Avaya IP Voice Quality Network Requirements User's Manual
Contents
1. WAN Bandwidth using Frame Relay or PPP L2 Protocol in Kbps Codec Type G 711p amp G 711p A G 726 G 726 32 G 729 amp G 729A G 711A w cRTP Ce w cRTP G 729A_ w cRTP 10 ms 102 4 73 6 70 4 41 6 46 4 17 6 Voice 20 ms 83 2 68 8 51 2 36 8 27 2 12 8 Payload 30 ms 76 8 67 2 44 8 35 2 20 8 11 2 Size 40 ms 73 6 66 4 41 6 134 4 17 6 10 4 50 ms 71 7 65 9 39 7 733 9 15 7 9 9 60 ms 70 4 65 6 38 4 33 6 14 4 9 6 Observations G 711 is inappropriate for most WAN connections because it uses more bandwidth than traditional TDM channels The G 711 column values can be used if voice quality is desired above the bandwidth cost Most point to point links use G 729A or G 729A w cRTP as a good compromise between voice quality rendered and bandwidth required The approximate bandwidth of the G 729 w cRTP headers varies slightly because even though headers are compressed to 12 bytes a full header of 48 bytes is periodically sent The period of full header transmission is usually somewhat configurable by the user The cells in H represent bandwidth required when using the default Avaya voice payload sizes Use these values to begin your WAN bandwidth calculations 6 2 LAN Bandwidth Comparison Ethernet is the reigning layer 2 protocol in most LANs world wide Yet there are several seemingly conflicting bandwidth values stated for VoIP in the LAN There are four2. agreement with the VPN provider to guarantee an acceptable level of service Before implementing VoIP with a VPN users should test their VPN network to make sure it meets the requirements specified in the Document Summary 10 3 Frame Relay Voice transported over frame relay can be subject to more delay and jitter when compared to ATM or point to point TDM circuits This is due to many factors which are not covered in detail here Instead Avaya offers remedies to protect voice traffic from the susceptibilities of frame relay in Appendix B Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 20 10 4 Multi Protocol Label Switching MPLS Voice using MPLS is an effective replacement for Frame Relay services because maintenance and operational costs are generally lower It is very important to choose a premium service offering from the MPLS provider to treat voice and video with real time delay jitter and packet loss needs The SLA Service Level Agreement should define this service and any penalties for lack of service The SLA should define which DSCP values from the customer will receive real time treatment 10 5 Network Address Translation NAT VoIP may not work well with networks that use NAT Network Address Translation because many NAT implementations do not support H 323 protocols The destination IP address is encapsulated in more than one header the Q 931 H 225 and IP headers Some NAT implementations change only the a
3. 3 Using DSCP 9f TOS EE 9 4 4 Using IEEE 802 1 DO 9 4 5 USMO VLAN EE 10 5 Network Parameters aiser een ae og REES RRE cole DEE 10 5 1 Network Packet Delay rr I I I EERE Ents 10 5 2 Network lte TEE 11 5 3 Packet EOSS aroni existent EE EE 12 5 4 Network Packet Mis Order EEE EEE Ents 12 5 5 TRANS COGING EE 13 5 6 ECW DEE 13 5 7 Silence Suppression and Voice Activity Detection cccccece cece esse eeeeeeeeeeeeeneenes 13 5 8 Network DUDIOX e uuusteuiegedute esd ged e Se US d DREES NEEN EENEG E E 14 5 9 Codec Selection EE 14 6 Bearer Bandwidth 2 rr I I I I I NIU EERE EEE EEE 14 6 1 WAN Bandwidth Comparison ccccccc cnn enn 15 6 2 LAN Bandwidth Comparison 8 0N KeN SG H NENNEN ENNER ENNER bead ENK EN dee EEN 15 7 Network Aesesement rr RR I I I I I IU UI I EEE EEE nts 16 8 PC Considerations using Avaya s One X Commupnicator eee ee eee eee teat eae en ees 18 9 Bandwidth Reouirements NNN NENNEN 19 9 1 Bandwidth Requirements using IP SoftPhone or IP Agent 19 10 Other Elements that Affect Volb etree enn nnn 20 10 1 WAN CONSIGEGratiONS uegeeg eg seed eg ative AEN E Dee Ee 20 10 2 VPN Virtual Private Network ccecccccec cece ren nnn nnn Enea 20 10 3 Frame Relay siscisvnactssnsdasriscaddedrksmieienersde vite Ee NNEREN ANEREN ge 20 10 4 Multi Protocol Label Switching ML 21 10 5 Network Address Translation NAT ccccccccceeee eee e ene n eens nese neta ene tnaenaennenas 21 APPe NdIX A vtvcienecoitiess veav
4. canceller The problem is exacerbated in VoIP systems If the one way trip delay between endpoints is larger than the echo canceller memory the echo canceller won t ever find a pattern to cancel Avaya s G650 all H 248 gateways Avaya One X software and all Avaya IP telephones incorporate echo cancellers designed for VoIP to improve voice quality 5 7 Silence Suppression and Voice Activity Detection Voice Activity Detection VAD monitors the received signal for voice activity When no activity is detected for the configured period of time the Avaya software informs the Packet Voice Protocol This prevents the encoder output from being transported across the network when there is silence resulting in additional bandwidth savings The Avaya software also measures the idle noise characteristics of the telephony interface It reports this information to the Packet Voice Protocol to relay this information to the remote end for comfort noise generation when no voice is present Aggressive VADs cause voice clipping and can result in poor voice quality but the use of VAD can greatly conserve bandwidth and is therefore a very important detail to consider when planning network bandwidth especially in the WAN Wide Area Network Avaya s Communication Manager Software all Avaya IP Telephones and Avaya One X products can all employ silence suppression to preserve vital bandwidth although this is not a best practice Issue 3 2 Copyrig
5. data network on the other hand is packet switched Data is less sensitive to delay and jitter but cannot tolerate loss The data philosophy has centered on providing reliable data transmission over unreliable media almost regardless of delay Bandwidth in the data world is largely shared so congestion and delay are often present and can cause problems for multimedia applications such as voice The factors that affect the quality of data transmission are different from those affecting the quality of voice transmission For example data is generally not affected by delay Voice transmissions on the other hand are degraded by relatively small amounts of delay and cannot be retransmitted Additionally a tiny amount of packet data loss does not affect voice quality at the receiver s ear but even a small loss of data can corrupt an entire file or application In some cases introducing VoIP to a high performing data network can yield very poor voice quality Therefore implementing VoIP requires attention to many factors including e Delay e Jitter e Packet loss e Packet mis order e Available bandwidth e Packet prioritization e Network design e Endpoint audio characteristics sound card microphone earpiece etc e Duplex e Transcoding e Echo e Silence suppression e Codec selection e Router and data switch configuration e Reliability e Scalability e Manageability e WAN protocols e QoS CoS policy e Encryption Decryption I
6. does not need to be because the expectation is that not all three branch offices will burst up to the maximum at the same time In an implementation like this the service is probably negotiated through a single vendor But it is likely that Dallas and Houston are serviced by the same LEC and that the frame relay is intra lata even if it was negotiated through an IXC such as AT amp T or WorldCom or Sprint The service between Dallas and the other two branch offices however is most likely inter lata Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 25 Issue and Alternatives The obstacle in running VoIP over frame relay involves the treatment of traffic within the CIR and outside of CIR commonly termed the burst range port rate no guarantees in burst range CIR guaranteed delivery within CIR 0 As the preceding figure illustrates traffic up to the CIR is guaranteed whereas traffic beyond the CIR typically is not This is how frame relay is intended to work CIR is a committed and reliable rate whereas burst is a bonus when network conditions permit it without infringing upon any user s CIR For this reason burst frames are marked Discard Eligible DE and are queued or discarded when network congestion exists Although experience has shown that customers can achieve significant burst throughput it is unreliable and unpredictable and not suitable for real time applications like VoIP Therefore the objective is
7. for objective factors Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 4 Network delay One way Between endpoints o 80ms milliseconds delay or less can but may not yield toll quality o 80ms to 180ms delay can give business communication quality This is far better than cell phone quality and in fact is very well suited for the majority of businesses o Delays exceeding 180ms may still be quite acceptable depending on customer expectations analog trunks used codec type etc See section 4 1 for more information Network jitter Jitter is a measure of the variability of delay Between endpoints o Toll quality suggests average jitter be less than the packet payload This value has some latitude depending on the type of service the jitter buffer has in relationship to other buffers packet size used etc See section 4 2 for more information Network packet loss The maximum loss of packets or frames Between endpoints o 1 or less can yield toll quality depending on many factors o 3 or less should give Business communications quality Again this quality is much better than cell phone quality o More than 3 may be acceptable for voice but may interfere with signaling See section 4 3 for more information Recommendations Avaya highly recommends consideration of the following list of Best Practices when implementing VoIP QoS CoS Quality of Service QoS for voice packets is obtained only
8. information rate is critical In a Frame Relay network any traffic exceeding the CIR is marked discard eligible and will be discarded at the carrier s option if it experiences congestion in its switches It is very important that voice packets not be dropped Therefore CIR should be sized to average traffic usage Usually 25 of peak bandwidth is sufficient Also Service Level Agreements SLAs should be established with the carrier that defines maximum levels of delay and frame loss and remediation should the agreed to levels not be met Network management is another important area to consider when implementing VoIP Because of the stringent requirements imposed by VoIP it is critical to have an end to end view of the network and ways to implement QoS policies globally Products such as HP OpenView Network Node Manager Prognosis Concord NetHealth and MRTG will help administrators maintain acceptable service Should a company not have the resources to implement and maintain network management outsource companies may assist with this need Avaya offers network assessment and redesign services should they be necessary Common issues Some common bad habits that can severely impact network performance especially when using VoIP include e Using a flat non hierarchical network e g cascading small workgroup switches together This technique quickly results in bottlenecks as all traffic must flow across the uplinks at maximu
9. not additive Most of the time there is no signaling on a call so two or more phones can use the same 1 450 Kbps bandwidth space His document does not cover signaling bandwidth Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 19 10 Other Elements that Affect VoIP 10 1 WAN Considerations Until WAN bandwidth becomes affordable at any speed delivering bandwidth to applications over the WAN will remain a formidable task When voice traffic is carried on packet networks different labeling or queuing schemes function to give voice packets priority over data packets The presence of large data packets may result in added serialization delay for VoIP packets across WAN links This is due to the fact that smaller VoIP packets are held in queue while larger data packets are processed onto the WAN link To avoid excessive delay there may be benefit to fragmenting the larger data packets and interleaving them with the smaller voice packets One technique is to adjust the packets by adjusting the Maximum Transmission Unit MTU size Minimum MTU size should be no smaller than 300 bytes and no larger than 550 bytes LAN based MTUs can be as large as 1500 bytes Note reducing the size of the MTU will add overhead and reduce the efficiency of data applications Other techniques such as Multilink PPP MPP Link Fragmenting and Interleaving LFI and Frame Relay Fragmentation FRF12 allow network managers to fragment larger packets and allow q
10. on the seats the gloss of the paint or the extreme exactness of the fit of the doors to the body Trish knows the paint will soon have chips the body will get dents and the interior will stain Quality in the above examples consists of entirely different values Therefore what one person values in quality may be almost irrelevant to another Both Bill and Trish purchased quality vehicles that have superior but different features 3 2 What is Voice Quality Defining voice quality is also difficult because the values of a small business can be greatly different than a business that is larger or located in another culture or country This is why Avaya presents choices using a tiered system of network requirements One number for delay or jitter or packet loss cannot satisfy all customers in all businesses and in all cultures Ultimately each business must decide if quality voice using VoIP requires the first tier of values or other tiered values specified in this paper 4 Prioritizing Voice Traffic In order for a VoIP solution to function well the network must be able to give voice packets priority over ordinary data packets and sufficient bandwidth must always be available Avaya s products for VoIP Communication Manager Software include several standard strategies to prioritize voice traffic These strategies include using class of service CoS prioritizing ports prioritizing services and using IEEE 802 1p Q to set the priority b
11. small networks complex networks How performed Remotely On Site Test with simulated VoIP traffic No Yes Diagnose problems No Yes Recommend or take corrective action No Yes Test various VoIP configurations No Yes Identify congestion points Sometimes Yes Discover topology Yes Yes Level of confidence that network is ready Low Medium High 8 PC Considerations using Avaya s One X Communicator Avaya s One X Communicator is PC software that simulates a telephone The perceived audio voice quality at the PC endpoint is a function of at least four factors 1 Transducer Quality The selection of speaker and microphone or headset has an impact on the reproduction of the sound 2 Sound Card Quality There are several parameters that affect sound card quality The most important is whether or not the sound card supports full duplex operation 3 End to End Delay A PC can be a major component of delay in a conversation PC delay consists of the jitter buffer and sound system delays as well as the number of other processes running and the speed of the processor 4 Speech Breakup Speech breakup may be the result of a number of factors Network jitter in excess of the jitter buffer size Loss of packets due to excessive delay etc Aggressiveness of Silence Suppression In an effort to reduce network load silence suppression is used to eliminate the transmission of silence However some silence suppression algorithms
12. AT Due to limitations in the H 323 VoIP standard VoIP conversations rarely work across NAT boundaries It is important to route voice streams around routers or firewalls running NAT or use a H 323 friendly NAT e Virtual Private Networks VPN VPNs present interesting challenges to VoIP implementations First the encryption used with VPNs adds significant latency to voice streams adversely affecting the user experience Second VPNs generally run over the Internet Because there is no control over QoS parameters for traffic crossing the Internet voice quality may suffer due to excessive packet loss delay and jitter For more information please refer to Avaya s VPN white paper Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 24 Appendix B The nature of frame relay poses somewhat of a challenge for VoIP This document presents a frame relay overview and then discusses an issue that affects VoIP across frame relay links Overview Frame relay service is composed of three elements the physical access circuit the frame relay port and the virtual circuit The physical access circuit is typically a T1 or fractional T1 and is provided by the local exchange carrier LEC between the customer premise and the nearest central office CO The frame relay port is the physical access into the frame relay network a port on the frame relay mux itself The access circuit rate and the frame relay port rate must match The virtual circ
13. AVAYA Issue 3 2 labs AVAYA IP VOICE QUALITY NETWORK REQUIREMENTS White paper Issue 3 2 August 2009 Developed by Avaya Inc Westminster Colorado Copyright 2009 Avaya Inc All Rights Reserved 1 Copyright 2009 Avaya Inc All Rights Reserved Printed in U S A TRADEMARK NOTICE Avaya and the Avaya Logo are trademarks of Avaya Inc and may be registered in certain jurisdictions All trademarks identified by or are registered trademarks and trademarks respectively of Avaya Inc All other trademarks are the property of their respective owners NOTICE While reasonable efforts were made to ensure the information in this document was complete and accurate at the time of printing Avaya can assume no responsibility for any errors Changes and corrections to the information contained in this document may be incorporated into future releases Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 2 Contents 1 DOGUIMENE SUMMARY EE 4 Avaya IP Voice Quality Network Requirements cccicecceeee cece eee e eee nen teense ne teat n tna tas 7 EXPIAMATIONS arenan ech ide wana tadveresesneed oamedated oncewes iveceded dee 7 2 MMCFOUU COM EE 7 3 Defining Quality EE 8 3 1 What iS el EE 8 3 2 What iS Voice QUAIILY NENNEN 8 4 Prioritizing VOICE Trafi oreore Seege SNE ENER EAA NEE NAAA 8 4 1 Understanding the difference between CoS and QOS sssssssssssrrssrrrserrrssrrrnsrrrsrrnns 8 4 2 USING eu EE 9 4
14. P quality are delay jitter and packet loss These elements are defined and influenced in the transport of IP both within and outside an enterprise To ensure good and consistent levels of voice quality Avaya suggests the following network parameters Note that these suggestions hold true for LAN only and LAN WAN connectivity All requirement values listed are measured between endpoints because this document assumes that IP telephony has not yet been implemented All values therefore reflect the network s performance without endpoint consideration This is why there is seemingly a discrepancy between the well known ITU T value for one way delay and the values listed The ITU T values are end to end values from the mouth of the transmitter to the ear of the receiver The network requirements listed are meant for the network only between endpoints so that your business data network can be assessed and modified if need be for successful deployment of real time applications like voice and video The requirement values are also useful for ongoing network monitoring by IT staff Upward trends in delay jitter or packet loss serve as a warning of potential voice quality problems Also please note that Business Communication Quality is defined as slightly less than toll but far better than cell phone quality This is the tier where most businesses experience the best trade off between voice quality and network infrastructure costs Requirements
15. Software feature called DCS with Rerouting Path Replacement This feature detects that the call coming through the main ECS has been routed from one tandem ECS through the main and back out to a third switch In these cases the system then re routes the call directly thus replacing the path through the main system with a more direct connection Avaya products minimize transcoding while non Avaya products may cause slight to excessive transcoding Shuffling and Hairpinning also reduce transcoding 5 6 Echo The two main types of echo are acoustic and electrical impedance although the sources of echo can be many Echo will result when a VoIP call leaves the LAN through a poorly administered analog trunk into the PSTN Another major cause is an impedance mismatch between four wire and two wire systems Echo also results when an impedance mismatch exists in the conversion between the TDM Time Division Multiplexing bus and the LAN or the impedance mis match between a headset and its adapter Impedance mis match causes inefficient energy transfer The energy imbalance must go somewhere and it is reflected back in the form of an echo Usually the speaker hears the echo but the receiver does not Echo cancellers which have varying amounts of memory compare the received voice with the current voice patterns If the patterns match the canceller cancels the echo Echo cancellers aren t perfect however Under some circumstances the echo gets past the
16. after a Class of Service CoS mechanism tags and Network elements treat voice packets as having priority over data packets Networks with periods of congestion can still provide excellent voice quality when using a QoS CoS policy Switched networks may use IEEE 802 1p Q Routed networks should use DSCP DiffServ Code Points Mixed networks may use both as a best practice Port priority can also be used to enhance DiffServ and IEEE 802 1p Q Even networks with plentiful bandwidth should implement CoS QoS to protect voice communications from periods of unusual congestion such as from a computer virus See sections 3 1 3 4 for more information Switched Network A fully switched LAN network is a network that allows full duplex and full endpoint bandwidth for every endpoint that exists on that LAN Although VoIP systems can work in a shared hubs or bussed LAN Avaya recommends the consistently high results a switched network lends to VoIP Network Assessment A Basic Network Readiness Assessment Offer from Avaya is vital to a successful implementation of VoIP products and solutions Contact the Avaya Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 5 representative or authorized dealer to review or certify your network Section 7 Network Assessment explains the options available with this offer e VLANs Placing voice packets on a separate VLAN subnet from data packets is a generally accepted practice to reduce both broadcast
17. and data traffic from contending for the same bandwidth as voice Other benefits become available when using VLANs but there may be a substantial cost for initial administration and maintenance Section 3 5 Using VLANs further explains this concept Cautions Avaya also recommends caution when using the following e NAT Be cautious when using NAT Network Address Translation Some implementations using VoIP endpoints behind NAT fail because H 323 messages contain multiple instances of the same IP address in a given message and NAT can fail to find and translate all of them Avaya s Communication Manager will work seamlessly with any static NAT application even if that NAT is not H 323 aware See section 10 4 Network Address translation and Appendix C for more information on using NAT e Analog Dial Up Be careful using analog dial up bandwidth lt 56K to connect two locations Upstream bandwidth is limited to a maximum of 56K but in most cases is less This results in insufficient bandwidth to provide toll quality voice Some codecs and network parameters provide connections that are acceptable but consider each connection individually e VPN Use Virtual Private Network VPN cautiously with VoIP applications Older systems can have large delays due to encryption decryption and additional encapsulation Many hardware based products encrypt at near wire speed and can be used Additionally if the VPN routes over the Internet wi
18. ck CRC The CRC is a mathematically derived value used to verify that the frame was received uncorrupted Again these four bytes take time to transmit on the network and should also be included in the bandwidth calculations Switch throughput calculations often do not include the preamble or the trailer The first column of values doesn t really apply to the business world Data switch manufacturers use these values for switch backplane speed calculations The second column of values is useful if VLAN numbers are not included in the transmission Before VoIP is implemented on a network these values are used The last column of values in light gray are accurate when using separate voice and data VLANs as Avaya recommends The ERGI field is the bandwidth for the default voice payload size set in Communication Manager Software 7 Network Assessment The Avaya Network Assessment for IP Telephony Solutions Offer is designed to provide assurance to Avaya customers that their data network is capable of supporting Voice over IP VoIP applications before installation of any Avaya application This Network Assessment for IP Telephony Solutions Offer is a flexible process allowing the customer to provide the Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 16 required network assessment data themselves or provide the data to Avaya through their network vendor The Network Assessment Services for IP Telephony Solutions consist of two d
19. ddress in the IP header resulting in a mismatch that prohibits the control of calls Avaya suggests using a firewall to guard against intruders but the firewall should not provide NAT functions for VoIP packets unless it is H 323 aware Appendix C shows an approved sample implementation of a firewall using selective NAT Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 21 Appendix A Network Design Recommendations In the early days of Local Area Networking network designers used hubs to attach servers and workstations and routers to segment the network into manageable pieces Because of the high cost of router interfaces and the inherent limitations of shared media hubs network design was fairly simple In recent years with the rise of switches to segment networks designers could hide a number of faults in their networks and still get good performance As a result network design has suffered VoIP will place new demands on the network Sub optimal designs will not be able to cope with these demands Even with switches installed a company must pay attention to industry pest practices in order to have a properly functioning voice network Because most users will not tolerate poor voice quality administrators should implement a sound network before beginning VoIP pilots or deployments Best practices Industry best practices dictate that a network be designed with the following factors in mind e Reliability redundancy
20. different types of Ethernet frames but let us confine this discussion to the most popular one Ethernet Version 2 EV2 The EV2 frame looks like this Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 15 Frm Chk Seq 4 octets Data Type 6 octets 6 octets 2 46 to 1500 octets Dest Address Src Address Many different values can be calculated using this frame alone depending on voice payload size and the inclusion of preamble or the 802 1Q shim The next table displays LAN bandwidth using a G 711 codec G 711 is most often used in the LAN because it gives the best voice quality and bandwidth is relatively inexpensive and plentiful compared to the WAN LAN Bandwidth using G 711 codec in kbps Ethernet Type EV2 with trailer EV2 with trailer EV2 with trailer but no and preamble and preamble preamble and 802 1Q 110 4 116 8 120 0 87 2 90 4 79 5 81 6 75 6 77 2 78 0 73 3 74 6 75 2 71 7 72 8 73 3 Notes The preamble consists of 8 bytes of alternating 1 s and O s with the last 2 bits both being 1 s This is the synchronization method used to alert all nodes on the asynchronous network or subnet that a frame is coming Since the preamble takes time to impress bits onto the network it should be counted in the overall bandwidth calculation The trailer is 4 bytes called the Frame Check Sequence FCS containing the results of a cyclical redundancy che
21. e Scalability e Manageability e Bandwidth Voice mandates the following additional considerations when designing a network e Delay e Jitter e Loss e Duplex Generally speaking these concerns dictate a hierarchical network consisting of at most three layers core distribution and access Some smaller networks can collapse the functions of several layers into one device The core layer is the heart of the network Its purpose is to forward packets as quickly as possible It should be designed with high availability in mind Generally these high availability features include redundant devices redundant power supplies redundant processors and redundant links In the current era core interconnections increasingly use Gigabit Ethernet The distribution layer links the access layer with the core It is here that QoS feature and access lists are applied Generally Gigabit Ethernet connects to the core and either Gigabit Ethernet or 100base TX FX links connect the access layer Redundancy is important at this layer but not as important as in the core The access layer connects servers and workstations Switches at this layer are smaller usually 24 48 ports Desktop computers and workstations are usually connected at 10 Mbps or 100Mbps and servers are connected at 100 Mbps or 1 Gbps Limited redundancy is used Some QoS and security features can be implemented here For VoIP to work well WAN links should be properly sized with suffici
22. ed by a firewall which filters unwanted traffic and protects against Denial of Service attacks firewall dependent 4 This architecture simplifies maintenance and monitoring 5 To add a DMZ to a firewall generally requires only the addition of two NIC cards or a quad card at minimal cost firewall dependent 6 TDM calls through the Definity or hairpin calls should process correctly 7 Firewalls can be load balanced with third party software or hardware for greater performance and reliability 8 This architecture represents the current industry best practices Issue 3 2 Private LAN Firewall Les DMZ Web Serer AMM ICM Server I Telephone Centreu CMS 2 Shuffled calls that is calls made from IP Softphones or IP telephones directly to a client on the Internet are not permitted due to NAT issues This can be remedied by locating the IP telephones on the Definity DMZ Copyright 2009 Avaya Inc All Rights Reserved 27 History Issue 3 0 Issue 3 1 Issue 3 2 Issue 3 2 Added bandwidth tables for all supported codecs and all payload options Changed bandwidth tables to add G 726 codec Removed descriptions of general server and gateway offerings Added general updates inclusion of MPLS as a WAN transport and the use of Auto negotiation as a best practice for servers and IP based Circuit packs Copyright 2009 Avaya Inc All Righ
23. ed packets in the phone buffers and it may appear that the network is losing packets when in fact they have been discarded intentionally 5 4 Network Packet Mis Order Network packet mis order is for VoIP very much like packet loss If a packet arrives out of order it is generally discarded as it makes no sense to play it out of order and buffers are small Specifically packets are discarded when they arrive later than the jitter buffer can hold them Mis order can occur when networks send individual packets over different routes Planned events like load balancing or unplanned events such as re routing due to congestion or other transient difficulties can cause packet mis order Packets traversing the network over Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 12 different routes may arrive at their destination out of order Network latency over multiple yet unequal routing paths can also force packet mis order 5 5 Transcoding Transcoding is a voice signal converted from TDM to IP or IP to TDM with or without compression and decompression If calls are routed using multiple voice coders as in the case of call coverage on an intermediary system back to a centralized voice mail system the calls may experience multiple transcoding including the one in and out of the voice mailbox Each transcoding episode results in some degradation of voice quality These problems may be minimized by the use of the Communication Manager
24. ent bandwidth for voice and data traffic Each voice call uses between 6 3 Kbps and 80 Kbps depending on the desired codec quality and header compression used G 729 is one of the most promising standards today using 24 Kbps of bandwidth Interoffice bandwidth demands can be sized Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 22 using traditional phone metrics such as average call volume peak volume and average call length Quality of Service also becomes increasingly important with WAN circuits In this case Quality of Service can be taken to mean classification and prioritization of voice traffic Voice traffic should be given absolute priority through the WAN and if links are not properly sized or queuing strategies are not properly implemented it will become evident both with the quality and timeliness of voice and data traffic There are four technologies that work well with VoIP ATM Frame Relay MPLS and point to point PPP circuits These technologies all have good throughput low latency and low jitter ATM has the added benefit of enhanced QoS Frame Relay MPLS and PPP links are more economical but lack some of the traffic shaping features of ATM Of the four technologies Frame Relay is the most difficult WAN circuit to use with VoIP Congestion in Frame Relay networks can cause frame loss which can significantly degrade the quality of VoIP conversations With Frame Relay proper sizing of the CIR committed
25. est statistical variance between packets Router vendors have many queuing methods that alter the behavior of the jitter buffer It is not enough to select the right size of jitter buffer one must also pair an appropriate queue unloading algorithm type with the jitter buffer The network topology can also affect jitter Because there are fewer collisions on a data switched network than on a hub based network there will be less jitter on the switched network Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 11 The Avaya G650 and G450 etc media gateways Avaya Communication Manager Server Avaya One X software and Avaya IP telephones have all incorporated dynamic jitter buffers to minimize delay by reducing the jitter buffer size as the network allows Note that this feature can exacerbate problems in an uncontrolled network Many good tools are commercially available to measure jitter delay and packet loss to help monitor and bring control to the network 5 3 Packet Loss Network packet loss occurs when packets are sent but not received at the final destination due to some network problem Problems caused by occasional packet loss are difficult to detect because each codec has its own packet loss concealment algorithm Therefore it is possible that voice quality would be higher using a compression codec G 729A compared to a full bandwidth G 711 codec Several factors make packet loss requirements somewhat variable such a
26. ewdseiduesa nase ent eatvid amecevoedd esther ean EEEN AEE conan neem armas 22 Network Design Recommendations ccceceee eee ene enna nena 22 BES ACEI COS eis iayat EE DEE 22 COMMON ISSUES oeenn EEEE sans Suk eer Tear ebais Sue meN ear eaanN eye EERON 23 ADPENCIX Biseve EE 25 EM EENEG ee e See event ge ee de eege gege de ege EE s 25 Issue and Alternatives ed ekeguusg ee eg ed KENNS eE dE SN Nee Ed ENN RENE ENNEN EEN d EN REN Seed er 26 Additional Frame Relay Information 26 AD DEIN EE 27 VoIP without using NAT encsieriavce cena chery ses cessed cians SSES ENEE NEEN mare AENEA 27 Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 3 Avaya IP Voice Quality Network Requirements 1 Document Summary This document contains basic network requirements that are foundational for good voice quality when using Avaya IP products and solutions over a data network No document can satisfy the detailed needs of every network and therefore this paper serves only as a starting point The document summary provides a short list of networking requirements allowances and recommendations Use this page as a checklist to determine if the network meets the minimum requirements for implementing Voice over Internet Protocol VoIP with acceptable quality The rest of the document contains basic networking and telephony concepts for those who haven t been exposed to a converged implementation It also explains why VoIP applications can yield poor
27. ht 2009 Avaya Inc All Rights Reserved 13 5 8 Network Duplex The ideal LAN network for transporting VoIP traffic is a network that is fully switched from end to end This is a full duplex network A network that has shared segments hub based is a half duplex network and can result in lower voice quality due to excessive collisions and should be avoided Ethernet connections from Avaya IP phones default to auto negotiation for speed and duplex to work seamlessly with Ethernet switches immediately Avaya recommends using auto negotiation for endpoints IP telephones servers and IP based Circuit Packs An acceptable alternative is to set speed and duplex values for Avaya IP Circuit Packs to 100Mb and Full duplex 5 9 Codec Selection Depending upon the bandwidth availability and acceptable voice quality it might be worthwhile to select a codec that produces compressed audio e AG 711 codec produces audio uncompressed to 64 kbps e AG 726 codec produces audio compressed to 32 kbps in Avaya s implementation although four different rates are available e AG 729 codec produces audio compressed to 8 kbps The following table provides comparisons of several voice quality considerations associated with some of the codecs supported by Avaya products Real toll quality voice must achieve a MOS Mean Opinion Score of 4 or above MOS scoring is technically a subjective method of measuring voice quality True Table 1 Comparison of Speech Cod
28. ing Standards Standard Coding Type Bit Rate kbps MOS G 711 PCM 64 4 3 G 726 ADPMC 32 4 2 G 729 CS ACELP 8 4 0 Generally G 711 is used within LANs because bandwidth is abundant and inexpensive whereas G 729A is used across WAN links because of the bandwidth savings and good performing voice quality G 726 is a newer codec choice that improves voice quality over G 729 and uses half the bandwidth of G 711 6 Bearer Bandwidth There are bandwidth requirements for both bearer voice and signaling This document covers bearer requirements only The following table shows the differences of various codecs with varying voice payload sizes for bearer traffic only Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 14 6 1 WAN Bandwidth Comparison Examples of Layer 2 protocols include Ethernet Frame Relay PPP ATM and others Note that 8 bytes was used for the layer 2 calculation contribution because the most used data protocols Frame Relay and PPP can use 8 bytes or less WAN bandwidth is expensive compared to LAN bandwidth driving the need to use compression in both voice payloads and possibly the headers of the TCP IP protocol stack The next table shows bearer bandwidth required using OSI layers 2 through 7 and several codecs for WAN connections G 729 is the most popular codec over a WAN The values in the dark gray cells represent bandwidth using the Avaya default of a 20ms payload
29. istinct services Avaya offers these two services to help customers determine if their data networks are ready for converged voice and data networking What is a BASIC A Basic Assessment BASIC is a high level survey of a customers network The BASIC consists of two parts a written survey that the end user fills out describing their voice and data network and a data gathering application The application is installed on a computer in the customers network and runs for a period of time usually a week The application discovers network devices e g routers and switches and monitors their performance What does a BASIC actually test A BASIC is a high level test of network health It is designed to identify gross network problems such as oversubscribed WAN links routing loops and overloaded devices The BASIC application runs from a central location identifying network devices i e discovery and then monitoring basic performance parameters via SNMP polling By monitoring a router s interface parameters the application can determine if the interface is oversubscribed dropping packets or experiencing errors The data gathered is presented as a map of the customers network and a graphs outlining device performance Customers who do not avail themselves of this offer assume responsibility for all network related problems with the IP Voice installation Also Avaya personnel may be required to charge a higher T amp M Time and Mate
30. its Avaya products are designed to work with most other popular switches and routers using open standards to provide end to end voice prioritization 4 1 Understanding the difference between CoS and QoS Class of Service CoS is a classification method only CoS does NOT ensure a level of Quality of Service QoS but is the method used by queuing mechanisms to limit delay and other Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 8 factors to improve QoS Most CoS strategies assign a priority level usually 0 7 or 0 63 toa frame or packet respectively Common CoS models include the IP TOS Type Of Service byte Differentiated Services Code Point DiffServ or DSCP defined in RFC 2474 and others and the IEEE 802 1p Q Quality of Service QoS involves giving preferential treatment through queuing bandwidth reservation or other methods based on attributes of the packet such as CoS priority A service quality is then negotiated Examples of QoS are CBWFQ Class Based Weighted Fair Queuing RSVP RESERVATION Protocol RFC 2205 MPLS Multi Protocol Label Switching RFC 1117 and others CoS or tagging is totally ineffective in the absence of QoS because it can only mark data QoS relies on those tags or filters to give priority to data streams 4 2 Using Ports One prioritization scheme assigns priority based on the UDP User Datagram Protocol port numbers used by the voice packets This scheme allows the use of networ
31. k equipment to prioritize all packets from a port range UDP is used to transport voice packets through the LAN because unlike TCP it is not connection based Because of the human ear s sensitivity to delay it is better to drop packets rather than retransmit voice in a real time environment so a connectionless protocol is preferable to a connection based protocol By using Communication Manager Software users can define a UDP port range for voice priority Routers and layer 3 data switches can then use these ports to distinguish priority traffic This priority traffic can be voice packets UDP signaling packets TCP or both This is an OSI model layer 4 solution and works on data coming to and from the specified ports or a port range 4 3 Using DSCP or TOS The DSCP prioritization scheme redefined the original Type of Service TOS byte in the IP header by combining the first six bits into 64 possible combinations This use of the TOS byte is used by Communication Manager Software IP Telephones and other network elements such as routers and switches in the LAN and WAN A DSCP of 46 101110 is suggested for the expedited forwarding of voice packets However with Communication Manager one can set any DSCP value as desired to work with a company s QoS scheme Note that older routers may require a DSCP setting of 40 101000 which is backward compatible to the original TOS byte definition of critical But again Avaya products and softwa
32. m 1Gbps versus traversing switch fabric up to 256 Gbps The greater the number of small switches layers the greater the number of uplinks and the lower the bandwidth for an individual connection Under a network of this type voice performance can quickly degrade to an unacceptable level e Multiple subnets on a VLAN A network of this type can have issues with broadcasts multicasts and routing protocol updates It should be avoided It can greatly impact voice performance and complicate troubleshooting issues e Hub based network Hubs in a network create some interesting challenges for administrators It is advisable not to link more than four 10baseT hubs or two 100baseT hubs together Also the collision domain the number of ports connected by hubs without a switch or router in between should be kept as low as possible Finally the effective half duplex bandwidth available on a shared collision domain is approximately 35 of the total bandwidth available Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 23 e Too many access lists Access lists slow down a router While they are appropriate for voice networks care must be taken not to apply them to unnecessary interfaces Traffic should be modeled beforehand and access lists applied only to the appropriate interface in the appropriate direction not all interfaces in all directions Additional concerns when implementing VoIP include e Network Address Translation N
33. may clip speech and have an effect on perceived audio quality Performance bottleneck in the PC Lower speed PCs or PCs with slow hard drives may have adverse interactions with sound playback and recording This can cause breaks in received or transmitted audio The best thing to do in this situation is to increase the processor speed increase the amount of RAM and or reduce the number of applications competing for the processor or hard drive resources One notable resource consumer is the Microsoft Find Fast program that launches from the Startup folder and runs in the background This application periodically re indexes the hard drive and consumes significant PC resources in the process Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 18 9 Bandwidth Requirements The bandwidth available to the user is very important Access to the network using slower connections such as dial up connections will degrade voice quality The best voice quality is achieved in both LANs and WANs when the bandwidth is owned by the customer Customer owned bandwidth can be shaped to optimize VoIP traffic Conversely bandwidth that is not controlled like the Internet cannot give consistent sound quality because it cannot be optimized for VoIP Because factors of delay jitter and packet loss are exacerbated over the Internet we do not recommend using the Internet for voice applications at this time 9 1 Bandwidth Requirements using 1 X C
34. ommunicator or IP Agent A dual connect system is commonly used in a Call Center for users working remotely The PC and the telephone can transmit frames across the same telephone line or on two lines Questions concerning the amount of bandwidth the PC uses and its effect on voice are answered here The bandwidth used by the PC for signaling is very low However it is difficult to express this value in bits per second due to the variability in how quickly the buttons are pressed and how many feature buttons are used during a call The following graph is a 50 second average call showing the bandwidth needed with several buttons pushed Remember that even with a 56K V 90 modem the upstream bandwidth is no greater than 33 6K and the downstream is anywhere from 28 8K to 53K The speed of each connection is determined by the PSTN line conditions at the time the call is placed IP Agent Average Dual Connect Bandw idth Requirements Used for Signalling 1600 1400 1200 wal I e I I ee a GIN a BE TRA Bits Transmitter Note that during most of this call the bandwidth required is zero X Axis The maximum bandwidth needed is never greater than 1 450 Kilobits at any one point in time This is small compared to even a slow 28 8 Kilobit transfer rate as it represents less than 5 of the 28 8Kbs available bandwidth at any point in time Bandwidth required for signaling is almost moot compared to the available bandwidth for voice Signaling is
35. r to choose which quality level best suits their specific business needs 5 1 Network Packet Delay Network packet delay is the length of time it takes a packet to traverse the network Each element of the network adds to packet delay including switches routers and the distance traveled through the network firewalls and jitter buffers such as those built into H 323 audio applications like the Avaya IP SoftPhone or Microsoft NetMeeting Router delay depends not only on hardware but also on configurations such as access lists queuing methods and transmission modes Delay latency can have a noticeable affect but can be somewhat controlled in a private environment LAN WAN because the enterprise manages the network infrastructure or SLA When using the public network there are inherent delays that one cannot control The next page suggests guidelines for one way network delay Again there is a trade off between voice quality and the technical and monetary constraints with which businesses confront daily The E Model a voice quality measurement algorithm lists tiers of delay for voice quality These tiers directly relate to Avaya s suggestion for delay parameters The Tiered measurements from the E Model R Values are as follows Best Quality is rated toll quality or above R94 very satisfied Business Communication Quality is R80 R92 satisfied Possibly acceptable is rated R70 R80 Some unsatisfied Issue 3 2 Copyrigh
36. rden on the IP telephones and PCs by freeing them from having to analyze irrelevant broadcast packets VLANs a layer 2 feature are created in data switches A voice VLAN can be manually applied to an IP telephone or provided by a DHCP server CoS tagging and QoS policies can be applied at OSI layer 2 by using VLANs Separate voice and data VLANs are an option that makes sense for most customers and is highly recommended by Avaya Not only should IP phones be in a VLAN separated from data but the IP cards like the MedPro CLAN and VAL should also be in a VLAN devoid of data Note however that VLAN implementation and maintenance can be substantial and again is an option even as a best practice Proper VLAN implementation is not trivial and Avaya can help with planning and implementation through its Converged Services Group 5 Network Parameters There are a number of network parameters that affect voice quality This section lists some of the more important ones The concept of quality has different meanings to different people IP telephony quality can be engineered to several different levels to accommodate differing business needs A small company may choose to implement IP telephony with very good sound instead of buying newer networking equipment to support excellent voice sound A large call center company may want excellent voice sound as part of its corporate strategy Avaya therefore presents options in network requirements to allow the custome
37. re allows users to set any of the 64 possible DSCP values to work with your voice quality policy The TOS byte is an OSI model layer 3 solution and works on IP packets on the LAN and possibly the WAN depending on the service provider 4 4 Using IEEE 802 1 p Q Yet another prioritization scheme is the IEEE 802 1Q standard which uses four bytes to augment the layer 2 header IEEE 802 1Q defines the open standard for VLAN tagging Two bytes house 12 bits used to tag each frame with a VLAN identification number The IEEE 802 1p standard uses three of the remaining bits in the 802 1Q header to assign one of eight different classes of service Again with Communication Manager Software users can add the 802 1Q bytes and set the priority bits as desired Avaya suggests you use a priority of Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 9 6 for both voice and 5 for signaling IEEE 802 1p and IEEE 802 1Q are OSI layer 2 solutions and work on frames 4 5 Using VLANs VLANs provide limited security and create smaller broadcast domains through software by creating virtually separated subnets Broadcasts are a natural occurrence in data networks from protocols used by PCs servers switches routers NOS etc Creating a separate VLAN for voice reduces the amount of broadcast traffic and unicast traffic on a shared LAN that the telephone will receive Separate VLANs result in more effective bandwidth utilization and reduces the processor bu
38. results when data traffic on the same network doesn t seem to have problems Voice quality is always a subjective topic Defining good voice quality varies with business needs cultural differences customer expectations etc The requirements below are based on the ITU T EIA TIA guidelines and extensive testing at Avaya Labs Note that while Avaya s requirements will meet or exceed most customer quality expectations the final determination of acceptable voice quality lies with the customer s definition of quality and the design implementation and monitoring of the end to end data network Quality is not measured by one discrete value where a number where 8 is good and 9 is bad There is a tradeoff between real world limits and acceptable voice quality Lower delay jitter and packet loss values can produce the best voice quality but may also come with a cost to upgrade the network infrastructure to get to the lower network values Another real world limit is the inherent WAN delay over a trunk linking for example the U S West coast to India This link could easily add a fixed 150ms delay into the overall delay budget and is beyond the control of an enterprise Perfectly acceptable voice quality is attainable but will not be toll quality Therefore Avaya presents a tiered choice of values that make up the requirements Voice quality is made up of both objective and subjective contributors The objective elements in assessing VoI
39. rials rate if assistance is needed since troubleshooting will be more difficult without the assessment data What is a DETAILED A Detailed Assessment is an in depth assessment designed to accurately determine a customer s readiness for VoIP The DETAILED consists of two parts the first part analyses the network by discovering network devices and then injecting simulated VoIP traffic into the network at selected locations While simulating the traffic it simultaneously measures network device performance The analysis not only identifies problems it determines the root cause of the problem so that it can be corrected The second part of the DETAILED the network optimization is a service that defines the requirements for optimizing a network for VoIP and correcting problems found during the assessment It provides the customer with the steps that need to be performed in order to make the network VoIP ready In addition this service can make corrections to the customer s network A variety of services can be performed including re engineering portions of the network reconfiguring devices and other actions How to Choose between a BASIC and DETAILED The following table should be used as a guide for deciding which service best meets the customer s requirements Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 17 Condition BASIC DETAILED Network Size 1 2 locations 2 or greater
40. s the following e Packet loss requirements are tighter for tones other than DTMF than for voice The ear is less able to detect packet loss during speech variable pitch than during a tone consistent pitch This includes fax TTY and modem over IP transmissions e Packet loss requirements are tighter for short continuous packet loss than for random packet loss over time Losing ten contiguous packets is worse than losing ten packets evenly spaced over an hour time span e Packet loss may be more noticeable for larger voice payloads than for smaller ones because more voice is lost in a larger payload e Packet loss may be more tolerable for one codec over another e Even small amounts of packet loss can greatly affect TTY TDD device s ability to work properly e Packet loss for TCP signaling traffic increases substantially when loss is over 3 due to retransmissions Network packet loss The maximum loss of packets or frames between endpoints should be e 1 or less can yield toll quality depending on many factors e 3 or less should give Business communications quality Again this quality is much better than cell phone quality e More than 3 may be acceptable for voice but may interfere with signaling Like delay values Avaya gives customers a tiered approach of packet loss to balance network costs and limitations with business directives Remember that too much delay jitter or packet mis order can cause dropp
41. ssue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 7 This document provides basic network guidelines to ensure good voice quality when implementing VoIP This document also examines some of the more important components that affect VoIP and gives suggestions to help avoid problems during implementation 3 Defining Quality 3 1 What is Quality Quality is a word that is used by almost all manufacturing and service providers Quality however is an ambiguous term representing superiority of that product or service But quality can mean different things to different people Consider Bill a person who wants to buy a quality vehicle Bill goes to a dealership and sees a luxury sports sedan It is a quality vehicle The stitching on the leather seats is uniformly 0 2 apart on all seams The finish consists of 10 color coats and 2 high polymer sealant coats The fit between the doors and the body is consistently 0 167 Bill buys the luxury sports sedan and is happy with the quality Now consider Trish a person who also wants to buy a quality vehicle Trish lives in rugged mountain terrain miles from anyone and must cross a boulder field just to get to work Trish is looking for a vehicle that has high ground clearance a stiff suspension and 4 wheel drive to get her to town and back consistently without breaking down Trish buys a Sport Utility Vehicle and is happy with the quality Trish doesn t care about the stitching
42. t 2009 Avaya Inc All Rights Reserved 10 Avaya s Tier suggestions are Network delay Between endpoints meaning LAN WAN measurements not including IP phones e 80ms milliseconds delay or less can but may not yield toll quality e 80ms to 180ms delay is considered business communication quality This is much better than cell phone quality and in fact is very well suited for the majority of business needs e Delays exceeding 180ms may still be quite acceptable depending on customer expectations analog trunks used codec type etc The ITU T has recommended 150ms one way delay including endpoints as the limit for excellent voice quality This value is largely misinterpreted as the only range to calculate a network delay budget for IP telephones A network delay budget of 230ms proved almost imperceptible in lab experiments at Avaya One way network delays in excess of 250ms can cause the well known problem of talk over when each person starts to talk because the delay prevents them from realizing that the other person has already started talking Certainly long WAN transports must be considered as a major contributor to the network delay budget one major WAN service provider averaged 75ms delay from Los Angeles to New York Los Angeles to Paris was found to be about 145ms Some WAN service providers can lower delay in their network if it is negotiated and recorded as part of the SLA Service Level Agreement Even so s
43. t most IXCs convert the long haul delivery of frame relay into ATM That is the frame relay PVC is converted to an ATM PVC at the first frame relay switch after leaving the customer s premise It is not converted back to frame relay until the last frame relay switch before entering the customer s premise This has significance because ATM has built in Class of Service CoS A customer can contract with a carrier to convert the frame relay PVC into a Constant Bit Rate CBR ATM PVC ATM CBR cells are delivered with lower latency and higher reliability As a final note the reader should understand that under the best circumstances frame relay is still inherently more susceptible to delay than ATM or TDM Therefore after applying the best possible queuing mechanism one should still expect more delay over frame relay than would be present over ATM or TDM Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 26 Client Endpoint ie a P330 Switch Definity DMZ Zone of Control Appendix C VoIP without using NAT Prowler Definity Not NATed CentreVu CT Message Care Server Email Fax server Se Router Features 1 The TN799C and TN2302AP are attached to the LAN through a separate DMZ off the firewall 2 Connections internal or external to the TN799C and TN2302A4P are not subject to Network Address Translation 3 The Definity DMZ is still protect
44. taying within 150ms end to end may not be possible Finally one way end to end delay over 300ms can cause port network instability A network assessment is highly recommended to measure latency and other factors and make recommendations to solve any latency issues before implementing a VoIP solution 5 2 Network Jitter Jitter is a measure of variance in the time it takes for communications to traverse from the sender application to the receiver as seen from the application layer from RFC 2729 Taxonomy of Communication Requirements for Large scale Multicast Applications Jitter is thought of as the statistical average variance in delivery time between packets or datagrams Jitter can create audible voice quality problems if the variation is greater than 20ms assuming an existing 20ms packet size Symptoms of excessive jitter are very similar to symptoms of high delay because in both cases packets are discarded if the packet delay exceeds half the jitter buffer size To compensate for network jitter many vendors implement a jitter buffer in their voice applications The purpose of the jitter buffer is to hold incoming packets for a specified period of time before forwarding them to the decompression process A jitter buffer is designed to smooth packet flow eliminate jitter In doing so it will also add packet delay Jitter buffers should be dynamic to give the best quality or if static should generally be sized to twice the larg
45. thout SLAs in place sufficient quality for voice cannot be guaranteed unless delay jitter and packet loss adhere to the parameters listed above See section 9 2 VPN Virtual Private Network for more information Issue 3 2 Copyright 2009 Avaya Inc All Rights Reserved 6 Avaya IP Voice Quality Network Requirements Explanations 2 Introduction Voice over Internet Protocol VoIP is the convergence of traditional voice onto an IP data network to provide better application integration by using a common protocol and to lower costs by using ISPs and melding separate support staffs Other real time traffic such as uncompressed video and streaming audio is also converging onto data networks VoIP is very complex because it involves components of both the data and voice worlds Historically these worlds have used two different networks two different support organizations and two different philosophies The voice network has always been separate from the data network because of the protocols used and the characteristics of voice applications are very different from those of data applications Traditionally voice calls have had their own dedicated bandwidth throughout the circuit switched network This provided an environment where five nine of reliability became the standard Interactive voice traffic is sensitive to delay and jitter but can tolerate some packet loss problems that were rarely an issue with circuit switching The
46. to prevent voice traffic from entering the burst range and being marked DE One way to accomplish this is to prohibit bursting by shaping the traffic to the CIR and setting the excess burst size B determines the burst range to zero However this also prevents data traffic from using the burst range as well Another possible alternative is to size the CIR above the peak voice traffic level and then prioritize the voice traffic so that it is always delivered first The underlying assumption here is that the network administrator has an expectation of peak voice traffic By sizing the CIR to meet or exceed the peak voice traffic and then applying priority queuing on the interface so that VoIP is serviced first we can intuitively assure that voice traffic will not enter the burst range The problem with the latter method however is that the actual queuing mechanism is not always intuitive Even though the aggregate voice traffic throughput cannot exceed the CIR it is possible that a voice packet could be sent in the burst range The technical workings of this are beyond the scope of this document But simply stated it is possible that a voice packet would arrive right after many data packets have already been transmitted in the CIR range such that the voice packet ends up in the burst range when the router processes it However the latter method is certainly worth trying Additional Frame Relay Information One good piece of knowledge is tha
47. ts Reserved 28
48. ueuing mechanisms to speed the delivery of Real Time Protocol RTP traffic without significantly increasing protocol overhead or reducing data efficiency Also header compression protocols like CRTP Compressed Real Time Protocol can and should be used between WAN links Hardware based CRTP is effective with very minimal delays but software CRTP can add significant delay 10 2 VPN Virtual Private Network There are many definitions for Virtual Private Networks VPN In this white paper VPNs refer to encrypted tunnels carrying packetized data between remote sites VPNs can use private lines or use the Internet via one or more Internet Service Providers ISP VPNs are implemented in both dedicated hardware and software but can also be integrated as an application to existing hardware and software packages A common example of an integrated package is a firewall product that can provide a barrier against unauthorized intrusion as well as perform the security features needed for a VPN session The encryption process can take from less than 1 milli second to 1 second or more at each end Obviously VPNs can represent a significant source of delay and therefore negatively affect voice performance Also as most VPN traffic runs over the Internet and there is little control over QoS parameters for traffic crossing the Internet voice quality may suffer due to excessive packet loss delay and jitter Users may be able to negotiate a service level
49. uit is a logical connection between frame relay ports that can be provided by the LEC for intra lata frame relay or by the inter exchange carrier IXC for inter lata frame relay The most common virtual circuit is a permanent virtual circuit PVC which is associated with a committed information rate CIR The PVC is identified at each end by a separate data link connection identifier DLCI Houston Branch Office Dallas Corporate Data Center 256k port 1 DLCI200 OLCT30 DLcI300 Denver Branch Office e o XC frame relay i de network DLCI301 E localloop access circuit e permanent virtual circuit Slk par Phoenix Branch Office This hypothetical implementation shows the Dallas corporate office connected to three branch offices in a common star topology or hub and spoke Each office connects to a LEC s CO via a fractional T1 circuit which terminates onto a frame relay port at the CO and onto a frame relay capable router at the customer premise The port rates and the access circuit rates match PVCs are provisioned within the frame relay network between Dallas and each branch office The CIR of each PVC is sized so that it is half the respective port rate which is a common implementation Each branch office is guaranteed its respective CIR but it is also allowed to burst up to the port rate without any guarantees The port rate at Dallas is not quite double the aggregate CIR but it
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