Understanding SIP�Part VI: Testing SIP Interoperability

Despite its status as a standard, some compatability problems exist for SIP. Fortunately for IT managers, a number of resources exist for testing a resolving these issues.

By Mark A. Miller | Posted Jun 21, 2005
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This tutorial will conclude our systematic study of SIP, and discuss test plans that have been developed for SIP networks. Here are links to the preceding articles on SIP, for your easy reference.

The large proliferation of Internet Protocol (IP)-centric networks, plus the ease with which SIP-based solutions can be integrated into that IP-centric network, makes for one of those "good news/bad-news" stories. The good news is that it is easy to find products that support SIP. For starters, check out www.sipcenter.org, where you will find traditional voice equipment manufactures including Avaya, Ericcson, and Nortel Networks; networking manufacturers including 3Com, Cisco Systems, Hewlett-Packard, Lucent Technologies, and Sun Microsystems, plus many, many others. The bad news is that since all of these products were independently developed, the likelihood of a seamless, problem-free multi-vendor integration is relatively small (review the tutorial on H.323 Interoperability Testing).

Fortunately, there are several good resources that can assist the network manager with the SIP interoperability and system testing challenge. The first solution (and probably the easiest) solution is to purchase all of your SIP hardware, software, and applications from a single vendor, such as one of those firms noted above. This solution yields two results: first, you make one salesperson very happy with a (presumably) large contract; and second, you put the responsibility of interoperability testing right back on the shoulders of that chosen vendor.

Unfortunately, the reality of existing systems and resulting amortization schedules, plus the economic challenge of a fork-lift upgrade, frequently render such a single-vendor solution impractical. This leaves the network manager responsible—at least in part—for planning a network implementation that includes subsystems from multiple vendors.

The good news is that some of the SIP interoperability testing work has already been outlined by several organizations.

The first resource is the International Multimedia Teleconferencing Consortium (IMTC), which has developed a SIP Interoperability Test Plan that is available at http://www.imtc.org/interops/. This plan details five different testing scenarios:

  1. Point-to-Point Audio/Video with no Servers, which verifies the INVITE, OK, ACK and BYE messages that we examined previously.
  2. Endpoint Registration with a Registrar, which verifies the endpoint registration functions.
  3. Point-to-Point Audio/Visual Call using Redirector, which verifies the redirected signaling functions.
  4. Point-to-Point Audio/Visual Call using Proxy, which verifies that a proxy server can pass call information between two domains.
  5. Point-to-Point Audio/Visual Call using Gateway, which verifies that a SIP phone can access a telephone on the other side of a gateway, such as a Public Switched Telephone Network (PSTN) gateway.

The second resource is Columbia University (New York, NY), which sponsors SIP Interoperability Test Events several times a year. These events are known as SIPITs, which stand for SIP Interoperability Tests, and are weeklong events that allow SIP product implementers to test their products alongside those of their competition. And while many network managers would not fall into the category of product developers, who would be actual participants in the testing events, the information regarding past tests noted at http://www.cs.columbia.edu/sip/sipit/, plus the SIP Test Scenarios developed for the program, and available at http://www.cs.columbia.edu/sip/sipit/scenarios_files/frame.htm, may prove inspirational, as they indicate the level of detail that is required to assure multi-vendor system operation.

Finally, a server benchmarking model, called SIPstone, is available at http://www.sipstone.org/. This paper, also developed at Columbia University, details some of the metrics that should be used to evaluate SIP Proxy, Redirect and Registrar servers, to assure that they have the horsepower required for the expected workload. As noted above, this paper describes tests that may not be personally performed by the network manager, but nevertheless details the underlying performance issues that the network manager should be aware of.

As has been said many times "forewarned is fore-armed", and the wise network manager spends some time considering a SIP network interoperability test plan before signing contracts with all the eager vendors.

Copyright Acknowledgement: © 2005 DigiNet ® Corporation, All Rights Reserved


Author's Biography
Mark A. Miller, P.E. is President of DigiNet ® Corporation, a Denver-based consulting engineering firm. He is the author of many books on networking technologies, including Voice over IP Technologies, and Internet Technologies Handbook, both published by John Wiley & Sons.

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