A CPE Based VoIP Monitoring Solution
Voice over IP (VoIP) has become a widely accepted technology, but like most IP applications it requires attention to
detail to be successfully deployed. Tools like PacketProbe™ can be used to ensure successful deployments, and can
also be used to help with network design, ensuring that the most efficient topologies are implemented.
Additionally, PacketProbe™ can be used for ongoing network management to allow for maximum customer uptime and to allow
the Service Provider to meet or exceed their Service Level Guarantee.
GL's PacketProbe™ is an advanced CPE(Customer Premises Equipment) based VoIP monitoring reporting and
diagnostic appliance, stemming from GL's suite of market leading voice analysis tools.
PacketProbe™ passively monitors VoIP traffic carried over WAN/LAN by producing real-time per call and
per-stream voice quality metrics. Call Detailed Records (CDRs) along with voice quality statistics including Mean Opinion
Score (MOS) and other vital diagnostic information provide network managers immediate visibility into service quality, call
volumes, and call details. Service providers can rapidly drill down and diagnose voice quality related issues.
PacketProbe™ software can be used as an Real-time Monitoring, Reporting and Diagnostic tool. It can fit
seamlessly into any existing standards based management or reporting environment, such as SNMP or RADIUS.
PacketProbes™ can provide vital voice call quality statistics, Call Detail Records and Quality of Service metrics at the
end of each call.. Optionally, GL offers its own Monitoring and Reporting System,
GL’s Linux based routers, along with PacketProbe™, can monitor VOIP traffic through a number of configurations. Performance will depend on the type of router used, based on their CPU and memory characteristics. The basic units, the GLR1000, and GLR2000, which are low-cost routers intended for monitoring lesser numbers of calls, followed by the GLR3000 designed for DS3/E3, BGP, and demanding Ethernet applications.
- Reports Call Detailed Records (CDRs) and Quality metrics per call/stream, including:
- Call summary statistics including caller number, callee number, start time, duration, etc.
- Listening and Conversational Mean Opinion Score (MOS-LQ, MOS-CQ)
- Listening and Conversational R factors (R-LQ, R-CQ)
- R-factors for burst and gap conditions – R-Burst, R-Gap
- Max/min/average gap, delay, jitter
- Missing, discarded, out of sequence and duplicate packet statistics
- Max/min/average round trip delay
- Codec and SSRC identification
- Supports wide range of industry standard and proprietary codecs (see list in specs)
- Automatically detects active VoIP calls based on VoIP call signaling or RTP session. Supports SIP, MGCP, MEGACO,
- Call Quality analysis using optimized ITU-T G.107 with ETSI TS 101 329-5 Annex E
- Supports Japanese TTC JJ201.01 VoIP monitoring requirements
- Flexible hardware and software platform
- Available WAN interfaces
- 8xT1E1, T3E3, OC3/STM1, Ethernet
- Available LAN interfaces
- 10/100 or 10/100/1000 Ethernet
- Available DTE interfaces
- Access Types
- Management and Reporting
- Supports Linux or Windows based applications
- Industry standard reporting mechanism like SNMP, Radius, TCP/IP
- Multiple PacketProbes™ can provide network wide voice quality visibility through GL's
NetSurveyorWeb™ or other NMS systems.
Real-time Monitoring and Reporting
PacketProbe™ is able to report said metrics to a local console or to a centralized database via industry standard
protocols like TCP/IP, SNMP and RADIUS. A Network Management System (NMS) like GL's NetSurveyorWeb™ would
then allow many simultaneous users to read and query the database using a conventional web browser. Multiple
PacketProbes™ coupled with GL's NetSurveyorWeb™ can provide complete and immediate network-wide voice
quality visibility. To learn more about NetSurveyorWeb™, please visit GL's
PacketProbe™ reports Call Detailed Records along with vital voice call quality statistics. This CDR along with QOS
is available at the end of each call. At the central location, with appropriate database analysis tools, one can examine
collected data in real-time and identify a number of specific problems that affect voice quality. A typical VoIP service has
thousands of active sessions, any of which may experience transient problems.
PacketProbe™ can be deployed as a standalone probe within networks or integrated within cpe like Routers, Bridges, WAN Terminators, or Firewalls at strategic locations for passive traffic monitoring using port spanning or port mirroring of Ethernet switches or using external network-taps. Both configurations are shown below.
As a stand-alone applicance PacketProbe is completely transparent to your existing network and is able to interface
with WAN or LAN. It can even function as the demark point between WAN and LAN.
In any scenario it passively monitors VoIP traffic and produces real-time per call and per-stream voice quality metrics
necessary for network administrators to isolate and diagnose voice quality problems.
Each packet is copied from the IP Packet Stream, and then time stamped and filtered. RTP packets are sorted by call
and analyzed for jitter, packet loss, and sequence. The Jitter Buffer Emulator (JBE) inspects every RTP packet header,
estimating delay variation and emulating the behavior of a fixed or adaptive jitter buffer to determine which packets are
lost or discarded. From these measurements and other configuration data such as codec – an accurate end to end MOS
score is calculated. PacketProbe™ also collects quality reports (i.e Jitter, RTD ..) sent by VoIP endpoints and correlates
with data obtained by internal calculations.
- Call quality analysis using Optimized ITU-T G.107 with ETSI TS 101 329-5 Annex E
- Supports Japanese TTC JJ201.01 VoIP monitoring requirements
Call Quality Metrics
- Listening and conversational quality MOS Scores with ACR, ITU and TTC scaling – MOS-LQ, MOS-CQ
- Listening and conversational quality R-factors – R-LQ, R-CQ
- Separate R-factors for burst and gap conditions – R-Burst, R-Gap
- Automatically detects active VoIP calls based on call signaling or RTP session
- SIP, MGCP, Megaco, H323/H225
- Packet loss rate, packet discard rate, burst length/density
- G.711, G723.1, G.726, G.728, G.722.x, G.729A, G.729B, GSM, FR etc.
- Percentage degradation due to loss, jitter, codec, delay, recency
Interface Protocol Compatibility
- Network monitoring interface – IPV4/V6, UDP, RTP (RFC3550), RTCP (RFC3550)
- Approved versions/ releases/ builds of Linux
- MS-Windows 7 / 8.1
* Specifications are subject to change without notice.
Back to VoIP Analysis and
Simulation Index Page