VoIP is dead, over and
"out!" IP interactive communications (IC) or unified communications (UC) is
"in!" IC or UC is much more than voice. Even the acronyms suggest so - "I
see" and "You see." It's more than just video. It's insight
gained from the availability of your family's, friend's or colleague's
presence information and the ability to collaborate for business, learning or
pleasure with the simultaneous use of data applications. Sounds like network
nirvana!
But how will this really
work? IC and UC services and applications will only become valuable when we can
use them to reach anyone, anywhere, anytime. To paraphrase Metcalfe's Law: the
usefulness, or utility, of interactive communication equals the square of the
number of users. Consequently, IC/UC must span multiple IP networks --
business, residential and mobile; wireline, wireless and cable. Today's
consumers and businesses will be satisfied with--and pay money for--nothing
less.
Our only options for
delivering this network nirvana are the Internet or the Federnet - a federation
of managed IP networks. There are seven fundamental IP network precepts driving
the emergence of the Federnet for IC/UC.
1. In IP, We Trust No One
Service providers and
enterprises delivering interactive, unified communications need to protect their
service and application infrastructure against DoS and DDoS attacks. These
attacks might be malicious or non-malicious. Malicious attacks are pre-meditated
by hackers, the black hats. For most of them, their goal is no longer just the
personal gratification that they could do it, but to benefit financially. For
example, shorting the stock of their service provider target. Terrorist cyber
attacks are another type of malicious attack. Their goal is destroying the
infrastructure used by a democratic, capitalistic society. Non-malicious attacks
might result from recently upgraded endpoints that start registering every
single second or an overload to an American Idol voting telephone number.
Regardless of the type of attack, a successful attack will conclude in a variety
of losses including SLA promises, customers, a reputation, a complete business,
and perhaps even human lives.
Subscribers are not capable
of protecting themselves from everything. They need and want to trust their
service provider. They need to ensure that their communications are actually
established with their intended callee whether the callee is personally known to
them or not. They don't want their IP PBXs, voice servers or PCs to be
maliciously attacked or personally besieged or harassed by calls from any
anonymous user on the Internet. They demand guarantees relative to the identity
of callers and the ability to block unknown callers. For some communications,
they will demand privacy or confidentiality in their communications.
2. Addresses will Perpetually
be a Collection of Heterogeneous Schemes
Despite what the Internet
purist want and want you to believe, the Internet is not open end-to-end. For
security reasons data firewall/NAT devices are everywhere protecting large web
sites, enterprise networks and even PCs at home. Developed only for IP data
applications, they only allow data into a network if it has been requested from
someone or something inside the protected network. But voice is different. I
need to be able to call you from outside your network. I can't, and you
can't call me -- a proverbial Catch 22.
Additionally, more and more
enterprises are using managed network-based MPLS VPNs (RFC 2547), private
networks in simpler words, for secure managed network connections between
locations. But these network islands do not currently have secure bridges to
those of their service provider or other enterprises capable of supporting IP
interactive communications.
Many service provider VoIP
trunking and IP PSTN termination network islands have been built using private
address spaces. So interconnection with others also using overlapping private
address spaces is impossible. In developing countries like China that are
starved for IP address space, "public" networks are being built in a way to
conserve what little IPv4 address space they have using NAT devices everywhere.
The promise of IPv6, while
eliminating our thirst for address space, will only increase the address space
babel. It will be years, maybe decades, maybe never before everything uses an
IPv6 address. Even then firewalls/NAT devices will not be eliminated in light of
their important security role.
The problem of address
space mediation doesn't only exist at the IP layer. Another type of address
space problem relates to telephone number incompatibilities. SIP URIs like shourihan@acmepacket.com
are the ultimate solution, but we aren't even close to that being a reality
any time soon. Believe it or not, even in the world of VoIP there is a
requirement to add or strip number -- add a "1", strip the "011" -
before passing them on to another VoIP network that has its own, often myopic
view of the world.
3. SIP is Not the Only
nor a Single Signaling Protocol
The reality today is that
we live in a multiprotocol signaling world. Since all next-generation services
architectures including 3GPP IMS, ETSI TISPAN, ATIS, the Multi-Service Forum,
and PacketCable have embraced SIP, SIP will ultimately be THE protocol for new
wireline, wireless and cable services. However, SIP will NOT be the only
protocol for sometime. Today H.323 exists in many different types of networks.
These networks include international PSTN trunking and termination networks.
Most new IP PBX offerings being deployed today are still using H.323. New H.323
voice and video services are being deployed by service providers in countries as
different as Italy and China. MGCP is also being deployed for new voice services
by several US ILECs. PacketCable today uses NCS, an MGCP derivative. Even H.248
aka Megaco is also being deployed for new services.
It would be bad enough if
we just had these high-level differences in signaling protocol "languages."
But we also have different "dialects" of each language. H.323 is notorious
for its multiple versions (1 through 4), annex, service and configuration
options, and incompatible vendor implementations such as Cisco, Clarent,
VocalTec and others. In the world of SIP, while most implementations are
compliant with the current RFC 3261 standard, there are still products that
adhere only to the old standard RFC 2543. Even RFC 3261 is not "single"
standard. It offers multiple options for transport protocols -- UDP, TCP and
SCTP; multiple options for signaling security -- none, TLS, MTLS, DTLS, IPSec;
multiple options for media security - none, SRTP, IPSec. The transport of DTMF
digits may be carried in-band within the media flow using RFC 2833 or
out-of-band within the SIP signaling messages. Another very obscure level of
signaling protocol incompatibility relates to response or cause codes. Within
some deployments there are requirements to translate SIP cause codes such as
"404 - Server Not Found" to "503 -- Service Unavailable" before passing
them into another network to precipitate the correct "network busy signal"
vs. incorrect "dead air" service behavior. While all these different options
provide tremendous choice and flexibility, they also guarantee incompatibility
and lack of interoperability between networks.
4. Codecs will Never Converge
to a Couple - One for Audio, One for Video
While the world will
ultimately standardize on SIP for new deployments, codecs will never converge to
only a couple -- one for voice and one for video. Codecs are the algorithms for
digitizing analog voice and video so they can be transported in IP packets.
There is an even greater existing selection of standards to choose from today
with more new codecs being invented. Within the traditional wireline world,
voice codecs are typically the ITU G.7xx series and video codecs are the ITU H
series. In addition to the type of codec, there are also different options for
frame sizes -- 10, 20 or 30 ms, for example, which add to the complexity. In
mobile wireless, an entirely different set of voice and video codecs are used
that feature adaptive dynamic support for multiple bit rates to optimize
bandwidth utilization over the radio access network. Lastly, new codecs are
being developed to further improve quality while minimizing bandwidth
utilization. Some of these new codecs include iLBC, iSAC, Speex and
Microsoft's new OCS codecs. Every call between endpoints not supporting at
least one common codec, wireless and wireline phones for example, will require
transcoding.
5. Infinite Bandwidth, QoS and
Signaling Resources End-to-End Will be a Myth Forever
Today's IP networks are
constructed using a selection of different QoS mechanisms, networks links with
different bandwidth and different-sized call, application and media servers.
Voice can't tolerate excessive delay or jitter, so all these resources must
have enough capacity and performance to support a new call. The choices in QoS
mechanisms include IEEE 802.1 p & q, ToS, DiffServ and MPLS. Unfortunately,
these deployed mechanisms don't extend beyond the domain of single IP network
domain. They don't operate on the transit links between providers or on the
access links (T1/E1, DSL, frame, etc.) connecting enterprise or residential
locations to the service provider backbone. These links, regardless of their
size or bandwidth throughput, do have a finite capacity. If a link is at
capacity, and more traffic - just one more call -- is placed on the link, the
quality of all active calls will deteriorate, not just that last call.
Similarly, the servers delivering interactive communication services - SIP
proxies, H.323 gatekeepers, MGCP call agents, NCS call management servers (CMS),
3G IMS CSCFs, softswitches, application servers and media servers - also have
finite capacities for call handling. Consequently, they also face the same
potential overload issues.
6. Some Sessions are more Valuable
than Others
On the Internet all packets
are equal and are delivered on a best efforts basis. Any packet has the same
probability of getting dropped or delayed as any other. In the voice world, we
need the ability to provide special handling for particular calls or sessions.
In the presence of voice server overload, service providers need the ability to
gracefully reject the low value American Idol televoting call to support the
high-value enterprise video conference spanning multiple locations. Emergency
calls, E9-1-1 calls in the US, also need special handling in terms of
prioritization and possibly pre-emption if service resources are oversubscribed.
Lawful intercept capabilities supporting government regulations like CALEA in
the US must always work to support law enforcement in their pursuit of
pedophiles, drug cartels and terrorists.
7. Business Models will never
be Homogeneous
Interactive communication
services are potentially so functionally rich that business models will never be
homogeneous. It's not just voice. It's video and multimedia sessions such as
truly interactive, collaborative white boarding. There are person-person
communications and multi-party conference sessions. Distance learning will push
"conferencing" to a new level of interactivity and control. Sessions will be
available with or without QoS support. QoS can be invoked on a per session basis
or even mid-session via a "turbo" button. And don't forget the
highly-profitable options like directory assistance. Service providers today
compete for this business on the basis of cumbersome, difficult-to-use
automation vs. the ability to talk to a real human. Consequently, any subscriber
"on-net" service (where on-net means the network of a single provider or
federation of providers) will likely have some combination flat rate - $ per
month per person or residence, and variable rate - $ per session, per minute or
maybe even per packet. There might even be a model where you buy interactive
communications services from an ITSP and QoS from your facilities-based
transport provider.
The business relationship
between providers for sessions that need "off-net" support is still a big
unanswered question. How will revenues or costs will be allocated and traffic
exchanged. Will it work like the Internet where big backbone providers
"peer" and exchange equal volumes of traffic and smaller ISPs pay for
backbone connections? If transcoding is required, which service provider pays
for that? How will calls to the PSTN that are still locked into the fundamental
money per minute model of that world be handled?
IC and UC, the Future - the
Federnet
Our network nirvana can
only be realized by connecting IP networks together in a way that enables the
end-to-end delivery of trusted, first class IC/UC to anyone, anywhere, at
anytime. Each of these seven precepts is driving the need for more intelligence,
not less, into our IP networks at their borders. The best-effort, insecure
Internet will never hack it. IC the future is the Federnet. UC?
About
the Author
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Seamus Hourihan, who first
coined the term "session border control," has driven industry
recognition of the critical importance of the new product category
created by Acme Packet. He is a prolific speaker on the conference and
tradeshow circuit and is the author of several networking guidebooks and
numerous articles. Recently, Seamus was recognized by Internet Telephony
magazine as one of the "Top 100 Voices of IP Communications."
Seamus brings over 25
years of experience to Acme Packet in executive management, marketing,
product management and business development roles at voice over IP, IP
networking, web infrastructure and computer companies. Seamus was vice
president of marketing for internetworking leader Wellfleet and, after
its merger with Synoptics, Bay Networks (now Nortel) for nearly seven
years. During his tenure, annual revenue grew from $10 million to $2
billion and Inc. recognized Wellfleet as "America's Fastest Growing
Company" for two consecutive years. More recently, Seamus was vice
president of marketing for Pingtel, widely recognized as a leader in SIP
products and technology. He also held management positions at Data
General, MASSCOMP and Bright Tiger and has operated his own consulting
company. Seamus holds an AB degree from Dartmouth College and an
MBA from Babson College.
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About Acme Packet
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Acme Packet (NASDAQ: APKT),
the leader in session border control solutions, enables the delivery of
trusted, first class interactive communications--voice, video and
multimedia sessions--across IP network borders. Our Net-Net family of
session border controllers supports multiple applications in service
provider, large enterprise and contact center networks--from VoIP
trunking to hosted enterprise and residential services to fixed-mobile
convergence. They satisfy critical security, service assurance and
regulatory requirements in wireline, cable and wireless networks; and
support multiple protocols--SIP, H.323, MGCP/NCS and H.248--and
multiple border points--interconnect, access and data center. Our
products have been selected by over 420 service providers in 81
countries, including 23 of the top 25, and 76 of the top 100 service
providers in the world.
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