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345 Encinal StreetSanta Cruz, CA 95060 1 (831) 426-5858poly.comPreparing Your IPNetwork for VideoConferencingAND WORKAROUNDS FOR LESS THAN IDEAL CONDITIONSWHITEPAPERPoly Infrastructure EngineeringAugust 30, 2019Copyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved3725-86432-001A

AbstractCarrying real-time audio and video traffic over a network brings a unique set of concerns that require specializedconsiderations when compared to “normal” data traffic (i.e. web pages, e-mail, file downloading, etc.). Attentionmust be paid to designing and configuring IP networks for real-time audio/video (A/V) traffic in order to deliver agood quality experience to all users. Following a set of “best practices” for network design and configuration, as wellas in deployment/configuration of individual products, helps ensure a high-quality encounter for all users. If anetwork is less than ideal and/or “best-practices” cannot be strictly followed, you can employ a variety of remedies tomitigate, if not eliminate, perceived quality issues. Please contact a Poly support representative for further details andassistance.Call Quality IssuesAudio and video participants may experience call quality problems such as choppy audio/video, missed words inspeaking, blurry video, jumbled audio, dropped calls, video artifacts (rainbows, pixilation, lines, shadows, ), “talkover,” and many others. These problems usually stem from one or more of the following conditions, whichparticularly affect the real-time nature of A/V communications: Latency/Delay – Network latency/delay is the time it takes a packet to travel between two points in thenetwork. For A/V communications, this equates to the time between when a person makes a sound andwhen the person on the other end of a call hears that sound. Or it is the time between when a gesture ismade on camera and when the other participants see it happen on their screens. There is always some delayin a network as packets are routed and processed by one device to another on their network journey. Whenthe delays begin to be “too” long, users may begin to experience talk-over, where one person starts talkingover another, because they think it is their “turn.” As delays increase, other quality issues are perceived suchas lip-sync issues (audio and video are not synchronized and people look as if they are being “dubbed”),jumbled audio/video, or even packet loss (as network services may begin discarding outdated packets). Forreference, delays of just 200ms are human-perceptible and over-talk can start happening. Packet Loss – Packets on a network may be discarded or dropped for a variety of reasons: dropping ofoutdated packets, data corruption from interference, poor signal strength (wireless), various networkconfigurations, etc. As packet loss increases, users experience audio/video “cut out,” jumbling, and variouson-screen artifacts. The goal of every network design and operation is to have 0% average packet loss. On acall without any corrective features (such as Poly’s LPR), packet losses going over 1% may start to becomenoticeable by participants, and at 5% audio quality degrades significantly and video (people or content) isgenerally unusable.Packet loss is determined in UC devices by a jump in the UDP sequence number. UC devices usually providedebugging tools to capture network traffic data for analysis. This jump can also be observed by standardnetwork tools. Jitter – Is the inconsistency in delays/latency of packets going between two network points. For A/V calls,packets not travelling smoothly/consistently between two endpoints can result in packets arriving out-oforder and/or getting dropped before processing, which leads to packet loss and the associated call qualityCopyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved2

issues. Average jitter is typically desired to be 40ms or less for good call quality. Jitter is typically the resultof slow network speeds, congested networks, and/or improper network routing configuration. Bandwidth Overuse (network congestion) – Networks, or individual network segments, being overutilized,or having more packets than they can handle at any point in time, can lead to many follow-on problems. Likea road over congested with too many cars on it, packets may fill up a network pipe and becomebottlenecked. The result is that these packets may be delayed (increasing latency and jitter), rerouted(increasing latency and jitter), or outright discarded (increasing packet loss) depending on networkconfigurations and conditions. It is generally recommended to keep peak utilization (the most networkbandwidth used at any point in time) at 75%-80% or less of the total available bandwidth. For example, fora 1 Gbps network link, no more than 750 Mbps of traffic should ever be placed over it at any one time.Fixing the Network and Best-PracticesTo address the conditions that lead to poor call quality, addressing the network itself is usually the most effectiveplace to start.1. Most importantly, monitor the network and the attached infrastructure devices (switches, routers, cables,wireless access points, firewalls, etc. ). Most network devices have built in monitoring facilities to record thepacket loss, jitter, latency, and utilization values, among others. Monitoring devices/services/tools can alsobe attached at various network points to provide greater details. Without monitoring, network issues willonly be known once outages or poor call qualities are reported. Even then, diagnosing the specific causesmay be difficult in any medium to large network.2. Make sure all network hardware is in proper functioning order. Many network issues can be traced back to afaulty cable, loose connection, malfunctioning switch, or the like. The distributed and self-healing nature ofnetworks may not make failing hardware obvious, as degradation rather than full outage, is often the onlysymptom.3. Invest in bandwidth and hardware where needed. Since bandwidth overuse can lead to many networkingproblems, it is important to keep peak utilization at 75%-80% or less between any two points at any time.Upgrading link speed (i.e. 100Mbps - 1Gpbs - 10 Gbps), adding new links, installing faster highercapacity/faster hardware, etc. can help to alleviate both bandwidth and latency concerns. This is generallythe #1 thing that can be done to help resolve call quality problems.4. Ensure your call QoS (Quality of Service) for A/V calling by changing your network configuration (layer 2and/or layer 3) to prefer real time traffic (RTP) over other types of traffic and real-time devices (endpoints,MCUs, etc.) over other types of devices (i.e. web servers). Configure this in the switches, routers, andfirewalls that form the network. There are a variety of schemes that can be employed to accomplish this:DiffServ DSCP packet marking, TOS packet marking, and separate VLANs with high priority routing for A/Vdevices; to name a few. These types of configurations help ensure that the latency, jitter, and packet-loss areminimized automatically by your network if/when bandwidth becomes overutilized. It also helps insulate A/Vtraffic from other non-real-time traffic (i.e. call quality won’t suffer just because someone else is transfering alarge file). Note: In many individual products, particularly Poly devices (phones, video endpoints, andCopyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved3

infrastructure), many of these QoS features are also supported. Together with the rest of the networkinfrastructure configuration, these features can provide for true end-to-end QoS and higher quality calls.Consult individual product manuals for QoS setting capabilities.5. Fix routing and/or other related configuration/issues. Packets not taking the most efficient and/or direct pathbetween two points can lead to increased latency and jitter (and possible packet loss as a result). Often animproperly configured route can be sending packets in an inefficient direction, or even unnecessarily over acongested network path. It is important to properly configure and periodically test network routes, asincorrect routing can creep in, but this may not be immediately obvious except in the possible loss of callquality.6. Deactivate firewall ALG and RTP gateway features. Even though many popular firewalls offer A/V ALG(Application Layer Gateway) features, they often lack the nuances, expertise, and general feature setsneeded to properly implement A/V calling services correctly. As a result, they may end up erroneouslydropping packets, increasing latency, and/or generally interfering with A/V calls in undesirable ways. Wheresecurity is needed, a purpose-built A/V security SBC (Session Border Controller) device, such as the PolycomDMA-Edge system, should be employed. With the proper ports forwarded to the them, these SBC devicestake over the security duties for A/V calls from the firewall and still ensure the full range of A/V callingfeatures, while at the same time minimizing latency, jitter, and packet loss.7. Minimize the use of VPNs for A/V calls. VPNs introduce latency/delays to the A/V packets and often do notsupport the QoS settings/features needed to prioritize real-time-traffic over other types of traffic throughthem. It is desirable to avoid VPNs for all A/V packets wherever possible. With proper firewall configuration,and use of security SBCs, VPNs are often unnecessary to obtain equal or greater security levels.8. Ensure that the network core devices are configured to allow many and rapid UDP-RTP packets from single IPaddress sources. There are security configurations in many switches/routers/firewalls that can interfere withboth the signaling and UDP-RTP packets needed for A/V calling. While in a call, an A/V device may sendseveral UDP-RTP packets every few milliseconds. Network devices should be configured to not see thisvolume of traffic as an “attack” and incorrectly filter (i.e. drop, AKA: packet-loss) these packets.9. Deploy MCUs and other critical A/V infrastructure near the center of the network where bandwidth is usuallyhigher and the average distances, and thus latency, between endpoints and these core components areminimized.10. Remember that the best-practices used to ensure quality of a network within a site should also be employedbetween network sites (i.e. between offices via MPLS links) and up to the internet if A/V calls will traversethese. Cross site and internet uplink providers offer a variety of service levels and options. These ServiceLevel Agreements (SLAs) contractually ensure network characteristics such as bandwidth, maximum latency,and maximum jitter across their links. It is important to size these links correctly for the expected trafficbandwidth.11. Time synchronization is very important in modern networks and many voice and video applications areuniquely susceptible to incorrect time synchronization. Reliable NTP (Network Time Protocol) servers withCopyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved4

very high stratum (1,2, or 3) should be used for all network connected devices, and multiple servers shouldbe configured wherever possible (3 or more recommended).Fixes at the Device and in the ProductAside from general network changes to help support A/V calls, individual A/V devices may have features andconfigurations to help improve overall call quality.1. QoS settings – Many devices, especially Poly devices, support features such as DiffServ and IP Precedencepacket marking, and VLANs. These QoS settings, in conjunction with like configurations in the network itself,can greatly improve call quality.2. Call Quality Selection – When placing an A/V call, the desired maximum call quality (in bit-rate) can often beselected (either configured and/or selected at call time). It is important to properly configure endpointdevices not to place calls with bit-rates higher than desired or for which there is insufficient network capacity.On the network a call will consume its bit-rate roughly a 20% overhead of bandwidth. For example, a 100Mbps link can only handle 10 6Mbps A/V calls, with no other traffic (e.g. (100Mbps * .75) / (6Mbps (.2 *6Mbps))). If more calling capacity is desired, or if other network services use these links, these endpointsshould be configured to use a reduced video quality to accommodate. For context and reference, with astatic background (i.e. not a movie or sporting event video, but rather just a person talking in front of acamera), basic 720p HD video calls can happen with less than 2Mbps and good SD calls can happen with aslittle as 384 -768Kbps.3. LPR (Lost Packet Recovery) – This is a Poly proprietary feature that attempts to recover lost packets at thecost of network bandwidth and some extra packet latency. If both Poly A/V endpoints (audio phones, videophones, room systems, Polycom RMX MCUs, etc.) in a call support the LPR feature (and have it configuredactive), the LPR feature detects any packet loss during a call and automatically attempt to recover the lostpackets. In most environments LPR adds significant improvements;, however, in some cases LPR may causean increase in overhead. For this reason, LPR may be disabled for troubleshooting purposes in any Polydevice. See individual product documentation or contact Poly support for details. For more innovations/lost-packet-recovery.html.4. DBA (Dynamic Bandwidth Allocation) – This is a Poly proprietary feature that automatically and graduallydecreases the video bit-rate (video quality) if packet loss is detected between two Poly endpoints (includingthe Polycom RMX MCU). The feature also automatically increases the video bit-rate, to configured call levels,if the packet loss condition later subsides. DBA may be disabled in any Poly device. See individual productdocumentation or contact Poly Support for details.5. LPR DBA PVEC – Between Poly products with all three features configured active (LPR recovers lostpackets, DBA prevents packet loss, PVEC hides problems from loss packets), audio or video networkpacket losses of up to 10% can still provide acceptable call quality and packet losses of 3% or less are usuallyunnoticeable on most networks (per independent testing, see f).Copyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved5

6. Employ Polycom DMA system Bandwidth Limitation Features – The Polycom DMA Callserver product (SIPregistrar proxy, H323 gatekeeper, virtual meeting room conference call manager) contains a feature thatmodels a network topology and controls the maximum bandwidth of calls over segments of that network.Using this feature, the network administrator can ensure no single call overloads and congests a networksegment. This feature can help prevent network congestion from occurring. See the Polycom DMAadministration guide regarding Site Topology and Bandwidth Limitation features or contact your Poly supportrepresentative for details.7. Polycom RMX MCU, AVC Mitigation for Content Issues Over a Lossy Network – Because of how sharedcontent video is encoded and transmitted during a conference, one bad/noisy endpoint in a call cannegatively impact the content video quality for other conference participants. The Polycom RMX MCU has 3configuration system flags that can be modified to help mitigate these effects:a. MAX INTRA REQUESTS PER INTERVAL CONTENT - The maximum number of refresh(intra) requests per 10-second intervals allowed for an endpoint. Beyond that number, contentsent by this participant is identified as “noisy,” and it’s refresh requests are suspended. Defaultsetting: 3b. CONTENT SPEAKER INTRA SUPPRESSION IN SECONDS - The interval, in seconds,between content refresh (intra) requests sent from the MCU to the content sender due to refreshrequests initiated by other conference participants. Additional refresh requests received withinthat interval are deferred to the next interval. Default setting: 5c.MAX INTRA SUPPRESSION DURATION IN SECONDS CONTENT - The duration inseconds to ignore the participant’s requests to refresh the Content display Default setting: 10If shared video content quality issues are experienced, an example set of settings could be:MAX INTRA REQUESTS PER INTERVAL CONTENT 2,CONTENT SPEAKER INTRA SUPPRESSION IN SECONDS 10,MAX INTRA SUPPRESSION DURATION IN SECONDS CONTENT 10.For details and/or help configuring these settings please contact your Poly support representative.8. Finally, if a video endpoint participant notices call quality issues during a call, the participant can mute theirvideo. Poly video endpoints provide a function to deactivate/mute a participant’s video stream. This actionallows for the maximum available bandwidth for audio and content video and may help to alleviatetemporary network issues.ConclusionBy following general networking best-practices and employing the use of specially designed Poly product features,overall call quality can be increased and a high-quality of audio/video services can be deployed. For help withnetworks or Poly products specifically, please contact a Poly support representative for assistance.Copyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved6

ReferencesPoly/Polycom Publications (2008) Lost Packet Recovery lostpacket-recovery.htmlWainhouse Research (2008, January) Polycom’s Lost Packet Recovery (LPR) Capability fGardiner, Berni (2008) QoS: What Is It? What Do We Need It? owledge-1.pdfGrech, Mat (2018, December 20) Acceptable Jitter & Latency for VoIP: Everything You Need to e-jitter-latency/Rivenes, Logan (2016, June 22) What is Acceptable Jitter blejitter/CISCO Whitepapers (2013, June 17) Best Practices in Core Network Capacity Planning Document ID:1506945661136136 routers/wan-automationengine/white paper c11-728551.htmlCISCO Whitepapers (2006, February 2) Understanding Jitter in Packet Voice Networks (Cisco IOS Platforms) DocumentID: 18902 oice-quality/18902-jitter-packet-voice.htmlLynch, Kathryn (2009, July 29) Is your network ready to handle oconferencing/Wilson, Marc (2019, March 20) Packet Loss – What is it, How to Diagnose and Fix It in your Networkhttps://www.pcwdld.com/packet-lossRobert, Jason & Patterson Carey (2012, April) Video Encoding Settings for H.246 odingh264/Also see the Poly/Polycom product document library for specific products: https://documents.polycom.com/It includes guides, papers, and manuals for all Poly/Polycom/Plantronics products including endpoints andinfrastructure.Copyright Plantronics, Inc. (Poly – formerly Plantronics and Polycom) 2019, All Rights Reserved7

must be paid to designing and configuring IP networks for real-time audio/video (A/V) traffic in order to deliver a good quality experience to all users. Following a set of “best practices” for network design and configuration, as well as in deployment/configuration of individual products, helps ensure