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Special Focus
April 2003

Is Your LAN Ready For SIP?


As companies migrate to VoIP for their internal networks, they must determine whether the LAN will be able to handle voice communications on top of the data that is already present. IT managers need to know what equipment needs to be added to the network, if current switches, hubs, and routers will need to be changed and what effect data will have on the voice and SIP signaling. In addition, software programs like firewalls, Network Address Translation (NAT) software, security systems, and management systems running on the network may cause additional problems for VoIP.

So how does one address these issues? A bulk call generator/test solution can be used to generate numerous, simultaneous calls, to verify the ability of the network to handle the capacity of VoIP traffic, in addition to performing measurements to help determine the quality of voice transmissions and capabilities of product features. When a test set is being considered, the following should be looked at:

� Protocol analysis capability -- view SIP messages for call setup and ensure they are being transmitted and received properly.

� Packet measurement capability -- measure packet loss, jitter, delays, throughput, and out of order packets. These measurements help debug and identify components that cause problems. These measurements should be recorded over a period of time to determine stability and �what if� scenarios.

� Call generation capability that will exceed the highest usage expected on the network.

� Voice encoding with the various codec types that will be used on the network. Depending on the type of voice encoding used, the network may require very different amounts of bandwidth.

� Ability to measure voice quality of real voice transmissions, in addition to packet simulations.


As of late, test equipment manufacturers are responding to user demand for test sets that are inexpensive and flexible enough for the enterprise and small service provider. In addition to just providing tools that monitor protocols and packets, there are now test systems that perform load testing, feature testing, voice quality tests, and packet measurements all in one product. In addition, enterprises can test their analog, ISDN, and TDM interfaces with the same product they use for VoIP testing. Traditionally, network equipment manufacturers have been the main users of these test sets since they cost over $100,000. Today, a low capacity system can cost less than $30,000 and can simulate over 500 simultaneous SIP calls.


When a bulk call generator is connected to a network, it acts as many IP telephones generating calls that simulate users on the network. The users can be calling another telephone, accessing messages from a voice mail system or setting up a conference call. To fully test the system, one must first verify all components of the system will operate as they are designed.

� Links: The physical links and interfaces must be tested for bandwidth and proper transmission.

� Signaling servers: The SIP signaling between the telephones, proxy server, DNS server, and other signaling servers must be interoperable.

� Messaging servers: The voice mail, announcement server, and other messaging systems must have proper call setup, media transmission, and good voice quality.

� Telephones and conferencing servers: Communication between telephones must have little delays and good voice quality. This should include calls through conferencing servers.

� Databases: Queries to databases must be quick and accurate for routing, authentication, and customer information. A call generator can initiate the signaling that requires database queries and measure the time to complete the procedure.

� PSTN gateways: Media and signaling must be properly transferred to the PSTN for call routing and good voice quality.

� Switches and Routers: Network equipment must also provide the correct routing and prioritization to reduce delays and dropped calls. They must have enough memory to buffer the large data packets while a voice or SIP signaling packet is being transmitted.


It is critical to run multiple types of tests on the network, because each test will provide a different view of the network.

� System performance: The first test should be to determine the capability and quality of a call to be set up from any one point to any other point. When the call is made, correct call setup can be verified, and delays and voice quality measured. Calls can be made to or through the various components of the system as described above to verify their functionality. If any problems are identified the individual components must be isolated and the tests repeated.

� Feature Performance: To verify that the features work as designed, calls can be made to the devices in the network that perform the feature. The number of calls should stress the device and demonstrate various feature scenarios.

� Capacity: Every device and link needs to be tested for capacity. Generate enough calls to stress each element and link in the system. It is wise to start with a few calls and increase the amount until problems become apparent. Monitor the various components to see memory usage, Digital Signal Processor (DSP), processor speed, and buffer load.

� Voice Quality: Every device that handles the media must be tested for its ability to pass the information without causing degradation to the quality. Some test systems will look at packet errors, packet loss, and delays to determine voice quality. However, the best evaluation is to send an actual voice message and listen to it. The second best evaluation is to send the voice file and measure the resulting quality with the PESQ algorithm. This algorithm compares the sent file with the received file and gives a score based on the differences that might be heard. Other algorithms like PSQM and PAMS are not as suitable for VoIP as PESQ.

� Protocol analysis: A protocol analyzer will determine whether the product meets a standard, sends the correct messages for call setup, and interoperates with other devices on the network. It shows messages that caused errors and the result or reactions from the errored messages. A protocol analyzer is used to look at SIP messages communicated between devices.

� Packet analysis: Many test sets measure packet characteristics such as packet loss, out of order packets, jitter, and delayed packets. These measurements are important for determining how well the network is performing as well as the cause of problems in the network. For example, packet loss could be due to the buffer capacity has been exceeded or routers not giving priority to the voice packets. Packet delays could be due to processors overloaded or links nearing maximum capacity. Overloading of memory and CPU in routers can easily cause jitter in VoIP traffic.

� Prioritization: Voice packets tend to be smaller than data packets and routers are designed to prioritize larger packets over the smaller ones. Consequently, a number of QoS or priority schemes such as RSVP, DiffServ, and MPLS have been created so that voice traffic is given a higher priority. The ability of a device under test to prioritize packets can be measured with a bulk call generator that provides prioritization of its packets.

Measuring the utilization of the network equipment while the tests are running is imperative. This includes looking at memory, DSP performance, CPU utilization, link bandwidth/congestion, queuing and forwarding capabilities, and database lookup speed.


Once testing has been completed, the weakest links will have been identified and network performance determined. Questions that will be answered as a result of testing include:

� Were calls completed to all devices?

� Were the SIP messages sent according to specifications RFC 2543 or 3261?

� Did all devices interoperate?

� Did all features perform as designed?

� When capacity was increased, which component failed first? Can the configuration be changed to increase capacity?

� Did voice quality drop when certain devices were involved, or certain configurations were used?

� Were all protocols, codec types, and routing able to be handled properly by all devices in the network?

Tom Wingfield is product manager at Spirent Communications. Spirent Communications is a worldwide provider of integrated performance analysis and service assurance systems for next-generation network technologies. The company�s solutions are designed to accelerate the profitable development and deployment of network equipment and services by emulating real-world conditions in the lab and assuring end-to-end performance of large-scale networks. For more information, visit www.spirentcom.com.

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