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SIP-based IP telephony monitoring

Are you using IP telephony in your organization? Do you sometimes experience problems with calls that do not connect, or do not fulfill your speech quality expectations? Perhaps you want to work more proactively and make sure that your business critical IP telephony service works well?

Keep on reading, you have come to the right place.

Combined monitoring of call quality and SIP server performance

Netrounds can test and monitor not only how your network influences call quality, but also how well your SIP-based IP telephony server behaves.

Each Netrounds probe consists of a SIP user agent that can be configured in the probe view. You select one of your probes to act as a hub (central node) and one or more probes that act clients in the setup.

Once the monitoring is running, each probe will periodically run through the following loop:

  1. The user agent registers towards your SIP server and measures the response time
  2. The user agent initiates a SIP call towards the hub probe and measure call setup time
  3. The probe measures VoIP quality during a time-limited call using MOS (Mean Opinion Score) using parameters measured on a per-packet basis (packet jitter and packet loss).
  4. The call is terminated and the call hang up time is measured.

The result is shown in long-term historical graphs which give you a good overview of how well your SIP-based IP telephony works over time.

As usual in Netrounds, you can configure alarm thresholds and get weekly SLA reports that visualize the health of your IP telephony service.

Requirements

In order to perform SIP-based monitoring you need to activate test package Non-stop Network Monitoring in your Netrounds subscription.

Features


What are the benefits?

  • Make sure your SIP server behaves well and responds within defined time limits
  • Simultanous monitoring of call quality (MOS) and SIP signalling performance for each call
  • Provides long-term historical data and automated SLA compliance reports

How does it work?

  • You deploy Netrounds' active network probes at strategic locations in your network
  • Start monitoring where your probes make real VoIP calls using SIP signalling
  • Probes measure SIP signalling performance and call quality
  • The cloud-based Netrounds server stores measurements from each of the participating probes
  • Inspect the measurements in real-time using a web browser
  • Set up SLA compliance reporting using e-mail
  • Set up alarm triggering using e-mail, SNMP or towards Nagios/op5
Feature
Description
VoIP signalling
SIP, RFC 3261. Configurable signalling transport using TCP, UDP or TLS.
VoIP media RTP over UDP transport
Supported codecs G.711 a-law and u-law and GSM-EFR
Measurement metrics

Registration time via REGISTER (ms)

Unregistration time via REGISTER using expiry timer = 0 (ms)

Call setup times via INVITE (ms)

Call hangup times via BYE  (ms)

Voice quality (Mean Opinion Score, MOS) based on jitter and packet loss

 

Download your free trial of Netrounds, or request a personal presentation and demo for more information on how Netrounds can help you.