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VoIP Troubleshooting
Case in Point
Voice Quality Parameter Tuning
To continue our efforts to help you with your network needs, we will be making several real-world net- work troubleshooting case studies available to you.
The attached case study, Voice Quality Parameter Tuning, discusses a a multi-national oil company that had developed a design to use VoIP trunks across the enterprise for its private telephone network. All PBXes are similar models from vendor A. All VoIP gateways are of similar models but from a different vendor. Problem: During the test and turn-up of the service, many voice quality problems, such as sylla- ble clipping, garbled speech and crackling, were observed.
New case studies will be released every 2 to 3 weeks and will cover all topologies and a variety of net- work problems. Taken from Enterprise companies, Service Providers and Network Equipment Manufac-
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VoIP Troubleshooting Voice Quality Parameter Tuning
A large multinational oil company developed a design to use VoIP trunks across the enterprise for its pri- vate telephone network. All PBXes are similar models from Vendor A. All VoIP Gateways are of similar models but from Vendor B.
During the test and turn -up of the service, many voice quality problems, such as syllable clipping, garbled speech, and crackling, were observed.
Case Study:
Voice Quality Parameter Tuning
Problem:
During the test and turn-up of the service, many voice quality problems, such as clipping, garbled speech, and crackling,
were observed
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VoIP Troubleshooting
Case Study:
Voice Quality Parameter Tuning
Action
• Review existing IP and VoIP network design Findings
• IP network design is sound and has been in service for five years with no major problems
• Insufficient use of compression, interleaving and prioritization methods for voice could cause bandwidth and jitter problems
• Some international links have significant delays
We reviewed the existing IP and VoIP network design. The IP Network design was sound and had been in service for many years with no major problems. We did find insufficient use of compression, fragmen- tation, and interleaving and queuing methods for voice that could cause bandwidth and jitter pro blems.
Some international links also had some significant delays.
Case Study:
Voice Quality Parameter Tuning
Action
• Use analyzer to identify traffic types and quantity Findings
• Discovered that main data traffic on links were large file transfers
• VoIP has introduced a large amount of new traffic to network
•
Identified packet losses of 2-5% which is too high for voiceWe used an analyzer to identify the traffic types and traffic quantity for the links in question.
We discovered that the main data traffic on links were large file transfers. The VoIP implementation had introduced a large amount of new traffic to network. By checking the router interface statistics, we identi- fied packet losses of 2-5%.
Since the implementation didn’t take full advantage of the voice gateway’s ability to manage bandwidth
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VoIP Troubleshooting
Case Study:
Voice Quality Parameter Tuning
Root Causes
• Compression methods such as Compressed RTP and VAD were not used thereby increasing bandwidth requirements for voice
• Insufficient interleaving over WAN links (FRF.12, MLPPP) created significant delays and jitter
• Insufficient initial design
The root cause of this problem was in the design. The designers neither understood the capabilities of the equipment nor the impact that the existing traffic would have on the voice traffic. It is imperative that designers have a data traffic baseline with which to work. Also, they must understand that voice data has different performance characteristics than traditional data such as FTP, SMTP, or client/server communi- cations. Once these are understood, then the designer has to understand the capabilities of their equipment so as to recognize whether the overall solution will be viable. Failure on any of these steps is likely to result in failure or degraded performance during implementation and operation.
Case Study:
Voice Quality Parameter Tuning
Solutions
• Enabled VAD, comfort noise, and RTP Header Compression
• Increased de-jitter buffer settings to minimize packet losses and smooth playback
• Used either FRF.12 or MLPPP as appropriate to provide voice/data interleaving in order to minimize delays
• Modified queuing on the router to provide priority for voice over other non-time sensitive data traffic
We turned up VAD on the gateways and noticed some improvement in voice quality. We then enabled RTP header compression, enabled appropriate fragmentation/interleaving mechanism (FRF.12, MLPPP), and changed the queuing method to give priority to voice data. After these changes, the voice circuits’
reliability and quality were acceptable.
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VoIP Troubleshooting
Glossary
ASN.1 ... Abstract Syntax Notation.1 ATM ... Asynchronous Transfer Mode Bc ... Burst Committed
Be ... Burst Excess
BECN ... Backward Explicit Congestion Notification CAC... Call Admission Control
CB-WFQ... Class Based Weighted Fair Queuing Codec... Coder Decoder
CIR ... Committed Information Rate CQ ... Custom Queuing
DHCP... Dynamic Host Configuration Protocol DTI... Domestic Trunk Interface
E&M ... Ear and Mouth
FECN... Forward Explicit Congestion Notification FIFO... First In First Out
FRF... Frame Relay Forum FTP... File Transfer Protocol FXO... Foreign Exchange Office FXS ... Foreign Exchange Station GIS... Geographical Information System ICMP... Internet Control Message Protocol IETF ... Internet Engineering Task Force IPv4... Internet Protocol version 4 ISDN ... Integrated Services Digital network ITU... International Telecommunication Union LLQ ... Low Latency Queuing
Megaco ... Media Gateway Control
MGCP ... Media Gateway Control Protocol MLPPP... Multi Link Point to Point Protocol MOS... Mean Opinion Score
PAMS... Perceptual Analysis Measurement System PBX... Private Branch Exchange
PESQ... Perceptual Evaluation of Speech Quality PQ... Priority Queuing
PQ-WFQ... Priority Queue Weighted Fair Queuing
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VoIP Troubleshooting
SIP... Session Initiation Protocol SLA ... Service Level Agreement SMTP... Simple Mail Transfer Protocol SS7 ... Signaling System number 7 TAC ... Technical Assistance Center TDM ... Time Division Multiplexing VAD... Voice Activity Detection VoIP ... Voice over Internet Protocol VBR... Variable Bit Rate
VQT... Voice Quality Test WFQ ... Weighted Fair Queuing
WRED... Weighted Random Early Detection
Bibliography Books:
Integrating Voice and Data Networks, Scott Keagy, Cisco Press, ISBN 1-57870-196-1
IP Telephony, The Integration of Robust VoIP Services, Bill Douskalis, Prentice Hall, ISBN 0-13-014118-6
IP Telephony, Packet-based Multimedia Communications Systems, Olivier Hersent, David Gurle, Jean-Pierre Petit,
Addison Wesley, ISBN 0-201-61910-5
Signaling System #7, Travis Russell, Third Edition, McGraw Hill, ISBN 0-07-136119-7 Voice Over IP, Uyless Black, Prentice Hall, ISBN 0-13-022463-4
Useful URLs:
Collection of VoIP References on the Web,
http://www.whipper.uwc.ac.za/~mjeffrie/intoreference.htm Collection of H.323 Links,
http://www.packetizer.com/h323link.html
SIP: Session Initiation Protocol, RFC 2543 (Proposed Standard), http://www.ietf.org/rfc/rfc2543.txt Media Gateway Control Protocol, RFC 2705 (Informational), http://www.ietf.org/rfc/rfc2705.txt Megaco Protocol Version 1.0, RFC 3015 (Proposed Standard), http://www.ietf.org/rfc/rfc3015.txt