Is Your LAN Ready For SIP?
BY TOM WINGFIELD
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
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:
ï¿½ Protocol analysis capability -- view SIP messages for call setup and
ensure they are being transmitted and received properly.
ï¿½ 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 ï¿½what ifï¿½
ï¿½ Call generation capability that will exceed the highest usage expected
on the network.
ï¿½ 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.
ï¿½ Ability to measure voice quality of real voice transmissions, in
addition to packet simulations.
NEW TECHNOLOGIES IN TESTING
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
ï¿½ Links: The physical links and interfaces must be tested for bandwidth
and proper transmission.
ï¿½ Signaling servers: The SIP signaling between the telephones, proxy
server, DNS server, and other signaling servers must be interoperable.
ï¿½ Messaging servers: The voice mail, announcement server, and other
messaging systems must have proper call setup, media transmission, and
good voice quality.
ï¿½ Telephones and conferencing servers: Communication between telephones
must have little delays and good voice quality. This should include calls
through conferencing servers.
ï¿½ 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.
ï¿½ PSTN gateways: Media and signaling must be properly transferred to
the PSTN for call routing and good voice quality.
ï¿½ 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.
TYPES OF TESTS
It is critical to run multiple types of tests on the network, because
each test will provide a different view of the network.
ï¿½ 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.
ï¿½ 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
ï¿½ 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.
ï¿½ 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.
ï¿½ 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
ï¿½ 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
ï¿½ 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
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:
ï¿½ Were calls completed to all devices?
ï¿½ Were the SIP messages sent according to specifications RFC 2543 or 3261?
ï¿½ Did all devices interoperate?
ï¿½ Did all features perform as designed?
ï¿½ When capacity was increased, which component failed first? Can the
configuration be changed to increase capacity?
ï¿½ Did voice quality drop when certain devices were involved, or certain
configurations were used?
ï¿½ 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 companyï¿½s 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
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