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  • 1. Agenda Service Provider VoIP Architectures Recognizing Voice Quality Issues Service Provider VoIP Classifying Voice Quality Attributes to Root Quality Issues Cause Address Voice Quality by Implementing Quality of Service (QoS) Faisal Chaudhry, CCIE Echo Talal Siddiqui, CCIE Miscellaneous VAD, Clocking, Fax, Modem, DTMF © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 1 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 2 Long Distance Voice High Level Overall Voice Architecture Architecture PSTN Gateway Data Network Softswitch PSTN SSP SLT SSP Softswitch (MGC) SLT SGW Softswitch SGW (MGC) Transit/Tandem M Voice M Gateways VoIP GW Gateway Remote sites over WAN Gateway IP Local VoCable VoIP GW Services /VoETTx Network Call Agent PSTN VoIP Bearer traffic Gateway Residential SIP Proxy PBX MG H.323 (IMS) Access Gatekeeper IP Phone GK SSP SLT Signalling : MGCP, H.323, SIP, SLT SSP SGW SGW SS7 (SIGTRAN) Voice CPE Business & Call Gateway Gateway Manager Residential © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential Access 3 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 4 Voice over Broadband (Cable, ETTx …) Cable Telephony Solution – Cable MSO Announcement • TWC went in Customer PoP Super PoP VM Server Softswitch service in March PremisesModem Subscriber A Cable Modem Cable uBR7246 Cisco BTS 2003 Softswitch Other Networks • Supplementary Cable Cable Modem Modem SS7 SS7 A Link Services (CFx, Router CW etc) for Multi- Internet uBR10012 Layer IP Network Analog Switch MGX Subscriber B (MGX) VSM Triple play services subscribers HFC DS-3, OC-3, Si V IMT Plant OC-12 Core PSTN MxU V Res. Villas • Service is $39.95 Subscriber C Cable Cable Modem Router Class 4/5 Modem Switch SS7 GW SBC per month and Session Border Controller includes AS5x00 PSTN unlimited calls in Cisco 3660 MF, FGD Interconnect US Cable Cable Modem Modem uBR7200VXR 911 AT Voice Mail/ Annc CALEA MF, FGD IP-IP CableLabs UM Server Server Server Interconnect OPS qualified Feature Servers Subscriber D http://www.linerunnermaine.com/index.cgi © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 5 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 6 1
  • 2. Customer Snapshot Telecom Italia: China Unicom FINANCIAL TIMES Wednesday Telecom Italia takes Europe's biggest step in Voice over IP with Cisco Systems and Italtel (…….) 100% of Telecom Italia's national Rome-Milan voice calls and 50% of its international European voice calls now run over a converged “VoIP/MPLS" 1E HRB 1E SY network WRMQ CC 12 E LZ HHHT 2E 1E 1E 1E “We chose VoIP because we could save two thirds of BJ 1E 2E DL YC 2E TJ SJZ 1E XN 1E TY 2E JN 2E 2E our transit operating expenses and give our XA Services Tibet 1E 1E WH 2E ZZ 1E 2E 10 E customers and shareholders a better service. By the Calling Card CD 1E CQ 1E 1E AH 2E NJ SH 2E end of 2003, we estimate that 80% of Telecom Italia's Post-paid 1E GY 1E 1E CX 2E 2E FZ HZ 2E transit voice traffic will travel over the Cisco Systems Pre-paid re-chargeable KM LZ GZ 2E XM 2E and Italtel Multiservice solution. We chose Cisco 8E Pre-paid non re-chargeable 12E 1E SZ Systems and Italtel because they had the most Single stage dialing: HN reliable solution. Cisco Systems has more National & International Transit experience in IP technology and VoIP than any Scalability 350 Cities other infrastructure company and the Directory gatekeeper technology partnership with Italtel provides additional expertise 1000s of E1 Capacity in the switching carrier environment" Extensibility Foundation for move into residential 700M Minutes per Month broadband telephony said Stefano Pileri, head of Telecom Italia's domestic network © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 7 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 8 Business Service: Managed Hosted IP Telephony Managed & Hosted IP Telephony configuration SP Hosted SP Hosted Access Managed Voice service for Enterprises by SP: Infrastructure Service (DSL, ETTx…) CPE Sites Central Central On-premises or Centralized Hosted Call Control LDAP Voicemail H.323 Interface to the PSTN/PLMN (Centralized or on- Softswitch Gatekeepers (or SIP) premises) SS7 Signaling GK Interface to Legacy equipment (such as PBX, GK Voicemail) Call Control MGCP Dial plan Management TDM Voice Billing Firewall Customer Premises Equipment e.g. Analog or IP Phones Voice Gateways LAN Switch Media Resources Need to cater for QoS, Resiliency, Security … (Conf. Bridge, Xcode …) © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 9 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 10 Managed IP Telephony with Call Control on customer premises Agenda SP Hosted Infrastructure Access CPE Sites Service Provider VoIP Architectures H.323 Voicemail Recognizing Voice Quality Issues Softswitch Gatekeepers SS7 (or SIP Proxy) (optional) Call Control Classifying Voice Quality Attributes to Root Signaling GK PSTN Cause GK MGCP Address Voice Quality by Implementing Firewall Access TDM Voice Router Quality of Service (QoS) Echo Call Control Voice Gateway LAN Switch Miscellaneous (optional) Voicemail VAD, Clocking, Fax, Modem, DTMF PSTN © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 11 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 12 2
  • 3. Categorizing and Defining the Symptoms Noise Noise Absolute Silence Conversation is still Intelligible; presence of Cause: Aggressive Voice Activity Detection (VAD) static, hum, crosstalk intermittent popping Voice distortion Clicking Cause: Clock Slips or Other Digital Errors Problem that affects the voice itself Echoed voice Crackling Garbled voice Cause: Poor Electrical Connection, Electrical Interference Volume distortion Crosstalk Cause: Signal Leakage Due to Wires Located in Close Proximity © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 13 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 14 Noise (Cont.) Echoed Voice Hissing Listener Echo Cause: VAD Cause: Long Echo Tail; Echo Canceller Is (ECAN) Not Activating Static Talker Echo Cause: Codec Mismatch; Enhanced by VAD Cause: Long Echo Tail; ECAN Is Not Activating Tunnel Voice Cause: Tight Echo with Some Loss © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 15 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 16 Garbled Voice Volume Distortion Fuzzy Voice Choppy Voice Cause: Too Much Gain on the Signal Cause: Consecutive Packets Lost or Excessively Delayed Disabling DSP Predictive Insertion Where Silence Is Muffled Voice Inserted Instead Cause: Overdriven Signal or Some Other Cause That Eliminates or Reduces Signal Level at Frequencies Inside the Key Range Synthetic Voice for Voice (Between 440 and 3500) Cause: Single Packet Loss or Delay Beyond the Bounds of the De-Jitter Buffer Playout Period Soft Voice Cause: Attenuated Signal Underwater Voice Tinny Voice Cause: A Common Cause of This Problem Is G729 IETF and Cause: Overdriven Signal that Eliminates or Reduces Signal Pre-IETF Codec Mismatch Level at Frequencies Outside the Key Range for Voice (Between 440Hz and 3500Hz) © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 17 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 18 3
  • 4. Agenda Problem and Root Cause Association Service Provider VoIP Architectures Synthetic Voice Packet Loss Recognizing Voice Quality Issues Fuzzy Voice Classifying Voice Quality Attributes to Root Absolute Silence Delay Cause Static and Hissing Echo Return Loss Address Voice Quality by Implementing Quality of Service (QoS) Muffled Voice Gain Adjustment Echo Soft Voice Jitter Miscellaneous VAD, Clocking, Fax, Modem, DTMF Choppy Voice VAD Listener Echo © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 19 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 20 Classifying Voice Quality Attributes to Root Cause Basic Guidelines for of Voice over IP Transmit voice traffic the fastest way possible Quality of Network VAD, Codecs Delay is bad (worsens echo, awkward conversations, etc.) Service Transmission • Absolute silence Minimize as many sources of delay as possible Loss Plan • Clipping Goal: keep delay to less than 150ms • Static and hissing • Loss • Gain adjustment • Underwater voice Transmit VOIP packets as a steady, smooth stream • Jitter • ERL Any delay should be consistent • Delay Inconsistent delay is called “Jitter” • Synthetic voice • Talker echo Synchronization, Compensating for Jitter creates additional delay • Robotic voice • Listener echo Cabling • Choppy voice • Tunnel voice • Crackling Drop any packets received out of order • Periods of silence • Fuzzy voice • Clicking • Muffled voice Voice does not tolerate delays…it’s better to drop the packet • Crosstalk • Tinny voice CODEC logic can compensate for some dropped packets Above all…it’s gotta sound good (subjective) © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 21 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 22 Voice QoS Requirements Agenda End-to-End Latency Hello? Hello? Service Provider VoIP Architectures Avoid the Recognizing Voice Quality Issues “Human Ethernet” Classifying Voice Quality Attributes to Root Cause CB Zone Address Voice Quality by Implementing Satellite Quality Quality of Service (QoS) High Quality Fax Relay, Broadcast Echo 0 100 200 300 400 500 600 700 800 Time (msec) Miscellaneous Delay Target VAD, Clocking, Fax, Modem, DTMF ITU’s G.114 Recommendation: ≤ 150msec One-Way Delay ITU’ One- © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 23 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 24 4
  • 5. Elements That Affect End-to-End Delay Calculate Delay Budget Coder Queuing Serialization De-Jitter Buffer Propagation Delay Delay 25 ms Delay 6 ms Delay 3 ms 50 ms 32 ms PSTN (Private Line Network) IP WAN Los Angeles Munich Fixed Delay Variable Delay Campus Branch Office Coder Delay G.729 (5 msec look ahead) 5 msec Propagation CODEC Queuing Serialization Jitter Buffer Coder Delay G.729 (10 msec per frame) 20 msec and Network Packetization Delay—Included in Coder Delay Fixed Fixed Variable Variable + (6.3 µs/Km) Queuing Delay 64 kbps Trunk ~6 msec (Can Be Reduced (Can Be Reduced 20–50 ms (Sampling rate) Using LLQ) Using LFI) Network Delay Serialization Delay 64 kbps Trunk 3 msec (Variable) Propagation Delay (Private Lines) 32 msec Network Delay (e.g. Public Frame Relay Svc) N/A—Private Line N/A—Private Line Goal: End-to-End Delay (Should Be < 150 ms) 150mx As per ITU G.114 Recommendations De-Jitter Buffer ~50 msec Delay Total ~116 msec © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 25 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 26 Voice QoS Requirements Voice QoS Requirements Factors which Degrade Voice Quality Factors which Degrade Voice Quality Packet Loss Packet Loss Voice Voice Voice Voice Voice Voice Voice Voice 4 3 2 1 4 3 2 1 Voice Voice Voice Voice Voice Voice Voice Voice Voice 4 3 2 1 4 3 2 1 3 Voice Voice 3 Reconstructed Voice Sample 3 Voice Current Cisco gateway DSP CODEC algorithms can correct Reconstructed Voice Sample for 30 msec of lost voice by using predictor algorithm 3 Generally 1 G.729A voice packet contains 20 msec of voice Lost packets induce “clipping” and temporarily expand the jitter buffer, which increases end-to-end latency One lost Voice over IP packet causes a MODEM retrain; Two drops can cause a call disconnect Causes of packet loss: Network quality, network congestion and delay variation © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 27 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 28 Voice QoS Requirements Voice QoS Requirements Factors which Degrade Voice Quality De-Jitter Buffer Operation Variable Delay – Jitter Normal Operating Sender Receiver Queue Depth Voice Frames Voice Frames From Network V V V VV VV VV VV VV V V V V to DSP decode PBX Network PBX Variable Fixed Playout Arrival Rate Over Flow: Under Flow: Rate = Codec Frame Queue Fills If Quiescent Point: Queue Empties If = Codec Frame Rate +/- ∆ Voice Frame Specified in mSec Voice Frame Rate Arrive Too Fast Arrive too Slow Sender Transmits t When voice call starts, the de-jitter buffer fills up to the quiescent point As voice frames arrive too fast, the queue fills Sink Receives As voice frames arrive too slowly, the queue empties D1 D2 = D1 D 3 = D2 t Depth of queue varies with network operation Over-/-under flow will cause gaps in speech and underwater voice © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 29 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 30 5
  • 6. Voice QoS Requirements Voice QoS Requirements De-Jitter Buffer Operation Summary for Voice Variable Delay—Jitter Buffer Under/Over Runs Latency ≤ 150 ms Cisco GW DSPs Uses an Adaptive Jitter Buffer Voice which has10 msec of “Extra” Buffer Jitter ≤ 30 ms One-Way Requirements Packet Dropped If Instantaneous Jitter Is > 10 msec Loss ≤ 1% 50ms of possible Jitter Buffer 17–106 kbps guaranteed priority bandwidth per call Bandwidth for Voice-Control traffic per call Smooth Network CODEC SCCP: 150 bps + Layer 2 overhead Benign CAC must be enabled (Integrated Drop sensitive Services) Delay sensitive <=10ms UDP priority Calculated Jitter Buffer Based on Variable Network Delay in msec (packet RTP Timestamp) © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 31 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 32 Enabling QoS QoS Operation Differentiated & Integrated Services Standards Differentiated Services QoS Approach Classification: DSCP settings & attribute within a packet to differentiate (L3) RFC 3246: If mark packet with EF then PHB is to move this packet to front of queue during congestion. Mark at the edge! Trust Boundary: Define and Enforce a Trust Boundary at the Network Edge QUEUEING AND Scheduling: Assign Packets to One of Multiple Queues (based on CLASSIFICATION AND MARKING (SELECTIVE) DROPPING Post-Queuing Operations Classification) for Expedited Treatment Through the Network Provisioning: Accurately Calculate required Bandwidth for All Applications Plus Element Overhead Out of band signaling for BW: RSVP Central Site PSTN SRST Router IP WAN Campus Branch Office © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 33 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 34 QoS Operation Enabling QoS Differentiated Services QoS Approach DiffServ QoS Approach in SP Summary Queuing Classification Classification Classification Classification Policing Marking Shaping Policing Scheduling Marking Shaping Marking Scheduling Scheduling Core Router Edge Edge Router Router 1st Operation in Diff Discard Misbehaving Mark closest Congestion Post Queuing Serv is to classify: Traffic: to to the source: Management & Operations: - All traffic looks Maintain Network Mark Traffic Avoidance: Fragmentation Classification same until Integrity According to Congestion due to & Interleaving. Classification Core classification & behavior and speed mismatch Shaping: Policing marking Business (store & may Control Bursts Marking Router Marking - Identify and Split Policies reorder). and Conform Shaping Prioritize, Protect Traffic Scheduling Traffic into Scheduling Different Classes and Isolate Traffic Classification based on Markings Scheduling © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 35 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 36 6
  • 7. QoS is Needed to Minimize Packet Loss, Delay and Delay Variation Queuing on the IP Phone (Edge device) Where QoS Is Needed Mini-switch inside the IP Phone re-tags the packets from Central Site Remote Branch access port In your network, what edge device is used? L2 or L3? Type of L2? Si WAN IP Phone Switch ASIC Si Untrusted: Phone ASIC Will Re- Write CoS = 0 COS = 5 QoS—Campus Access QoS—Campus Distribution QoS—WAN QoS—Branch • Define trust boundary • Layer 3 Policing • Low-latency Queuing • Classification and Trust COS = 5 • Classification on IP • Multiple Queues on All • Link Fragmentation and Boundaries on IP Phone , Phone and Access Ports; Priority Queuing Interleave Access Layer Switch and Switch for VoIP Router • Bandwidth Provisioning • Multiple Queues on IP • WRED Within Data • Multiple Queues on IP • Admission Control Phone and All Access Phone and Access Ports Queue for Congestion Ports COS = 0 COS = 7 Management © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 37 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 38 Queuing on the IP Phone (Edge device) Queue 0 (PQ) Queue ? TX Voice RTP Packets RX Queue 1 Queue 2 QoS in Campus Queue 3 • Priority Queue (PQ) is for VoIP RTP traffic (CoS = 5) • Round-Robin Scheduling with a PQ Timer © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 39 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 40 Is QoS Needed in the Campus? Campus QoS Considerations Typical Campus Oversubscription Ratios Core Si Si “Just throw more bandwidth at Typical 4:1 it. That will solve the Distribution Oversubscription problem!” Si Si Typical 20:1 Oversubscription Access Maybe, Maybe Not; Campus Congestion Is a Buffer Management Issue © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 41 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 42 7
  • 8. Access, Distribution and Core Queuing Tx Buffer Congestion Area’s Where QoS Maybe a Concern Ethernet Switch Data Flows “Hog” Data Tx Buffer Si Si TX TX TX TX TX TX RX Core Access Distribution Data RX To Core Data TX Output buffers can reach 100% in campus networks RX When an output buffer congests, dropped packets occur at the ingress interfaces Voice QoS required when there is a possibility of congestion in buffers RX Additional Flows, Including Voice, Multiple queues are the best way to “guarantee” voice quality Can Not Get Access to Tx Buffer © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 43 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 44 Guaranteed Voice Requires Multiple Queues Campus QoS—Access Layer Control and Management Plane Traffic Ethernet Switch Queue Assignment Core Data Based on CoS/ToS Si RX Classification Call Signaling is control LAN Switches plane for VoIP traffic Data Distribution RX All VoIP Control Plane Si Si To Core Traffic should be Classified LAN Switch Data TX as DSCP=AF31 in the VoIP Access RX Gateway or from the LAN 4/2 4/3 4/4 Switches e.g. SIP: UDP 5060 Voice Skinny Control: TCP 2000-2002 RX Voice Put Into MGCP Control: UDP 2427 Call Control “Delay and Drop” H.323 Control: TCP 1720 (e.g. Softswitch) MGCP Gateway Sensitive Queue TCP 11000- 11999 H.323 Gateway © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 45 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 46 WAN Edge QoS Design Considerations QoS Requirements of WAN Aggregators Campus Distribution/Core Queuing/Dropping/ Switches Shaping/Link-Efficiency Policies for Campus-to-Branch Traffic WAN Aggregator (WAG) QoS in WAN WAN LAN Edges WAN Edges © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 47 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 48 8
  • 9. Scheduling Tools Scheduling Tools Queuing Algorithms Congestion Avoidance Algorithms TAIL DROP WRED Queue Voice 1 1 3 3 1 0 1 2 1 2 0 2 0 3 2 1 3 Video 2 2 0 3 3 3 Data 0 3 Congestion can occur at any point in the network where there Queueing algorithms manage the front of the queue are speed mismatches i.e. which packets get transmitted first So no reordering if no congestion! Congestion avoidance algorithms, like Weighted-Random Scheduling tools define how packet exits a device Early-Detect (WRED), manage the tail of the queue Only activated when congestion occurs i.e. which packets get dropped first when queuing buffers fill WRED can operate in a DiffServ compliant mode which will Cisco IOS-based software queuing in WAN routers drop packets according to their DSCP markings Low-Latency Queuing (LLQ) used for highest-priority traffic (voice/video) Class-Based Weighted-Fair Queuing (CBWFQ) used for guaranteeing bandwidth to data applications Cisco Catalyst® switches use hardware queuing © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 49 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 50 Link-Specific Tools Link-Specific Tools Link-Fragmentation and Interleaving Link-Fragmentation and Interleaving Fragmentation and Interleave Not Needed on Links Greater Than 768 kbps Serialization Voice Data Can Cause Excessive Delay Before Real-Time MTU Elastic Traffic MTU Data Data Data Voice Data 214 ms Serialization Delay for 1500 Byte Frame at 56 kbps With Fragmentation and Interleaving Serialization Delay Is Minimized Serialization delay is the finite amount of time required After Elastic MTU Elastic MTU Real-Time MTU Elastic MTU to put frames on a wire For links ≤ 768 kbps serialization delay is a major factor Mechanisms: affecting latency and jitter Pt to Pt Links: MLPPP For such slow links, large data packets need to be Frame Relay: FRF.12 fragmented and interleaved with smaller, more urgent ATM: MLPPP over ATM voice packets ATM/Frame-Relay SIW: MLPPP over ATM and FR © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 51 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 52 Scheduling Tools Enabling QoS in the WAN LLQ and Class_Based-WFQ Algorithm Traffic Shaping Line Without Traffic Shaping Rate Central With Traffic Shaping Site PQ Packets Do Not Go Through R Fragmentation: on Low Link Speeds, You Cannot Put Large (i.e., Video) Packets in the PQ with Voice Packets Low Latency Queuing Link Fragmentation E1 Traffic Shaping Limits the Transmit Police and Interleave Rate to a Value (R) Lower than Line Rate Voice Frame Relay PQ Video Why Is It Needed or ATM TX Interleave Ring 1 1. Line Speed Mismatch 64 Signaling kbps E1 E1 E1 Packets Packets 2. Remote to Central Site 2 In Critical Data CBWFQ Fragment Out Over-Subscription 3. To Prevent Bursting Above 3 CIR = 64 kbps Remote Sites WFQ Best Effort Committed Rate (CIR) Layer 3 Queuing Subsystem Layer 2 Queuing Subsystem 1 2 3 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 53 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 54 9
  • 10. Sources of Trouble for VoIP Provisioning Voice Is Not Free—Especially on Low Speed Links—Engineer the Network for Data, Voice, and Video Voice Video Voice/Video Data Routing etc Call Admission Control Control (CAC) LLQ = 33% Sum of Traffic = 75% Link Capacity Link Capacity = (Min BW for Voice + Min BW for Video + Min BW for Data) / 0.75 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 55 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 56 Introduction Why Is Call Admission Control (CAC) Needed? Topology-Unaware CAC Circuit-Switched Packet-Switched Vendor specific Headquarters Networks Networks implementations Call IP WAN Link’s LLQ Is static configuration in call Processing Provisioned for Two processing agent Agent PSTN IP WAN Calls (Equivalent to Two “Virtual” Trunks) Define the logical “site” Max. 5 Calls Max. Physical IP WAN to match physical Max. 8 Calls Trunks Link No Physical Limitation site 3 Calls on IP Links; Third Call Rejected Third Call Can Go Through, but Voice Configure a max Call Router/ Control Quality of All Calls number ... PBX STOP Gateway Degrades Call Admission of calls or max amount Control Blocks Third Call of bandwidth “site” 1 “site” 2 “site” N Branch 1 Branch 2 Branch N © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 57 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 58 Topology-Unaware CAC Topology-Unaware CAC “Real” Network Topology Aspects Spoke Site 1 Spoke Spoke Spoke Spoke Spoke Spoke Dual-Homed Branch Sites Hub Site Spoke Spoke Site 2 Site Y Redundant Hub Sites 2nd Tier 2nd Tier or SP-Provided MPLS IP WAN Dual WAN Links Hub Hub Spoke Site 1 Spoke Site 2 ... Spoke Site N Spoke Spoke Fully Meshed Core 1st Tier Hub Site X Site N Core Core Multiple Paths 1st Tier Limited to: Hub 1st Tier Hub Hub-to-Hub Links Simple hub-and-spoke topologies 2nd Tier Hub 2nd Tier 2nd Tier Hub 2nd Tier Hub Hub Simple MPLS-based topologies Multi-Tier 3rd Tier Hub 3rd Tier Hub 3rd Tier 3rd Tier Architectures Hub Hub Spoke Spoke Spoke Spoke Spoke Spoke Spoke Spoke Spoke Spoke Spoke Spoke © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 59 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 60 10
  • 11. Topology-Aware CAC Call Admission Control Principles Integrated Services Summary Based on communication Topology-Unaware CAC Topology-Aware CAC between call processing Based on static configuration Based on real network within call processing agent resources agents and the network on available resources Does not react to network Requires a signaling Call topology changes protocol (RSVP) Can be applied to any Processing (e.g. link failures) Agent Reacts to network WAN network topology RSV P Limited to simple topologies topology changes Dynamically adjusts to Limited to a single call Call IP WAN processing agent No topology limitation topology changes Processing (Any Topology) Agent Examples: Cisco Can be used by different call Requires a signaling CallManager® “static” processing agents protocol locations, Cisco IOS® gatekeeper zones Examples: Cisco RSVP (Resource CallManager RSVP-enabled locations, Cisco H.323 & SIP ReSerVation Protocol) is Voice gateways (AS5x00, the first industry standard 3800 etc), Cisco multiservice for QoS signaling IP-IP gateway © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 61 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 62 RSVP Principles RSVP Principles IETF standard RSVP-Unaware Routers RFC2205, RFC2207, RFC2208, RFC2209, RFC2210, and others If Bandwidth on Any Link Ignore and Forward All If There Is Sufficient Throughout the Network Is RSVP Messages Bandwidth Throughout Not Sufficient, the Topology-aware CAC signaling protocol Reservation Fails the Network, the Reservation Succeeds Works with any WAN topology 80 24 72 RSVP Bandwidth Pool Uses existing routing protocols Provisioned on Each 80 Router Interface with: Device 72 24 2 Dynamically adjusts to link failures and topology changes ip rsvp bandwidth ... 48 0 24 64 30 24 Device Unidirectional reservations 96 4 Device 24 30 6 Reservations are receiver-initiated 1 40 64 24 64 48 48 48 56 40 64 Maintains “soft state” in RSVP-enabled routers RSVP Signaling Uses Device 24 0 48 Same IP Route as the 3 Operates transparently across non-RSVP routers Data Stream That Legend: Needs Reservation 56 = kbps Allows for partial or gradual deployment across network Remaining in RSVP Bandwidth Pool © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 63 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 64 End devices CAC in H.323 & SIP End devices CAC in H.323 & SIP Gateway Resources: PSTN Side CAC H.323 Gateway Resources: Network Side CAC GK GK PSTN Cisco ASAP PSTN PSTN Cisco ASAP PSTN RAI • Calls not delivered/accepted if • H.323: Gateway informs Gatekeeper if 100% DSPs High any set of required resources are D resource threshold exceeded using DSPs resources DS0, DSP D Low S S all in use 0 HDLC RAI1 0 framers • Resources monitored – GK selects GW based on RAI status 0% – DS0s, DSPs, HDLC framers • ICPIC2 or Delay & Loss thresholds • Hairpin, Play announcement or 1 RAI - Resource Availability Indicator – Release with ‘Resource Unavailable’ 2 ICPIF - Calculated Planning Impairment Factor (ITU G.113) © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 65 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 66 11
  • 12. General Guidelines for QoS in the WAN Agenda Use Low Latency Queuing (LLQ) anytime VoIP over the WAN Service Provider VoIP Architectures is involved Traffic shaping is a requirement for frame-relay/ATM Recognizing Voice Quality Issues environments Classifying Voice Quality Attributes to Root Use Fragmentation & Interleaving (LFI) techniques for all links Cause below 768Kbps Don’t use LFI for any video over IP applications Address Voice Quality by Implementing Properly provision the WAN bandwidth Quality of Service (QoS) Call admission control is a requirement where VoIP calls can over-subscribe the provisioned BW Echo Use cRTP carefully Miscellaneous VAD, Clocking, Fax, Modem, DTMF © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 67 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 68 Talker Echo Listener Echo Talker Echo (Most Common) Listener Echo (Less Common) Talker echo occurs when a talker's speech energy, transmitted down the Listener echo occurs at the far-end by circulating voice energy; primary signal path, is coupled into the receive path from the far end; the again, listener echo is generally caused by the 2W/4W ‘hybrid’ talker then hears his/her own voice, delayed by the total echo path delay transformers; caused by the “echo being echoed”; the talker’s time; if the ‘echoed’ signal has sufficient amplitude and delay, the result voice is echoed by the far end hybrid and when the echo comes back to the listener, the hybrid on the listener’s side echoes the can be annoying to the customer and interfere with the normal speech echo back towards the listener; the effect is the person listening process; talker echo is usually a direct result of the 2-wire to 4-wire hears the talker and an echo of the talker conversion that takes place through ‘hybrid’ transformers Delayed Echo of John’s Voice John’s Voice John’s Voice John Echo of John’s Voice Jane John Jane San Jose New York San Jose New York © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 69 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 70 Sources of Echo What Makes Echo a Problem? Hybrid Echo Acoustic For Echo to Be a Problem, all of the Four-wire Trunk Echo Following Conditions must Exist: Two-wire Two-wire Subscriber Subscriber loop Echo loop An analog leakage path between analog Tx and HYBRID HYBRID Earpiece Rx paths Echo S1 S2 Sufficient delay in echo return for echo to be perceived as annoying Echo Any Mismatch of Impedance vs. ZL Sufficient echo amplitude to be perceived as Tx Will Cause the Tx Signal to Appear annoying ZL 4-Wire Trunk Rs Rs on Rx, This Is Microphone Hybrid Echo + Z? Rx - Xfmr 2-Wire Subscriber Loop Schematic of a Hybrid Circuit © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 71 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 72 12
  • 13. Ecan Terminology Ecan Terminology Tail Circuit Tail Circuit Rin Rout Rin Rout ^ ^ H(t) H(t) IP Network ERL IP Network ERL ^ y(t) ^ y(t) - Sin Sout - Sin ERLE Echo Return Loss Enhancement (ERLE) refers to the additional echo loss obtained Echo Return Loss (ERL) is the echo level loss provided by the tail circuit. through the operation of the echo canceller. An echo canceller is not a perfect device, That is, if an speech signal entering the tail circuit from the network at a level of X dBm0 and the best it can do is attenuate the level of the returning echo. ERLE is a measure of (i.e. Rout – above), the level of the echo coming back from the tail circuit into the Sin this echo attenuation through the echo canceller. terminal of the echo canceller is (Rout – ERL) dBm0 i.e. It is the difference is level (in dB) between the signal arriving from the tail circuit at the Sin terminal of the echo canceller and the level of the signal leaving the echo canceller (and ERL = Rout - Sin entering the network) at the Sout terminal. ERLE = Sin - Sout © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 73 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 74 Ecan Terminology Eliminating Echo Tail Circuit Tail Circuit Rin Rout Rin Rout Output Attenuation E ^ H(t) C IP Network ACOM ERL IP Network ACOM A N ERL ^ y(t) Input Gain Sout - Sin Sout - Sin ERLE ERLE Give the echo canceller enough information to ACOM (aka Acombined) is simply the total echo return loss seen across the distinguish between echo and normal conversation; Rin and Sout terminals of the echo canceller. It is the echo return loss the only parameters you have control over are: Input level (input gain) seen by the network. Output level (output attenuation) ACOM = ERL + ERLE Echo canceller coverage © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 75 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 76 On-Net to On-Net Call Rules of Thumb Echo on one end is typically generated at other Caller A Caller B end 4dB 4dB RLR=–6dB CG4501 IP Network CG4501 SLR=8dB Bits don’t leak 2w–4w Hybrid 2w–4w Hybrid Echo is not introduced on digital links V Si Si V SLR=8dB ERL=26dB RLR=–6dB Introduced by 2 to 4 wire conversion in hybrid and 4dB 4dB impedance mismatch or via acoustic feedback ERL=26dB One Way Delay ERL must be 6dB for ECANs to engage OLR(A) = 8dB + 4dB + 4dB - 6dB = 10dB Be careful setting echo-cancel coverage; Longer coverage yields longer convergence time; TELR (A) = 8dB + 4dB + 4dB + 26dB + 4dB + 4dB - 6dB = 44dB Configure the coverage so that it is long enough to cover the worst-case for your environment, but no Talker Echo Loudness Rating (TELR) higher Overall Loudness Rating (OLR) © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 77 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 78 13
  • 14. Agenda Service Provider VoIP Architectures Recognizing Voice Quality Issues Classifying Voice Quality Attributes to Root Cause Voice Activation Detection, Cabling, Clocking Address Voice Quality by Implementing Quality of Service (QoS) Echo Miscellaneous VAD, Clocking, Fax, Modem, DTMF © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 79 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 80 Comfort Noise and VAD Troubleshooting Hissing, Static, Clipping Cabling 3640-6(config)#voice vad-time ? tH <250-65536> milliseconds 3640-6(config)#voice vad-time 750 Analog Gateways Cabling is the number Power Device A one cause of issues in analog connections PBX Cabling testing must be a part of IP Cloud PSTN implementation plan T0 T1 T2 Phone NTLP is a good source for verifying Fax cabling issues Power Background Noise Device B Attenuation Comfort Noise Modem Attenuation T1 T2 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 81 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 82 Synchronization Troubleshooting Clicking Sound Primary (1) LEC TDM Secondary (2) Internal (3) Slip IXC Fax, Modem & DTMF in VoIP networks Branch2-3745# show controller t1 3/0 T1 3/0 is up. Applique type is Channelized T1 Cablelength is long gain36 0db Transmitter is sending remote alarm. Receiver has loss of frame. alarm-trigger is not set Version info Firmware: 20040202, FPGA: 11 Framing is ESF, Line Code is B8ZS, Clock Source is Line. Current port master clock:local osc on this network module Data in current interval (103 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 398 Slip Sects, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 103 Unavail Secs © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 83 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 84 14
  • 15. Bearer Features DTMF Relay – Signaling dependant Bearer: RFC 2833 (MGCP, SIP, H.323) Signal: H.245 Alphanumeric, H.245 Signal, MGCP Notify, SIP Notify Fax and Modem Fax over IP in VoIP networks Fax Pass Through (Proprietary) T.38 Fax Relay Modem Support Modem Pass Through (Proprietary) Modem Relay © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 85 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 86 Virtual Fax/Modem machine Transport across the cloud Passthru/Upspeed Packet Voice Network Local Local ITU-T T.38 Relay PSTN PSTN Modem Relay Local Local PSTN PSTN T.30 T.30 T.30 T.30 Something mystical happens here © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 87 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 88 Passthru G.711 codec Upspeed to G.711 codec G.711 from PSTN G.711 from PSTN G.711 codec in RTP Stream G.729 codec in RTP Stream G.711 G.711 to PSTN G.711 to PSTN Local Local PSTN PSTN Local Local PSTN PSTN G.729 Voice stream Tone detected, Upspeed codec to G.711 No change in PCM stream received from PSTN, no T.30 de-modulation © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 89 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 90 15
  • 16. Fax relay with ITU-T T.38 T.38 Switchover H.323 G.711 from PSTN VoIP call established T.30 T.30 Fax data in UDP packet CED tone passed across hdlc requestMode preamble G.711 to PSTN requestMode Ack Local Local CloseLogicalChannel (RTP) PSTN PSTN Local CLC Ack Local PSTN OpenLogicalChannel (T.38) PSTN OLC Ack CloseLogicalChannel (RTP) G.729 Voice stream T.30 Demodulated at DSP, encapsulated CLC Ack in IP UDP packet OpenLogicalChannel (T.38) T.30 OLC Ack T.30 T.38 defines UDP header format and Fax Relay Established switchover method © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 91 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 92 T.38 Switchover SIP VoIP call established T.30 T.30 CED tone passed across hdlc Invite (T.38 in SDP) preamble DTMF 200 OK Local PSTN ACK in VoIP networks Local Fax Relay Established PSTN © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 93 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 94 DTMF in VoIP DTMF Options in VoIP networks SIP KPML C4/C5 SIP, H.323, MGCP RFC2833, RTP Caller Call Control performs Signaling Path Bearer Path INVITE 1 digit-by-digit collection SIP Phone 100 Trying and routes the call as SUBSCRIBE using KPML 200 OK soon as enough digits • Out of Band: NOTIFY (0) are collected 200 OK H.323: H.245-Alphanumeric, H.245-Signal KMPL is Keypad Markup UNSUBSCRIBE 1 SIP: SIP Notify, KPML (Draft) 200 OK Language draft-ietf- MGCP: MGCP Notify sipping-kpml • Bearer path: RFC 2833 describes how to carry DTMF, other tone signals and telephony events in RTP packets. © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 95 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 96 16
  • 17. References Echo http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm #91601 Voice Quality Degradation Symptoms http://www.cisco.com/warp/public/788/voice-qos/symptoms.html#clip Q and A Quality of Service http://www.cisco.com/warp/public/732/Tech/qos/ IP SLA http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configur ation_guide_book09186a00802b2a6c.html VoIP Troubleshooting Using “show call active voice” http://www.cisco.com/warp/public/788/voip/show_call_act_voice.html © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 97 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 98 © 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 99 17