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Convergence /

VoIP makes strides

Advanced features gain support and first H.323 to SIP interconnections demonstrated, but more work needs to be done.

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Miercom

Our latest round of interoperability tests shows that voice-over-IP products, which could only work together at basic levels last year, can now support advanced telephony features like call holding and call transferring. And in the first public test of products supporting competing voice-over-IP specifications, H.323 to Session Initiation Protocol internetworking was successfully demonstrated. This will let users with embedded H.323-based voice-over-IP equipment more easily roll out enhanced services, such as real-time messaging, that other protocols allow.

We've carefully documented voice-over-IP interoperability over the past 18 months (see VoIP vendors pass SIP test and Voice over IP is a (fast) moving target) because having point products work together end to end is the only way enterprise customers can use them as a replacement to existing telephone systems.

In this round, we invited vendors with products based on the International Telecommunications Union's H.323 and/or the IETF's SIP, challenging them to prove interoperability based on the same protocol but also between H.323 and SIP.

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Seven vendors answered the call - Avaya, Cisco, Dynamicsoft, Mediatrix, Pingtel, Quintum and Siemens ICN - with a total 14 products. Siemens, Dynamicsoft and Cisco ventured into the H.323-to-SIP internetworking arena with successful results.



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During the interoperability tests, we used Agilent Technologies' Agilent Advisor Version12.0, running IP Telephony Analyzer software, to help identify coding problems that frustrated interoperability.

On the H.323 side

Although Versions 3 and 4 of the ITU's H.323 umbrella specification have been available for more than a year, vendors tested with H.323 Version 2 because they were more comfortable with it.

Vendors testing H.323-based products didn't have serious problems interoperating at basic levels, but getting gatekeepers - complex, high-level call-control hardware and software combinations necessary for interdomain communications - to work together was still not ubiquitous.

In our tests a year ago, gatekeeper interoperability was almost unheard of because while these products may be standards-based, vendors have their own, often subtle, differences in implementing the H.323 spec that prevent interoperability.

This year, the Quintum Tenor gatekeeper and the Avaya IP600 gatekeeper worked with each other but neither could interoperate with the Siemens HiPath 5000 gatekeeper. We discovered that unless each gatekeeper supports one particular protocol feature - the H.323 Location Request/Location Confirmation (LRQ/LCF) message - they couldn't interoperate. Siemens tested its gatekeeper with a product release that did not support LRQ/LCF but says it plans to support LRQ/LCF in its next release.

Codec negotiation, a stumbling block last year, was a nonissue.

Advanced H.323 features

An important aspect of voice-over-IP interoperability is the delivery of features to IP phones. Previously, we've seen little support for the H.450 standard, a spec defined under the H.323 umbrella that details a rudimentary set of telephony features, such as call forwarding and call hold.

This time around, Siemens and Mediatrix demonstrated interoperability of numerous H.450-based services: call hold, call transfer, call busy, forward on hold, forward on busy and forward always. The Siemens products also support message waiting, but Mediatrix did not. The fact that Siemens and Mediatrix have a collaborative business partnership undoubtedly influenced their interoperability success.

With one notable exception in our group, H.323-supporting vendors have not been very aggressive in implementing the H.225 security standards, which also falls under the H.323 umbrella spec. Avaya provided a customized version of its IP600 IP PBX for the interoperability test bed, in which its inherent authentication feature - one security component under H.225 - was disabled. Avaya did so to interoperate with other H.323-based products on a wider set of overall features.

Implementation of the T.38 spec for faxing over an IP-based network - another spec defines under the H.323 standard - was nowhere to be seen in this latest round of interoperability testing. This isn't surprising because use of traditional fax machines is diminishing overall.

Mediatrix, Avaya and Quintum successfully demonstrated dynamic fax detection with various codecs and showed that they could automatically return to a normal voice path once the fax transmission was completed. But interdomain faxing between disparate voice-over-IP gatekeepers, the environment where T.38 applies, remains an open issue.

Quintum was the only vendor that successfully demonstrated tunneling based on H.245, which was verified by packet traces done on the Agilent Advisor test tool. H.245 defines tunneling as an H.323-based feature that conserves logical ports. This is appealing to service providers and carriers, but difficult to implement.

The setup of dialing plans, which is not specifically an H.323 issue, continues to frustrate basic interoperability. Vendors have different design philosophies in how they implement dialing plans because of ambiguities in the Q.931 specification (which addresses signaling in the H.323 network). This ambiguity clearly affects vendors' ability to interoperate out of the box.

Quintum's gatekeeper looks for a dialed number when setting up a call, while Siemens' gatekeeper lets its end points handle Q.931 signaling and instead looks for source and destination IP addresses to set up a call. So while the Siemens optiPoint phones could register with the Quintum gatekeeper, the reverse didn't work as well. Dialing plan issues also frustrated Mediatrix's ability to achieve out-of-the-box interoperability. Mediatrix products expect to see five-digit numbers, while other products expect seven-digit numbers. Once these issues were resolved with software patches, basic call set up and completion quickly occurred. But issues of this nature can be time-consuming to analyze and fix.

On the SIP side

We took Cisco 7960 IP phones with SIP loads through their paces against our SIP reference test bed, consisting of the Dynamicsoft SIP Proxy Server and SIP Location Server (the latter registers SIP users on the network) and Pingtel xpressa IP phones. The Cisco phones sailed through basic and advanced tests without incident, highlighted by a four-way conference call with two Pingtel xpressa phones and two Cisco 7960 phones, all registered to the same Dynamicsoft SIP Proxy Server.

We asked several vendors that we believed were either developing or shipping H.323/SIP gateway products to participate in this aspect of the interoperability testing. Only Siemens was willing to go on the record with its H.323/SIP Interworking Gateway.

Siemens' gateway is collocated on the same machine with the Siemens HiPath 5000 gatekeeper. A Siemens optiPoint 400 phone registered on Siemens' gatekeeper successfully placed a call to an analog phone on our simulated public switched telephone network via the Dynamicsoft SIP Proxy Server and Cisco's 5400 SIP-based IOS Voice Gateway. Voice quality was excellent.

This test constitutes a small first step, but a significant portent of things to come. Efficient H.323-SIP conversion is an important step on the path to ubiquitous voice-over-IP interoperability, holding the promise of extending SIP-based services and applications common in the carrier space and applications to existing H.323 networks common in enterprise deployments. SIP is maturing to the point where it's delivering on services, such as real-time messaging and Òfollow-meÓ applications, which are attracting more widespread use.

The downside to H.323-SIP internetworking is that H.323 and SIP are vastly different protocols, and interoperation is difficult. A recent IETF draft specifically addresses SIP-H.323 internetworking. This lengthy (124-page) document begs the question: Do we want SIP to do more than originally intended? The answer appears to be yes, but this necessarily adds to its complexity.

Jonathan Rosenberg, chief scientist at Dynamicsoft and co-developer of SIP, says it is a common misconception that SIP and H.323 can be internetworked perfectly. While basic phone calls and some simple features will internetwork - because they are common to both protocols - more advanced features, such as integrating e-mail and Web services into voice/video, won't.

What's next

Voice-over-IP products continue to mature, and vendors are taking interoperability among H.323 and SIP products more seriously as the demand for voice-over-IP-enabled services, such as real-time messaging and follow-me applications, increases. Many leading vendors, such as Cisco and Siemens, already incorporate both H.323 and SIP in their products, and others are working on them.

This year, look for increasing internetworking between H.323 and SIP products as greater acceptance of voice-over-IP among consumers fuels demand in the workplace, and generally greater acceptance of voice over IP among consumers.



Vendor Product VoIP Protocol(s) Supported
Avaya IP600 Internet Protocol Communication Server, R9.5; VoIP gateway and gatekeeper combination H.323
Cisco Cisco IOS Voice Gateway 2600, v12.2 (4)T1, VoIP gateway and gatekeeper; Cisco IOS VoIP gateway 5400, v12.2(2)XB; IP Phone 7960; Cisco SIP Proxy Server (CSPS), v1.2 H.323 and SIP
Dynamicsoft SIP Proxy Server, v5.2, and SIP Location Server, v4.0 (reference platform in SIP test bed) SIP
Mediatrix 1101 and 1104 H.323 Terminals, v2.1; 1204 VoIP Gateway, v2.1 H.323
Pingtel xpressa IP phones, v1.2.1.0 (reference SIP phones) SIP
Quintum Tenor VoIP MultiPath Switch; Tenor A400 4-Port Analog (gateway); Tenor D2400 24-Port T1 Digital (gateway); tested P3-1-6 software; all products support embedded gatekeeper H.323
Siemens OptiPoint 400 Standard IP phones, v3.0.20; HiPath 5000 gatekeeper, v3.0; H.323-SIP software-based gateway, collocated with HiPath 5000 H.323 and SIP (H.323 tested)
Test Equipment:
Agilent Agilent Advisor,v12.0, running IP Telephony Analyzer software; Voice Quality Tester, v4.1 H.323

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Yocom is senior editor, Percy is a senior technology analyst, and Frigo is a lab test engineer at Miercom, a network consultancy and product test center in Princeton Junction, N.J. They can be reached at byocom@mier.com, kpercy@mier.com and mfrigo@mier.com. Dama is an independent voice-over-IP consultant and can be reached at kdama@worldnet.att.net.


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