October 2000
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Pulling Together -- Interoperability
Through Open Standards
BY GREG GALITZINE
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Go Right To:
SIP... H.323
MGCP...
MPLS... Round Table
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This month's cover conjures up images of teamwork, of pulling together to
achieve a common goal. In our industry, one of the most hotly debated
issues is the state of interoperability between multiple vendors'
disparate equipment or, alternatively, the lack thereof. There are
bakeoffs and testing grounds throughout the industry, and TMC is no
stranger to these events. Our very own ConvergeNet� (on display this
month at Internet Telephony
Conference & EXPO in San Diego; October 4-6) is one of the longest
running, most successful interoperability events around. Our success and
the fact that so many other organizations are interested in proving vendor
interoperability all points to a single fact: Interoperability is
important to the Internet telephony industry.
We've taken a quick pulse of the industry and asked several vendors to
comment on the importance of interoperability. Respondents include an
Internet telephony service provider (ITSP), a next-gen Integrated
Communications Provider (ICP), and one of the so-called "arms
merchants" -- a company tasked with supplying products and solutions
to the industry.
We've also taken a moment to ask the industry's leading players to
describe a standard that falls within their particular area of expertise.
Below you'll find descriptions and definitions of several of the more
prevalent standards operating in the Internet telephony industry: SIP,
H.323, MGCP, and MPLS.
One thread you'll find throughout this whole section is the need for the
industry to work together toward a rather elusive goal: Only by pulling
together can we ever achieve the holy grail of interoperability.
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SIP: Designed For Interoperability
Jonathan Rosenberg, Chief Scientist, dynamicsoft,
Inc.
As next-generation networks move from the design stage to deployment, a
key challenge for service providers and network operators is how to ensure
interoperability between different communication protocols. The Session
Initiation Protocol (SIP) is an Internet standard specified by the
Internet Engineering Task Force that is used to used to initiate, manage,
and terminate interactive sessions between one or more users on the
Internet. SIP has gained tremendous momentum among service providers
because it can deliver enhanced services over next-generation networks.
However, one of SIP's most attractive features is that its simple, open
design supports interworking with ISUP and H.323 -- key protocols from
both the SS7 and IP environments. SIP's ability to work with these widely
deployed standards will not only speed the integration of the PSTN with IP
networks, but gives service providers with H.323 networks a clear
migration path to offer new services that can go well beyond VoIP.
SIP And ISUP
Let's look first at how SIP works with ISUP, the protocol used in SS7
networks to set up and manage basic call connections. One common network
scenario is IP trunking, also known as phone-to-phone, in which PSTN calls
hop onto the Internet over a gateway and then hop back off to the PSTN.
While SIP is used to route calls over the Internet, the question becomes
how to preserve feature transparency. It is generally believed that the
primary way to achieve feature transparency is to ensure that all the
information in the incoming ISUP is carried through the SIP network to the
gateway on the far end.
This can be accomplished with SIP-T (SIP for telephony), a simple
extension of SIP that preserves needed ISUP information as the call is
carried through the Internet. This is done by actually carrying the ISUP
message, byte for byte, in the body of the SIP messages. Since SIP
supports transfer of any type of content, ranging from JPEG pictures to
HTML Web pages, it can easily carry ISUP messages. What's more, SIP-T is
backwards compatible so that it will operate with all existing SIP phones.
If a phone gets a message with ISUP, but doesn't understand ISUP, those
bodies are simply ignored. SIP-T is currently a work item of the SIP
working group in the IETF.
SIP And H.323
How well does SIP interwork with H.323? Since H.323 is an ITU
standard, while SIP springs from the IETF, these two protocols are often
portrayed as rivals. As usual, the real picture is more complicated than
that, and network operators stand to benefit from closer interworking
between SIP and H.323.
Architecturally, these two protocols are very different, and the way
they handle many functions (such as multiparty conferencing) share few
similarities and are difficult to interwork. However, many softswitches
already support SIP to H.323 translations, and interworking between SIP
and H.323 has been demonstrated for some time now. There is currently work
going on within the IETF to develop guidelines for such interworking. It's
already been shown that basic functions like call setup and teardown will
work between these two protocols, but even more complex functions are
being investigated.
One of the key advantages of SIP/H.323 interworking is that a service
provider with deployed H.323 networks will be able to begin to gradually
realize some of the benefits of SIP services. SIP is rapidly emerging as
the clear choice for delivering applications; soon we'll begin to see the
advent of application servers that will likely support SIP and not H.323.
The International Softswitch Consortium's applications working group has
defined a typical VoIP application server based on SIP.
Service providers with H.323 networks can simply begin to use SIP/H.323
conversion boxes, or gateways, that will let them leverage their existing
investments in H.323 gear. They'll be able to deploy SIP application
servers that can work in H.323 networks via conversion boxes. They won't
have access to the powerful new features and services SIP defines, but SIP
itself is powerful enough to "know" what features a H.323 client
is able to execute. SIP-based applications can include negotiation
capabilities that can "step down" in function when talking to an
H.323 endpoint device.
The advantages to this approach are compelling: H.323 operators will
gain access to new applications. And as soon as they deploy a full SIP
gateway or SIP-enabled devices in their network, they will immediately
have access to the complete rich set of SIP services. So there's a logical
migration path here. The incentive for H.323 operators is to gain access
to these more powerful applications, but still be able to support older
H.323 endpoints. This approach lets network operators move up to a
SIP-based infrastructure as part of a planned network upgrade cycle --
protecting their existing investments while gaining access to the better
scalability, openness, and rich features SIP can deliver.
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H.323
Orit Levin, Product Manager, RADVision,
Inc.
Scope
H.323 is an ITU standard designed to accomplish a primary system goal:
moving real-time bi-directional multimedia (audio, video, data, fax)
across packet-based networks while maintaining connectivity to PSTN,
including SS7 circuit-switched networks and H.320 systems. Version 1 of
H.323 was released by the ITU in May 1996, and was designed mainly to
support small-to-medium videoconference systems. As the VoIP community
began to adopt H.323, versions 2 and 3 of the H.323 recommendation added
many features associated with pure VoIP systems, more services, and a
support for large distributed networks. Version 4 of H.323, to be ratified
in November 2000, reflects over four years of deployment experience. In
addition, H.323 Annexes provide specifications for designing H.323 systems
for specific environments or applications.
Building Blocks
H.323 both defines a system and also provides the protocols necessary
to achieve the defined system. The signaling protocols are defined by
H.225.0 and include RAS and Q.931. RAS, or Registration, Administration,
and Status, handles user management; Q.931, an adaptation and optimization
for packet networks of the original ISDN Q.931, handles call signaling;
H.323 also uses the H.245 protocol, which provides media control by
opening the media channels and negotiating the capabilities for each
endpoint. H.245 can also change the characteristics of the channel,
allowing for asymmetrical rate and codec adaptation between endpoints,
features necessary for multimedia interactive communications. H.323 also
uses the IETF RTP/RTCP protocol to move the real-time streams themselves
across the network.
The building blocks of H.323 are Gatekeepers, Terminals, Gateways and
Multipoint Control Units (MCUs). These last three entities are
collectively referred to as endpoints, which generate and/or terminate
call signaling and information streams.
- Terminals are "smart" endpoints that, in addition to call
signaling, can negotiate capabilities and media control.
- Gatekeepers are H.323 Servers that provide Location and Control
services, for various H.323 endpoints within their Zones. Among
services, traditionally provided by a Gatekeeper, are address
resolution, bandwidth management and resolving the Gateway's location
problem. The Gatekeeper is considered to be a control point, through
which the desired policies and back-end services are introduced to the
H.323 system. The Gatekeeper is also a typical place for extracting of
H.323 information, such as billing information and diagnostics.
- Gateways provide for real-time, two-way communication between H.323
endpoints and other endpoints of any kind, such as users connected to
a PSTN network.
- MCUs provide the capability for three or more endpoints to
participate in a multipoint conference. Two terminals in a
point-to-point conference may also be connected by an MCU if the call
is anticipated to later develop into a multipoint conference. H.323
has a concept of breaking the MCU into two distinct pieces -- the
multipoint processor (MP) for processing of multipoint media streams
and the multipoint control (MC) for control of one or many MPs. This
MC/MP split allows use of complementary protocols such as MCGP and
H.248 (MEGACO) to physically distribute the logical MCU components.
Annexes Soup
"Annex C" defines procedures for running H.323 over ATM.
The H.450 series defines how PSTN-like Supplementary services can be
built for H.323 systems.
H.323 has been optimized for VoIP support through "Annex E"
(creating a standard for running H.323 over UDP) and "Annex F"
(providing guidelines for Simple Endpoint Type (SET), allowing for media
types other than voice to be stripped away). "Annex J" defines
security procedures for SET. Additionally, Stimulus Signaling (defined by
"Annex L") allows a user to establish a simple H.323 call and
then communicate with a server via H.248 (MEGACO) procedures in order to
receive or invoke services implemented by the server.
"Annex G" defines procedures for simple text conversation
both for "classic" H.323 and SET devices.
"Annex M" defines a framework for conveying legacy protocols
over the H.323 backbone. Annexes M.1, M.2, and M.3 address QSIG, ISUP, and
DSS1 signaling protocols correspondingly.
H.323 also continues to make strides in mobility and use of the
Internet's improving infrastructure. The original RAS Registration concept
allows for User and Service Mobility built within and around H.323 (H.323
Annexes "H" and "I", and "H.246 Annex E").
"Annex K" introduces a concept of HTTP-based "service
control plane" within a "call control plane" of H.323.
As H.323 prepares for full inclusion as a service being provided over
Internet, work on H.323 procedures over Internet is being performed in the
context of "Annex O" (Internet Protocols and Technologies
complementary to H.323).
Early 2001 should bring ratification of "Annex N" (QoS in
H.323) and "Annex R" (Robustness). Based on deployment
experience, several important annexes are set for near-term second version
ratification. Among them are H.235 v2 (Security for H.323), "Annex
D" v2 (FAX over H.323), "Annex E" v2 (H.323 over UDP) and
"H.225.0 Annex G" v2 (Inter-Domain Communication).
Final Word
With four years of deployment experience and maturity behind it, the
H.323 protocol is stable and the most deployed solution. H.323 is being
used for a variety of multimedia applications such as Call Center (voice
and video), VoIP trunking applications, remote education and other
vertical business applications. H.323, as the most mature specification,
is the standard of choice for applications that require multi-vendor
components. H.323, together with protocols and toolsets such as MCGP, SIP,
and H.248, provides a robust framework for enhanced multimedia
solutions.
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MGCP And Its Variations
Jeff Lawrence, President & CEO, Trillium
Digital Systems, Inc.
Gateways reside at the boundaries between and within the telephony,
Internet, broadband, and wireless network infrastructures. Trunking media
gateways operate at the boundary between the Internet and the PSTN, and
perform the conditioning of circuit- and packet-based media streams. Other
types of media gateways perform similar functions at the boundary between
businesses, residences, access points, and the network.
Gateways may be decomposed into the Media Gateway Controller (MGC),
Signalling Gateway (SG) and Media Gateway (MG). The SG acts as the relay
for signalling information between the Internet and PSTN. The
"intelligent" Media Gateway Controller maintains the overall
call state and controls the "dumb" Media Gateway by sending
commands via the Media Gateway Control Protocol (MGCP), a fairly simple
protocol consisting of a small set of messages and procedures. The MG
executes the commands given by the MGC to control and condition the
circuit and packet media streams. Currently, the intelligence controlling
the Internet (in the form of name servers, gatekeepers, and media gateway
controllers) and the PSTN (in the form of service control platforms and
location registers) is distributed throughout the network. In the future,
the access and core network intelligence will reside in softswitches and
backend application servers. These softswitches will incorporate MGC
functionality and control the network elements using a variety of
protocols including MGCP.
MGCP and its related protocols have undergone a swift market driven
evolution. Unfortunately, as a result of differing market needs the IETF,
CableLabs, and the ITU have fragmented the MGCP standard into four
variations: MGCP, NCS/MGCP, MEGACO, and H.248. There are a number of
differences between these variations and there can even be different
implementations within the individual variations. This has created
confusion in the market and raised a number of interoperability and
interworking issues for communications equipment manufacturers and their
customers, the network operators and service providers. MGCP is driven by
the IETF and supports voice connections, while NCS/MGCP is driven by the
CableLabs PacketCable initiative and is designed to support multimedia
connections and operate efficiently in the cable infrastructure's
multipoint environment.
MEGACO and H.248, on the other hand, are essentially two names for the
same thing. Though these started as separate initiatives, the IETF MEGACO
Working Group and ITU Study Group 16 have agreed to align their activities
and publish a single document for MEGACO/H.248, which promises a much
richer set of capabilities, with a corresponding increase in protocol
complexity. The transaction model has changed from endpoints and
connections to contexts and terminations, which in turn will provide a
greater deal of transaction flexibility. MEGACO/H.248 uses text or binary
messages and can support multiple actions and commands per transaction.
Importantly, MEGACO/H.248 also supports improved failover procedures to
provide high availability in case of equipment or network failure. In
addition, MEGACO/H.248 provides easier-to-use and extensible packages to
support the events, signals, and statistics necessary to control and
manage the MGs. Unfortunately, these capabilities come at the expense of a
command and response structure that is different from that of MGCP and NCS/MGCP.
Due to these differences, MGCP is not completely compatible with NCS/MGCP
and neither of these is compatible with MEGACO/H.248. Although MGCP and
NCS/MGCP have dominant market share at this point in time, new equipment
and devices are being developed to support MEGACO/H.248. The
incompatibilities among MGCP's different variations will require the
development of gateways to translate between the installed bases of the
different variations. In addition to the differences in the variations,
there are differences in the individual implementations, which means each
softswitch will need to support the specific command and response sets of
the MGs under its control.
In the short term, interoperability problems can be alleviated if a
single vendor supplies the MGC and MGs. This, however, is not a long-term
solution. From a risk management and competitive perspective, network
operators and service providers will find it essential to move towards
multi-vendor solutions that will provide the flexibility to choose the
best feature, performance, and cost solutions to meet their particular
needs. However, true multi-vendor solutions will not be possible until
more extensive and industry-wide interoperability testing is performed.
The multiple standards and the limited number of interoperability tests
performed to date (most of which have been sponsored by CableLabs, the
University of New Hampshire, and the International Softswitch Consortium),
highlight how far the industry still must go to ensure the long-term
success of MGCP and the decomposed network architecture.
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MPLS: Enabling Enhanced IP Services In
The Access Network
Mark Veil, Product Manager, Integral
Access
The demand for converged IP-based services and applications has created
a host of new market opportunities for today's Integrated Communications
Provider (ICP). With each new market having its own unique set of demands,
the ICP must be capable of creating, delivering, and billing for graded
service offerings specifically tailored for these requirements.
As a result, today's providers need tools to effectively and
efficiently engineer and manage their networks for class and quality of
service (CoS/QoS). Fortunately, new IP-based technologies for supporting
enhanced broadband voice and data services over a single access network
can provide competitive carriers the opportunity to aggressively attack
new markets with broad and richly differentiated service offerings.
Multi-Protocol Label Switching (MPLS), in combination with effective
traffic management facilities, provides the best approach for managing QoS/CoS
in today's access networks and allows ICPs to deliver differentiated
IP-based services.
Engineering And Managing Network Traffic For QoS
QoS, which we will define as the collective measure of service levels
delivered to the customer premise and characterized by performance
criteria such as availability, response time, throughput, error rates, and
packet loss, enables certain traffic types to be given priority treatment
in the network. This capability is required to support specific latency
requirements of individual applications, or to meet customer-contracted
Service Level Agreement (SLA) obligations. To support QoS requirements,
network facilities must provide traffic management services such as
admission control, traffic classification, traffic marking, and traffic
conditioning. Once appropriately classified and groomed, traffic must be
aggregated and efficiently mapped onto the existing network topology for
the purposes of controlling network behavior and optimizing network
utilization and traffic performance. This is the domain of traffic
engineering and MPLS.
MPLS represents the best alternative for enabling traffic engineering
and QoS in the heterogeneous world of network communications. Although
originally intended as a means to enhance routing performance, continued
improvements in that area have shifted the application focus of MPLS to
its inherent capabilities for delivering efficient and scalable traffic
engineering and QoS in IP-based networks. MPLS operates at Layer
Two-and-a-half (L2.5) and is protocol-agnostic to the layers above and
below it. Its architecture is based on the multi-layer switch concept --
which cleanly separates the forwarding and control functions --with the
former being the domain of MPLS.
The power of MPLS stems from its ability to associate and allocate any
type of user traffic with a particular Forwarding Equivalency Class (FEC).
Each FEC represents an aggregation of traffic that will be treated in the
same manner as it traverses the network. Typically these traffic
classifications are based on the contents of the packet header, such as
the L2 and L3 source/destination address. In practice, however,
classifications may be derived from (and applied to) a virtually unlimited
range, combination, and granularity of packet attributes -- including
physical ingress port/interface, application protocol type or Ipv4 type of
service (ToS), and Ipv6 CoS markings.
These FECs are then mapped to label switched paths (LSPs) that have
been engineered to support specific traffic QoS requirements -- guaranteed
bandwidth or low latency for instance. LSPs behave in a similar fashion to
the more familiar ATM virtual circuit and frame relay data link connection
identifier (DLCI), but do so with much greater efficiencies.
Upon ingress to a particular MPLS domain, all packets are assigned a
label that serves to represent its FEC/LSP binding as well as a short-hand
reference to the contents of the IP packet header. In sharp contrast to
today's "longest-match" routing paradigm, MPLS-equipped routers
are able to perform ultra-fast forwarding of IP packets via "exact
match" label swapping. Moreover, MPLS overcomes the inherent
limitations of traditional destination-based routing by supporting both
explicit and constraint-based approaches for establishing LSPs. This
capability allows network administrators to bypass potential points of
congestion and direct traffic away from the default path selected by
today's Interior Gateway Protocol (IGP)-based networks and deliver precise
control of network traffic and behavior.
Enabling Enhanced Services With MPLS
For the resource-constrained local access loop, MPLS is best matched
with the IETF's Integrated Services (IntServ) architecture to provide
explicit QoS controls at the LSP level for delivering enhanced IP-based
services. The MPLS-IntServ combination delivers connection-oriented
behavior to connectionless access networks. Each LSP is apportioned and
allocated access bandwidth in accordance with its traffic contract
parameters to ensure the given application receives its appropriate level
of QoS. For instance, individual LSPs can be created to support the unique
requirements of services such as Voice over IP (VoIP), VoMPLS, or premium
data applications such as e-commerce or VPNs. In the core network, where
explicit end-to-end resource reservations are not practical and additional
packet processing overhead is not advantageous, pairing MPLS with another
IETF initiative, differentiated services (DiffServ) provide the required
levels of QoS management.
MPLS offers a practical QoS solution for ICPs that want to deliver
differentiated, IP-based services. MPLS enables all services --
packet-based voice services, tiered data offerings, and secure VPNs -- to
dynamically share the same link, delivering optimum resource utilization
for resource-constrained access networks. Moreover, service providers can
provision, account, and bill for differentiated services -- and turn
previously "flat" minutes into "fat"
revenue-generating minutes. Finally, MPLS promises to reduce the
complexities and costs associated with operating multiple access networks
and provides protection for existing technology investments, while
allowing carriers to follow a clear migration path to a converged,
IP-dominated world.
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Round Table -- ITXC, Nortel,
ICG...
We asked several industry-leading vendors for their
views on the Internet telephony industry. Their responses appear below.
A.)
How important is interoperability to the health and viability of
the Internet telephony industry?
B.) What are
the major hurdles to successful interoperability?
C.) What are
the major benefits of interoperability to service providers? To
business or residential customers? |
Jim Yu, Director of New Technologies, ITXC Corp.
A.) Interoperability is essential to the growth of the Internet
telephony industry. Without it, service providers have two choices: Either
they are forced to operate multiple networks -- a network for each type of
equipment -- or they are forced to operate their network relying on a
single vendor. Operating multiple networks is difficult to manage and not
cost effective to build out. Relying on a single vendor locks the service
provider into a single upgrade and migration path, and can not take
advantage of additional features or cost competition offered by other
vendors.
B.) The hurdle that the industry faces today is the lack of a
single interoperability reference to implement against. One organization
that is actively addressing this issue is the IMTC's iNOW! Activity Group.
iNOW!, which stands for "interoperability NOW!," is a
broad-based, industry initiative to quickly provide short term,
revenue-based IP telephony interoperability. A set of profiles were
developed to guide the implementation of VoIP equipment to ensure that
equipment built to the iNOW! specification will be able to exchange voice
and fax traffic seamlessly across vendor platforms.
C.) Interoperability in the VoIP space will enable service
providers to rapidly expand their footprints with customers that have
already bought equipment from multiple vendors. Service providers will
also easily expand their footprint by connecting their multiple vendor
networks to other multiple vendor networks.
Al Bender, Vice President, Applications Portfolio, Nortel
Networks
A.) The promise of Internet telephony is an open and programmable
environment. Open means standards-based and interoperable. With
interoperability, service providers will be able to mix best-in-class
products from different vendors to differentiate service. Interoperability
is less important at the gateway level. In general, gateways are dumb and
are becoming commodities. The key value of interoperability is at the
applications level. Next-generation multimedia services will be delivered
in Internet telephony architecture from SIP-based service platforms that
need to interoperate with the softswitch and the smart clients.
B.) The major hurdle is the completion of relevant standards:
H.248 with support for both lines and trunks, IPS7 for SS7, SIP-T for
softswitch-to-softswitch signaling, and native SIP for next-generation
services.
C.) The major benefits of interoperability are lower cost,
faster deployment, and differentiated services.
Mike Kallet, Executive Vice President, Products & Technology,
ICG Communications
A.) Critical. Only with interoperability can Quality of
Service (QoS) exist across different networks. This hasn't happened yet.
The hardware vendors -- the Ciscos, Lucents, and Nortels of the industry
-- must design and implement according to agreed upon standards. Hardware
vendors must test with each other and listen to companies (like ICG) so
they know what does and does not work. Providers like ICG must pressure
vendors to create and adhere to standards. With the industry calling for
standards, hardware vendors will be pushed to take that first and
important step toward interoperability.
B.) If a supplier can sell their goods with proprietary
technology, it has no incentive to design a product replaceable by
products from other companies. Even once standards are set, companies will
extend those standards, differentiating themselves and losing
interoperability. The thin line companies must walk is to adhere to
industry standards while differentiating themselves to consumers.
Another issue is complexity of networks. Multiple standards must unite
so a diversity of services can run on the network. We must offer QoS
across different technologies.
Finally, we are reinventing settlement issues -- reciprocal
compensation, tariffs, etc. The industry must determine billing and
revenue issues in light of regulatory mandates.
C.) For providers, the benefit is simply industry growth and
additional telephony purchases by customers. For customers, the benefit is
simplicity. The customer won't worry about QoS or the networks on which
traffic travels. Unless interoperability is achieved, IP telephony will
not thrive; there is too much information for the consumer to keep track
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