009.
Authors' Addresses
Jonathan Rosenberg
jdrosen.net
Monmouth, NJ
US
Email: jdrosen@jdrosen.net
URI: http://www.jdrosen.net
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Cullen Jennings
Cisco
170 West Tasman Drive
MS: SJC-21/2
San Jose, CA 95134
USA
Phone: +1 408 421-9990
Email: fluffy@cisco.com
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