Verification Involving PSTN Reachability:
Requirements and Architecture Overviewjdrosen.netMonmouthNJUSjdrosen@jdrosen.nethttp://www.jdrosen.netCisco170 West Tasman DriveMS: SJC-21/2San JoseCA95134USA+1 408 421-9990fluffy@cisco.com
RAI
dispatch
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.
This documents and the information contained therein are provided on
an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND
THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
INFORMATION THEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
The Session Initiation Protocol (SIP) was originally published as
RFC 2543 in May of 1999. This was
followed by subsequent publication of RFC
3261, 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.
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.
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, 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.
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
. 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.
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 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.
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
and consistent session
IDs , 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.
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.
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
, 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
have limited its deployment:
The majority of SIP deployments utilize phone numbers, often
exclusively. In such a case, domain routing cannot be used.
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.
Public ENUM, defined in , 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
. 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.
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
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.
From the discussion on the problems of inter-domain federation and the
solutions that have been attempted so far, several key requirements
emerge:
The solution should allow for federation between
any number of domains.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.The solution must work with deployments that
utilize any kind of endpoint, including non-IP phones connected
through gateways, IP softphones and hardphones.
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.
The solution should work worldwide, for any
domain anywhere.
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.
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.
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.
The solution must have built-in mechanisms for
preventing spam and DoS attacks. This mechanism must be fully
automated.
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.
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.
ViPR has several important properties that enable it to solve the
federation problem:
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.
ViPR does not require
any changes to endpoints. Consequently, it works with existing SIP
endpoints, or with non-IP endpoints connected through gateways.
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.
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.
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.
ViPR has nearly infinite scale. Any
number of domains can participate.
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.
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.
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
. 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.
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.
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.
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.
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
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.
A high level view of the architecture is shown in
. 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.
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.
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.
The first step is that the call agents form a single,
worldwide P2P network, using RELOAD
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 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:
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.
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.
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.
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
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) , and is
described more fully in .
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.
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. 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. illustrates how this ticket
is generated and used.
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:
The phone number that was just validated - in this case, +1 (408)
952-5432. The domain name that the originating side claims it has - a.com in
this case. 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 .
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
required to validate the signature. Once validated, it performs two
checks:
It compares the phone number in the call setup request (the Request
URI) against the phone number stored in the ticket.
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 - let not the
perfect be the enemy of the good.
The architecture in is overly
simplistic. ViPR allows the functionality embedded within the
call agent can be split up into three components, as shown in
:
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
each component in more detail.
The ViPR server is the heart of the system. It performs several key
functions:
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.
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.
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.
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.
It implements a distributed quota enforcement algorithm, ensuring that
malicious ViPR servers cannot store excessive data into the network.
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.
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:
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.
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:
It informs the ViPR server of the phone numbers to be stored in the
DHT for its domain.
It refreshes those numbers in the DHT, redoing the storage
operation periodically.
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.
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.
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 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.
The border element is responsible for the SIP layer perimeter security
functions. In particular:
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.
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.
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.
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
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.
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.
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.
The overall ViPR solution utilizes several protocols, each peforming a
different function.
ViPR utilizes the RELOAD protocol
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 . 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.
The ViPR usage is described in detail in
. 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.
Further details pending.
The ViPR Access Protocol (VAP) is documented in
.
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 , 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:
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.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.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.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.
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
.
The validation protocol is built using TLS-SRP
. 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,
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.
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.
defines this header field and the format of the ticket it contains.
This section provides call flows for the key use cases.
A call flow for the initial PSTN call and VCR upload is shown in
.
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
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.
This section provides the call flow for what happens on the
originating ViPR server when the timer fires, in
.
First, the timer that was set by the originating ViPR server in
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 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.
In this case, after the PSTN call of
, the timer fires, but the
originating ViPR server finds no match in the DHT. This is an
alternative case to the flow in .
In this case, shown in , a user makes a
call to a number which has been learned via ViPR.
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).
Security is incredibly important for ViPR. This section provides an
overview of some of the key threats and how they are handled.
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.
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.
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 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.
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.
This specification does not require any actions from IANA.A Usage of Resource Location and
Discovery (RELOAD) for Public Switched Telephone Network (PSTN)
VerificationCiscoEdisonNJUSjdrosen@jdrosen.nethttp://www.jdrosen.netCisco170 West Tasman DriveMS: SJC-21/2San JoseCA95134USA+1 408 421-9990fluffy@cisco.com
RAI
dispatchSession Initiation Protocol (SIP)
Extensions for Blocking VoIP Spam Using PSTN ValidationCiscoEdisonNJUSjdrosen@jdrosen.nethttp://www.jdrosen.netCisco170 West Tasman DriveMS: SJC-21/2San JoseCA95134USA+1 408 421-9990fluffy@cisco.com
RAI
dispatchValidation Access Protocol (VAP)CiscoEdisonNJUSjdrosen@jdrosen.nethttp://www.jdrosen.netCisco170 West Tasman DriveMS: SJC-21/2San JoseCA95134USA+1 408 421-9990fluffy@cisco.com
RAI
dispatchThe Public Switched
Telephone Network (PSTN) Validation Protocol (PVP)CiscoEdisonNJUSjdrosen@jdrosen.nethttp://www.jdrosen.netCisco170 West Tasman DriveMS: SJC-21/2San JoseCA95134USA+1 408 421-9990fluffy@cisco.com
RAI
dispatch