Key parameters for testing VoIP networks

Oct. 19, 2007
By Maya Gershon, Agilent Technologies -- Conventional wisdom says gaining a customer back from churn takes an investment four times greater than the original investment. As such, service providers must properly test their VoIP networks.

Conventional wisdom says gaining a customer back from churn takes an investment four times greater than the original investment. As such, service providers must properly test their VoIP networks.

By Maya Gershon, Agilent Technologies

Today's service providers, both local and global, are increasingly moving to voice over IP (VoIP). Whether it is the low price, the higher capabilities, the relative advantage VoIP gives them over competing service providers, the technological advances, or all of the above that prompt them to move from the legacy telephony network to VoIP, the migration has already begun. But are service providers ready for the move? Are VoIP networks set up and tested to support large numbers of users? Or are service providers at risk of losing customers during the move due to low network quality?

The legacy telephony network is a well oiled machine. Users are accustomed to superior quality, accessibility, and simplicity of use. In fact, cost is the key reason why most users switch to VoIP, not dissatisfaction with their current service. But this means that users have a choice, and they will not tolerate inferior quality. If moving to the VoIP network is too complicated, too expensive, or the audio quality is inferior to the network in place, users will simply continue purchasing voice service from their current provider.

In order to successfully deploy a VoIP network and compel users to switch from the legacy telephony network to the VoIP network, service providers and equipment vendors must thoroughly test each system. They must ensure that the quality enjoyed by the end user does not fall short of the quality to which they are accustomed.

That said, testing a VoIP network is no simple matter. Legacy communications networks were not originally intended to support real-time applications like voice and video. Without thorough testing of the VoIP network, then, the chances for a successful deployment are slim. In fact, Gartner analysts estimate that 75% of providers who have tried to deploy a VoIP network without thorough testing ahead of time have failed.

Testing parameters

Several key measurements must be made to ensure quality VoIP service delivery, including protocol testing, quality testing, and compliance with the legacy telephony network, among others.

Protocols. There are several VoIP protocols, the most common of which are SIP and H323. H323 is an ITU-T standard that uses the IP network to transmit video and voice. It was originally developed to support video conferencing but soon became one of the leading standards for VoIP. H323 includes other protocols, such as H225 for call signaling and H245 registration, gateway for transmission interface and control, RTP for continuous transmission of packets as well as multiple codecs for transmission processing.

An IETF standard, SIP is the competing standard for H323. SIP operates at the application level to create voice and video communications and is a less complicated alternative to the ITU-T's H323 protocol, implementing most of the basic features of H323 but without the full range of H323 capabilities.

Quality. A critical element of the service providers' triple-play bundle, VoIP nevertheless adds a layer of complexity to both the network and its testing process. Legacy communication networks were designed to support data applications, and, as such, the network is forgiving of delays, packet loss, and noise. A VoIP network, by contrast, is very sensitive to such impairments. The human ear does not forgive delays, for example, and can detect delays down to 70 msec. Furthermore, the human ear is also sensitive to delay variability or jitter, which can cause a speaker's voice to sound different and metallic.

To improve user experience and customer retention, service providers and equipment vendors must obtain real-time data on the number of packets lost, delays, and delay variability.

Support for a large number of users. When deploying a VoIP network, service providers should consider that the low cost of VoIP calls may generate a very large number of users within a short time. For successful deployment of a VoIP network, therefore, service providers and equipment vendors must test the network under loads of many concurrent VoIP calls and under extreme conditions.

Service providers and equipment vendors should know the system limitations in terms of the maximum number of concurrent active calls, the maximum call setup rate, the maximum RTP transmission rate, the delay created between one call and another, the maximum number of SIP and H323 calls concurrently, and the number of VoIP calls over IPv6. Furthermore, they should test the impact of denial-of-service (DoS) attacks under heavy call load to determine if such attacks degrade network performance or result in dropped calls.

Figure 1. Degradation of VoIP performance due to DoS attacks.

For tests such as these, the test equipment should be able to generate thousands of SIP and H323 calls concurrently with DoS attacks. The test results would then reflect system performance under a load of SIP calls combined with H323 calls and enable service providers to determine whether the system is affected by attacks and/or performance is impaired.

Compliance with the legacy telephony network. Service providers and equipment vendors also should consider the interface between the legacy telephony network and the VoIP network. The best example of this interface occurs when a VoIP subscriber wishes to communicate with a subscriber of the legacy telephony network. Service providers must ensure that the material difference between the two networks does not cause issues of delay, packet loss, or delay variability.

Information security. For the most part, the legacy telephony network maintains the privacy of calls. Obviously, it can be hacked, but that is a complicated endeavor. The VoIP network, by contrast, is far more accessible in terms of both attacks and privacy violations. All the information passing through the IP network is freely accessible unless steps are taken to protect it. Service providers often turn to information security mechanisms like firewalls and virtual private networks (VPNs), but such mechanisms must also be tested to ensure that they are functioning properly and not impairing the system performance.

Figure 2. VoIP over VPN.

In case of firewalls, service providers and equipment vendors should test performance data. What is the maximum number of calls the firewall can handle concurrently, and what is the utmost number of calls the firewall can handle concurrently when a range of network traffic is added?

To test this scenario, test equipment must simulate thousands of VoIP calls passing through the firewall and also replicate peer-to-peer (P2P) traffic to better simulate real life. P2P protocols like Kazaa, Gnutella, eDonkey, or Skype are used for downloading and transferring files from one computer to another. In today's network, about 80% to 90% of traffic comprises this P2P traffic. In order to create a realistic scenario of network traffic, therefore, test equipment must be capable of simulating these protocols as well as other user profiles. Such a complex test requires testing equipment for Layers 4 to 7, which can simulate various traffic profiles in addition to P2P traffic. To create a scenario that is even closer to reality, a simulated DoS attack should be added before testing the network.

To secure the information passing through the network and to maintain privacy of VoIP calls, enterprises, government offices, and private individuals often choose to deploy VPNs. A VPN is a secure pipe in which traffic is encrypted using the IPSec protocol. The encryption is intended to maintain privacy and security, but its use makes the VoIP network more complex. Service providers must ensure that the encryption functions as advertised and does not create a bottleneck in the network by transmitting a large number of SIP and H323 calls through IPSec pipes.

Figure 3. Realistic test scenario.


The VoIP network has attracted many users seeking convenient, high-quality, low-cost services. But are VoIP services really high quality? To successfully deploy a VoIP network, service providers and equipment vendors must ensure that all network components are thoroughly tested using scenarios as close as possible to reality and under a high call load prior to actual deployment.

Maya Gershonis an IP product specialist at Agilent Technologies' Data Network Operation (DNO) in Petah Tikva, Israel. She is responsible for the business development and marketing activities of the DNO next-generation IP communication test equipment portfolio in Israel, which covers communication test, network simulation, and troubleshooting equipment for LAN and WAN networks.