Case in Point To continue our efforts to help you with your network needs, we will be making several real-world network troubleshooting case studies available to you. The attached case study,, 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 syllable 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 network problems. Taken from Enterprise companies, Service Providers and Network Equipment Manufacturers, we will show you how companies, such as yours, have solved some of their network problems. You will have an opportunity to vote for the case studies of most interest to you. From these votes we will select the case studies for each posting. For additional case studies and to vote for the case studies you would like to see featured next, be sure to regularly visit our web site at: http://www.agilent.com/comms/casestudies Feel free to pass this link on to your colleagues or friends. 1
Problem: During the test and turn-up of the service, many voice quality problems, such as clipping, garbled speech, and crackling, were observed A large multinational oil company 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 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. 2
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, fragmentation, and interleaving and queuing methods for voice that could cause bandwidth and jitter problems. Some international links also had some significant delays. 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 voice We 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 identified packet losses of 2-5%. Since the implementation didn t take full advantage of the voice gateway s ability to manage bandwidth consumption and the router s ability to fragment, interleave and prioritize, some voice packets were getting dropped and those not dropped were experiencing significant jitter. 3
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 communications. 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. 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. 4
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 PRI... Primary Rate Interface PSQM... Perceptual Speech Quality Measurement PSTN... Public Switched Telephone Network QoS... Quality of Service RED... Random Early Detection RFP... Request For Proposal RSVP... ReSource reservation Protocol RTP... Real-Time Protocol RTT... Round Trip Time SDP... Session Description Protocol 5
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 6