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VoIP Analysis and Simulation
Overview
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, PacketScanWEB™.
Main Features:
- 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 Quality 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
- For narrowband and wideband voice - R and MOS Scores
- Degradation factors - % degradation due to loss, discard, delay, gap, jitter, and codecs
- Lost, Discarded, Out-of-sequence, Duplicate packet statistics
- Round Trip Delay (RTD) statistics
- Codec and SSRC identification
- Packet – Packet Delay Variation (PPDV - RFC3550), Mean Absolute Packet Delay Variation (MAPDV - G.1020) and Y.1540 Jitter Metrics
- Supports wide range of industry standard and proprietary codecs (see list in specs). CODEC’s supported include G.711, G.723, G.726, G.728, G.722.x, G.729A, G.729B, GSM, FR etc
- Automatically detects active VoIP calls based on VoIP call signaling or RTP session. Supports SIP, MGCP, MEGACO, H.323/H.255
- 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
- Distributed network architecture
Hardware Features:
- Flexible hardware and software platform
- WAN interfaces
- NxT1E1 (N=8), T3E3, OC3/STM1, Ethernet
- LAN interfaces
- DTE interfaces
- Access Types
- Management and Reporting
Applications
- PacketProbe™ can be integrated into third-party products like a Router, Network Interface Device, and In-line
or Passive Probes
- Linux or Windows based embedded application support
- Industry standard reporting mechanism like SNMP, Radius, TCP/IP
- Multiple PacketProbes™ can provide network wide voice quality visibility through GL’s
PacketScanWeb™ 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 PacketScanWEB™ would then allow many simultaneous users to read and query the database using a conventional web browser. Multiple PacketProbes™ coupled with GL’s PacketScanWEB™ can provide complete and immediate network-wide voice quality visibility. To learn more about PacketScanWeb™, please visit GL’s web site.
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.
Centralized Scenario
Deployment Scenarios
PacketProbe™ can be integrated and embedded into intelligent third-party network devices like Routers, Bridges, WAN Terminators, or Firewalls. PacketProbe™ can be also deployed as a standalone Probe within networks 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.
Inline PacketProbe™
Standalone PacketProbe™
PacketProbe™ Architecture
PacketProbe™ Architecture
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.
Specifications
Industry Standards
- 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
VoIP Signaling
- Automatically detects active VoIP calls based on call signaling or RTP session
- SIP, MGCP, Megaco, H323/H225
IP/RTP Metrics
- Packet loss rate, packet discard rate, burst length/density
Codecs Supported
- G.711, G723.1, G.726, G.728, G.722.x, G.729A, G.729B, GSM, FR etc.
Degradation Factors
- Percentage degradation due to loss, jitter, codec, delay, recency
Interface Protocol Compatibility
- Network monitoring interface – IPV4/V6, UDP, RTP (RFC3550), RTCP XR (RFC3611)
Supported Platforms
- Approved versions/ releases/ builds of Linux
- MS-Windows XP/Vista/7
Buyer's Guide:
* Specifications are subject to change without notice.
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