Product Brief Work Groups and Applications The T-BERD/MTS-5800, -6000A, and -8000 VoIP Analysis was designed for central office, metro, and government technicians who install, turn up, and troubleshoot voice over IP (VoIP) service. It is also made for engineers who maintain, troubleshoot, and are responsible for the evolution of Ethernet/IP networks that carry VoIP traffic. Solution Description VoIP test options for the T-BERD/MTS-5800, -6000A, and -8000 emulate an IP phone by placing and receiving VoIP calls at all Ethernet rates up to 10 G to ensure successful installation of VoIP services. Support for SIP, Cisco SCCP, and H.323 protocols, enables users to verify connectivity to the signaling gateway. Technicians can quickly and objectively prove acceptable call quality by analyzing audio delay, jitter, packet loss, MOS, and R-Factor in real time with good, fair, or poor quality ratings using configurable QoS thresholds. Extensive troubleshooting capabilities include visibility into the entire call setup signaling process and support for line rate capture and decode of both audio and signaling packets. Users can also verify the packet network’s ability to reliably transport voice traffic using the Triple-Play Application to generate and analyze the QoS for an audio/voice stream with simultaneous background traffic. This capability can prove acceptable voice/audio quality under load conditions and exposes potential high-bandwidth utilization problems, such as misconfigured voice traffic prioritizations. Applications y Install, verify, and troubleshoot SIP, Cisco SCCP, and H.323 Fast Connect signaling protocol VoIP services y Verify Ethernet/IP packet network suitability to reliably transport VoIP traffic with audio/voice analysis and simultaneous background traffic T-BERD®/MTS-5800, -6000A, and -8000 VoIP Analysis Verifying VoIP Services in Ethernet/IP Networks
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Product Brief
Work Groups and ApplicationsThe T-BERD/MTS-5800, -6000A, and -8000 VoIP Analysis was designed for central office, metro, and government technicians who install, turn up, and troubleshoot voice over IP (VoIP) service. It is also made for engineers who maintain, troubleshoot, and are responsible for the evolution of Ethernet/IP networks that carry VoIP traffic.
Solution DescriptionVoIP test options for the T-BERD/MTS-5800, -6000A, and -8000 emulate an IP phone by placing and receiving VoIP calls at all Ethernet rates up to 10 G to ensure successful installation of VoIP services. Support for SIP, Cisco SCCP, and H.323 protocols, enables users to verify connectivity to the signaling gateway. Technicians can quickly and objectively prove acceptable call quality by analyzing audio delay, jitter, packet loss, MOS, and R-Factor in real time with good, fair, or poor quality ratings using configurable QoS thresholds. Extensive troubleshooting capabilities include visibility into the entire call setup signaling process and support for line rate capture and decode of both audio and signaling packets.
Users can also verify the packet network’s ability to reliably transport voice traffic using the Triple-Play Application to generate and analyze the QoS for an audio/voice stream with simultaneous background traffic. This capability can prove acceptable voice/audio quality under load conditions and exposes potential high-bandwidth utilization problems, such as misconfigured voice traffic prioritizations.
Applications
y Install, verify, and troubleshoot SIP, Cisco SCCP, and H.323 Fast Connect signaling protocol VoIP services
y Verify Ethernet/IP packet network suitability to reliably transport VoIP traffic with audio/voice analysis and simultaneous background traffic
T-BERD®/MTS-5800, -6000A, and -8000 VoIP Analysis Verifying VoIP Services in Ethernet/IP Networks
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Solution BenefitsAs more time-division multiplexing (TDM) circuits are converted to VoIP trunks, metro technicians require a tool that can objectively verify VoIP-call quality and more quickly troubleshoot those services. This software test option looks beyond the transport layer and emulates the true end-user VoIP service or experience so technicians can cost-effectively ensure end-user satisfaction. Easy-to-decipher objective pass (green)/fail (red) QoS analysis organized by network layer, technicians can quickly identify and sectionalize problems, saving crucial turn-up and troubleshooting time. Integrated troubleshooting tools, such as capture and decode and real-time signaling logs, free up highly experienced technicians and eliminate the need for separate VoIP analyzer equipment. The unique ability to analyze voice/audio QoS with background traffic in one simple step using the Triple-Play Application lets users stress the circuit and guarantee proper functioning of Class of Service prioritizations.
Use Case: VoIP Call PlacementThe T-BERD/MTS-5800, -6000A, and -8000 can be used to emulate an IP phone to place and receive VoIP calls through the network by calling another test set at the far end, or any phone. In this scenario, the test set, configured in terminate mode, is used to place a VoIP call to a remote IP phone by first connecting or registering with the VoIP call manager or proxy server. Real voice or a pre-loaded audio clip can be transmitted through the network and can be listened to by connecting a Viavi Solutions-provided USB headset to the test set. Upon call setup, the test set analyzes audio/voice and transport QoS statistics in real time, such as jitter, packet loss, and MOS scoring, and indicating a pass/fail analysis.
Feature/Benefit Summary
Benefit Feature Description AdvantageVerifies true end user VoIP call experience with real voice or pre-loaded audio clips and ensures satisfaction before enabling VoIP services
IP phone emulation with SIP, SCCP, H.323
Place and receive VoIP calls using a variety of signaling protocols at all line rates from 10 Mbps to 10 G
Verifies connectivity to a signaling gateway or proxy server quickly and easily
Guarantees that all configurations are possible in one tool with flexible analysis and ubiquitous usage
Supports a wide range of CODECs
Complete set of standard ITU-T CODECs including G.711 μ-law/A-law, G.723.1, G.726, G.729a, G.729ab, and G.722
Multiple choices are available, encompassing all network and equipment possibilities
Lets users quickly identify and sectionalize problems during service turn-up and troubleshooting
Layered summary results
Presents QoS measurements as pass (green)/fail (red) organized by network layer (physical, Ethernet/IP, and RTP)
Provides repeatable, simple-to-understand results and detailed statistics
Ensures the service meets true customer QoS repeatedly and consistently
MOS and R-factor scoring
Patented Telchemy real-time assessment of voice quality
Integrated troubleshooting analysis in one easy-to-use tool frees up experienced technicians and eliminates the need for additional analyzer equipment
Troubleshooting with capture/decode and signaling log
Capture and decode line rate audio and signaling traffic as well as a real-time call setup signaling logs
Quickly and efficiently troubleshoot call setup problems with clearly identified errors
Ensures true quality of VoIP services and confirms proper voice traffic prioritization within the transport network
Audio/voice analysis with background traffic
Simultaneously generate and analyze line rate audio/voice streams with background traffic
Exposes high-bandwidth network-utilization problems that can negatively impact voice quality
T-BERD/MTS-6000A T-BERD/MTS-5800
Location A Location B
Network Provider
IP Network(LAN)
Demark Demark IP Network(LAN)
VoIP Call Manager
Central Office
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Simplified Triple-Play Setup and ResultsAn optimized triple-play traffic streams configuration removes the complexity of emulating multiple services with flexibility to allow different setups per stream. Users can observe QoS results and errors for each stream in a transport pipe view indicating simple pass/fail, or they can view it in a graphical format over time.
Use Case: Triple-Play Voice Analysis with Background TrafficThe T-BERD/MTS-5800, -6000A, and -8000 can be used to generate and analyze voice traffic at line rates with simultaneous background traffic to verify whether a packet Ethernet/IP network can reliably transport voice along with data without requiring a VoIP call manager or signaling. In this scenario, the test set is configured in terminate mode using the Layer 3 or Layer 4 Triple-Play Application which can generate multiple traffic flows including real voice, simulated video, and data. Different priorities and settings can be configured per traffic stream and key SLA/KPI parameters also can be measured per stream, such as delay, jitter, and loss. The pass/fail QoS measurements for voice/audio are analyzed while the transport pipe is filled with background traffic exposing any negative impact high-bandwidth utilization may have on voice quality
Simplified VoIP ResultsThe simple, easy-to-read call summary results are organized by network layer so that users can quickly identify and locate network problems. All transport call QoS measurements are provided in one table. Pass/fail indications are instantly revealed and can be saved to a report for verification proof.
To reach the Viavi office nearest you, visit viavisolutions.com/contacts.
viavisolutions.com
Frequently Asked Questions
Q: Which signaling protocols and CODECs does this option support and on which interfaces?
A: The T-BERD/MTS-6000A and -8000 MSAM and T-BERD/MTS-5800 VoIP options support SIP, Cisco SCCP, and H.323 Fast Connect signaling protocols on all Ethernet interfaces from 10 Mbps up to 10 GE. They also support these CODECs: G.711 μ-law/A-law, G.723.1, G.726, G.729a, G.729ab, and G.722.
Q: What is the advantage of using a tester for VoIP rather than simply listening to a call?
A: The test set’s in-depth statistics can be used to troubleshoot registration issues with SIP, SCCP, or H.323 fast connect protocols that cannot be accomplished with a phone. It also provides a qualitative MOS score that eliminates listener subjectivity and impartiality in determining call quality.
Q: Can I capture the voice call and playback later?
A: The test set supports capture and decode of both the voice call and the signaling packets. Filters can be used to capture a specific call or to capture just voice or just signaling packets. However, the test set does not support playback at this time.
Q: Is this a software or hardware upgrade to existing units in the field?
A: The VoIP option requires a software upgrade for all T-BERD/ MTS-5800, -6000A, and -8000 hardware configurations.
Q: Are there pre-requisites for this feature?
A: No, there are no pre-requisites. A Viavi-supplied headset is required to generate and hear voice/audio. The part number for the T-BERD/MTS-6000A and -8000 headset is CUSB-HEADSET. The part number for the T-BERD/MTS-5800 is HS-10-017673.
Ordering InformationDescription Part NumberT-BERD/MTS-6000A and -8000 MSAMIncludes SIP and H.323 CTVOIP(-U1)Adds SCCP to above CTSCCP(-U1)Headset for audio CUSB-HEADSETT-BERD/MTS-5800Includes SIP and H.323 C5VOIP(-U1)Adds SCCP to above C5SCCP(-U1)2.54 headset for audio HS-10-017673
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