By Joanna Elachi,
CommWeb.com
Nov 27, 2000

When CommWeb set out to test multiple vendors of so-called "IP PBX" vendors, we originally imagined we'd do some interoperability testing between different platforms. But as we talked to vendors, it became clear that standards remain very much in the air and that today's climate of multiple protocols is retarding voice-over-IP (VoIP) deployment. We also thought it would be a good idea to take inventory of the three major standards that are presently being debated as candidates for voice-over-IP signaling: H.323; MGCP/Megaco; and SIP. H.323

H.323 defines packet standards for terminal, equipment and services for multimedia communications over local and wide area networks communicating with systems connected to telephony networks such as ISDN. It evolved in the International Telecommunications Union from a series of videoconferencing standards. The initial version was adopted in June 1996 and addressed communication over IP-based local area networks (LANs). Version 2, adopted in January 1998, extended the protocol for wide area use and general purpose IP networks.

As it exists today, H.323 is an "umbrella" specification, covering several sub-protocols related to call setup and signaling. Chief among these are H.225, which defines the "call signaling channel", H.245, or the "call control channel", and RAS - registration, admission, status. Underlying these are the Real-Time Transport Protocol (RTP) and/or the Real Time Control Protocol (RTCP), which define the basic requirements for transporting real-time data over a packet network.

H.323 handles four major components for a network-based communication system: Terminals, Gateways, Gatekeepers and Multipoint Control Units. Gateways and gatekeepers help negotiate the PSTN interconnection, while multipoint control units (MCUs) enable multiparty audio and videoconferences.

Gateways make it possible to use standard telephones to talk over the Internet instead of multimedia computers. Gateways also handle addressing problems -- a significant issue in IP telephony. To call another multimedia PC user, you must have their Internet Protocol (IP) address. To call someone using a gateway service/product, you need only dial their phone number.

Gatekeepers are database servers that provide address translation and in some cases bandwidth management by mapping telephone numbers and IP addresses. To have a functional IP network that will handle call traffic originating or terminating at regular telephones, gatekeeper services must be used.

One of the major criticisms levied against H.323 is the time and complexity involved in setting up a call. First of all, the protocol uses multiple roundtrip messages to establish signaling and control for any call between two terminals. Moreover, H.323 requires that TCP connections be used to carry the messages, requiring an additional roundtrip exchange. The recently released version 3 is an improvement and includes a "Fast Connect" procedure that effectively consolidates the Q.931 messages exchanged between terminals and a tunneling procedure that lets H.245 share a single TCP connection with Q.931.

More information about H.323 can be found in the H.323 Primer and at the ITU-T home page.

MGCP/Megaco

The Media Gateway Control Protocol (MGCP) specifies communication between call control elements and telephony gateways. It was conceived partly to address some of the perceived inadequacies of H.323 at the level of centralized network infrastructure. MGCP, in its current form, is a combination of two earlier protocols, SGCP and IPDC.

The International Engineering Task Force (IETF), through its Megaco working group, is working on a standard that uses the same architecture and baseline as MGCP, but supports ATM.

MGCP's central goal is to remain simple. It puts call signaling, control and processing intelligence in call agents or media gateway controllers. Media gateways are telephony gateways that serve as multi-service packet networks, converting audio signals and data packets. They include trunking, voice over ATM, residential, access and business gateways, network access servers and circuit switches.

The MGCP call agent performs all the same call routing functions as a gatekeeper in H.323, but has much tighter control. It is a master/slave protocol, where the gateways are expected to execute commands sent by the call agents.

Megaco, also known by its ITU designation, H.248, leapt off the starting block at the first Megaco/H.248 Interop event at the University of New Hampshire in August 2000. The event, sponsored by the Multiservice Switching Forum (MSF), the International Softswitch Consortium (ISC), and the UNH Interoperability Lab, united more than 45 representatives from 17 participating vendors, and was the first major step in validating the proposed Megaco/H.248 protocol standard.

Participating vendors tested a range of Megaco-compliant Media Gateways (MG), Media Gateway Controllers (MGCs), parsers and test boxes. The "bake-off" involved placing calls between two nodes of a single gateway as well as between nodes on separate gateways. All tests included tests of media flow. Most implementations used real-time transport protocol (RTP) on an Ethernet network.

More information about Megaco/H.248 can be found at the IETF Megaco home page.

SIP

Session Initiation Protocol (SIP) is an application layer control protocol for creating, modifying and terminating sessions with one or more participants. Like MGCP, it is text-based. Work on SIP began in late 1996 as a way of inviting users to join Mbone sessions. It was not until 1998 when the spec was approved as an RFC by the Internet Engineering Task Force (IETF) that SIP began to gain acceptance as an IP telephony protocol.

SIP uses a "request-response" model. This is the same structural model used by HTTP. Unlike MGCP, a SIP call can be initiated and completed strictly between two clients, without the mediation of a call agent. To initiate a session, the caller sends a request to a callee's address in the form of a simple text command, and the callee responds with an acceptance or rejection of the invitation. The call will usually be mediated by a proxy server or a redirect server for routing purposes.

More information on SIP can be found at the SIP Forum website and at http://www.cs.columbia.edu/sip.

The Vendor Perspective

We talked to several IP telephony vendors to get their input on where the three VoIP standards are headed, and the advantages and drawbacks of each in implementation. Our small sample exemplified the diversity of the debate, holding positions ranging from non-participation to vehement commitment.

• 3Com
  Ikhlaq Sidhu, Vice President of the Internet Communications Group, told us that "five years ago, when the momentum for H.323 began, everyone was generally under the impression that the problem was solved and that no other standard was needed. But it turned out that as companies started to go after more advanced implementations and solutions, H.323 just wasn't flexible enough."

Instead, 3Com is focusing its development efforts on SIP. In their opinion, a central protocol like SIP is more reliable, more scalable, and enables faster feature development. Since it is so much a cousin to Internet protocols like HTTP, it is very easy to add features integrated with the Internet.

For example, Fed Ex could offer a "share the moment" service whereby when a package has been delivered, the delivery person's confirmation that it has been received would trigger a simultaneous call to both the sender and the recipient over the Internet.

Because of its scalability, SIP is best used in a wide area network (WAN) and less prevalent in a LAN PBX. MGCP's controller-based approach makes more sense in a LAN where scalability isn't much of an issue. In fact, in larger scale WANs, the trend seems to be going toward MGCP-controlled softswitches using SIP interfaces.

In Mr. Sidhu's opinion, SIP also allows sophisticated interoperability very quickly.

"Two years ago, six or seven vendors participated in the first SIP interoperability bakeoff. Within a day, they had most of their products completely interoperable. There were close to 50 at the last bakeoff and they also experienced similar results," Sidhu said.

"The other exciting thing about SIP is that if there is a problem, it's very easy to fix because all the commands are written in ASCII so any human can read the stream -- you don't need sophisticated debugging equipment."

• Avaya
  Avaya got into H.323 early in the game and continues to mainly support that protocol. Jim Coffman, Director of Advanced Sytems, told us that although H.323 is somewhat complex, it is ISDN-based and therefore it is relatively easy to build applications across it. On the other hand, SIP is lightweight with application layer routability and allows feature service to be more conveniently shared across feature servers.

Avaya is in the process of building a SIP stack because it is becoming clear that long distance vendors are considering SIP as the interface into their networks, and SIP will be supported by Avaya products in about a year.

"It will be interesting to see if SIP lives up to its promise and allows vendors to build feature-rich applications," Mr. Coffman said. "Avaya really tries to stay out of the protocol wars, but I can't help but worry that with all the rhetoric around SIP, some of the thinking around VoIP is being critically delayed."

• ESI
  Ken Teaff, ESI's IP Product Manager explained that their downloadable telephone supports proprietary protocols, but none of the major standards. For now, they're just waiting to see who wins the protocol wars. When one single standard becomes accepted, they plan to implement that standard into their products quickly. Speculation around ESI, however, is that things will probably move toward SIP.



• Shoreline
  Barry Castle, VP of Product Marketing, explained that H.323 has too large of a protocol stack to economically implement a telephone call -- it is just too complex and the market doesn't need it. SIP is a much easier protocol to deal with, and Mr. Castle speculates that this will encourage a massive increase in the ability to provide services. The ideal protocol pair will likely be SIP for exchange between softswitches and gateways and MGCP for the signaling to the handset.

Since enterprise customers inevitably have a phone next to a PC, Shoreline is tapping into that resident intelligence and links the phone to the PC using TAPI. Their Distributed Internet Voice Architecture (DIVA) software uses ShoreGear IP voice switches to distribute voice communications throughout the network.

Shoreline's focus is on open protocol support and ease of plug and play. Internet generation protocols, they say, lend themselves to being able to do that.

"IP telephony is the future, and we need protocols that were designed from the get go to be Internet aware," said Castle.

• Sphere
  "H.323 has a lot of legacy, so our products have the fundamentals to support it, but the next generation won't be H.323," explained Kurt Jacobs, Director of Marketing. "Just like everyone still has to support analog, we will have to support H.323 for many years to come."

SIP, in his opinion, is a lightweight protocol and very attractive as far as integrating with other voice apps. It is also extensible, but with that variation may come some interoperability problems.

MGCP is a more appropriate replacement for H.323. It's heavier and requires more processing power, but it works well for managing gateways because it doesn't have much variation. You don't want variation when you're dealing with hardware because then you'd have to constantly be re-releasing updates. Since Sphere makes its own hardware gateways and manages those gateways from central controllers, MGCP is their standard of choice.

Sphere products support H.323 for legacy, have the foundation for MGCP and are presently incorporating SIP.

• Tundo
  Tundo doesn't want to involve itself in the holy war over which standard is better -- they're letting the market to decide.

"Whatever emerges as the popular standard is going to be the one we go with," said David Dines, Director of Corporate Marketing for Tundo. "We're not going to get involved in speculating which is the 'better' standard, we'll go with the market."

Tundo products are H.323 compliant now, but they are working on supporting SIP. Dines speculates that SIP will have the same proprietary implementation issues that H.323 has had, but they are easily solved by a little tweaking to the bit stream.

At the core of the Tundo Network Telephony System (NTS) is the Tundo Distributed Open Telephony -- Operating System (DOT-OS). The DOT-Server supports H.323, H.450, TAPI, CSTA and SNMP. Megaco, SIP, XML will be supported in the future.

• Vertical Networks
  Scott Pickett, CEO and Vice President of Vertical Networks gave us his take on MGCP. Due to a need for a carrier-class standard, IPDC emerged first as a simple, voice-only VoIP standard, he said. This served as the baseline for MGCP, which has been adopted as the standard to use for the softswitch environment. Because MGCP doesn't handle key things like video and ATM encapsulation, Megaco/H.248, which uses the same packages as MGCP, is emerging to fill that gap.

"A couple of years out, we'll still see H.323 and MGCP. But in the most advanced equipment, SIP and Megaco will gracefully co-exist -- with SIP as the interface between media gateway controllers and Megaco working between the call agent and the media gateway, and between the call agent and the device," explained Pickett.

Vertical Networks participates in two interoperability forums -- the IMPC which provides legacy H.323 testing (Vertical still wants to be able to support H.323 products even though they're more expensive) and the ISC. The ISC organizes interoperability testing functions hosted by members and working groups and works to define implementation guides.

On The Wire

We thought it worthy of note that ComGates has developed the CMG/CSS 2000 Softswitch, which provides multi-protocol capability by automatically switching traffic between all existing standards. The idea is the CMG/CSS 2000 gives telephony networks and carriers deploying H.323, SIP, MGCP and Megaco interoperability so they can exchange traffic with one another. It adapts to the protocol being used by the connecting gateway and can also simultaneously run multiple transmission protocols on the same operating platform.

"In the current VoIP environment there are no consistent standards and no real indication as to which of the conflicting protocols will become the dominant choice in the future," said Jacob Tirosh, President and CEO of ComGates.

"The architecture of the CMG/CSS 2000 Softswitch relieves decision makers from committing to a particular protocol and leaves the door open for incorporating new protocols as they become popular. Our.. on-the-fly adaptability of the Softswitch provides carriers and networks with an enabling technology that ensures they can deploy systems in a timely manner with the confidence that their underlying infrastructure will remain up-to-date."