Auditing a network to add Voice


My customer would like to add VoIP over their network and they asked us for
an audit. the result of the audit would be simply "you guys are ready for

Breaking it down [high level] for me sounds like : (suggestions are more
than welcomed) :

1) Looking at hardware computation finite resources (cpu, memory...etc)
2) Looking at available bandwidth
3) QoS policy
4) High Availability and Fast Convergence

Any thing else?

They asked us to measure the KPIs (jitter, delay...etc) of their existing
traffic, is there a way to do that?


Sorry i forgot to add more detail.

We are not looking for IP Telephony type of voice but RTP from Media


Iperf can be used to measure jitter and delay as well as simulate a quasi VoIP call. You can also use mtr under Linux which provides jitter and delay measurements from one point to another point. A g.729 call (lower quality) takes about ~40kbps and a g.711 (high quality) used about ~100Kbps of bandwidth. With most of today's networks, the problem isn't bandwidth related, but more with jitter, delay, and packet loss through the network...personally I'm a big fan of deploying QoS through out an infrastructure...well at least in our WAN infrastructure.


Most VoIP solutions are RTP whether internal or via SIP solution from a service provider.

Hi Bret,

These guys are not looking for measuring traffic generated by a tool, they
want to measure what they have running now (not only Voice). I am not sue if
measuring what they have or generating traffic and measuring it is the same
thing. what do u think?


I'm not sure if Wireshark will let you do least with TCP, we do use Wireshark to analyze RTP traffic which provides jitter/loss data, maybe a vendor provided LAN analyzer would provide this information

I still think you're better of on using some type of tools and do the measurement in their network's live at various times of the day. Every path through the network is going to have different delays/jitter/loss at various times of the the day. You can probably get loss via RMON statistics in switches/routers, but delays/jitter requires that you are monitoring a data conversation at the TCP/IP layer and I'm not aware of network equipment (switches/routers) that watch individual TCP/IP layers to provide jitter/delay...that would require quite a bit of a devices resources.

If you run the apps on their network live, they you are basically going to get the information you need about the overall quality of their network they have in place today.

You forgot the most important thing, which ends up driving all the rest:

0) How much VoIP are they planning to do? VoIP for 25 people and VoIP
for 25,000 people are two totally different beasts.

One of the best active measurement products is the BRIX monitoring
system, now owned by EXFO. Active measurement systems have the
capability of sending out emulated application probes (for instance
G.711 calls), or alternatively simple ping tests to gather round trip
times (RTT), jitter, and packet loss. The tests are run, and the data is
gathered at random intervals over an extended time period, thus
providing a statistically accurate picture of network performance at
different times, and under various traffic blends and loads.

Using queuing theory, it can be shown that only 3 variables are required
to accurately predict network performance: RTT, jitter, and packet loss.
Designing a network which will produce the right combination of these 3
variables, mitigates the need for QoS, except as a failsafe to be used
in emergency cases such as DoS attacks. QoS-free networks (FIFO queuing
only) have been designed and implemented which easily support MPEG4
video, HD videoconferencing, and VoIP.