Copyright © 2004 OPNET Technologies, Inc. Confidential, not for distribution to third parties. Introduction to VoIP Technology Tutorials Session 1819

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP  What is VoIP?  Why is it used?  How is it used?  Applications and architectures  How does VoIP work?  Protocols  What do VoIP calls sound like?  QoS  How can I make sure that VoIP deployments will work properly?  Modeling and simulation Agenda

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP What is VoIP?  Carrying voice conservations over Internet protocol packet networks  Private  Public  There are other flavors of packetized voice:  VoATM  VoFR IP Network

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Why Use VoIP?  Cost savings  Integrated data and voice networks  Device interoperability using standards-based protocols  Flexibility in deriving new services

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Traditional Voice Versus VoIP  A traditional T1 can carry 24 telephone calls simultaneously  With VoIP, a T1 can carry 64 calls simultaneously! G.729 8kbps compression, 20 msec frame size = 24 kbps 1544 / 24 = 64 calls per T1 T1 = 1544 kbps, DS0 = 64 kbps, 1544 / 64 = 24 DS0 per T1

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Devices  IP Telephones  A telephone that directly connects to an IP network  Gateways  Provide bulk conversion of connections between signaling domains:  PSTN connections to VoIP connections  One VoIP signaling domain to another  Servers  Handle registration, authentication, telephone number to IP address conversion, bandwidth management, etc.

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP How is VoIP Used?  Applications and architectures  Consumer  Campus  Enterprise  Service Provider

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Consumer: IP-to-IP  Uses PC software to make calls over public and private internets  Free!!  But, no quality of service guarantees  Examples:  Microsoft NetMeeting TM  Skype TM  Hybrids  PC2Phone TM The Internet PC Modem

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Campus Applications – IP PBX  Connecting office telephones to PBX with VoIP links  Vendors  Cisco  Nortel PSTN LAN Switches IP Telephones Call Manager & IP-to-PSTN Gateway

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Enterprise Applications – Toll Bypass  Connecting enterprise PBXs with VoIP links to avoid paying for long distance charges  Vendors:  Nortel  NEC  Avaya  Toshiba  Ericsson  Cisco Private Data Network or VPN Public Switched Telephone Network (PSTN) PBX Router

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Service Provider Applications – Local Access  Using broadband access to provide local and long distance telephone service  Example Services:  Vonage  ATT CallVantage TM  Packet8  Broadvox  Time Warner Cable PC Broadband Modem Splitter Ordinary Telephone Broadband Service Provider PSTN Access Provider ISP

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Service Provider Applications – Trunking  Carrying voice traffic between switches over long haul network  Allows for consolidation with data networks  Example Hardware:  Nortel  Sonus *LATA = local access and transport area LATA #2 Private Data Network LATA # 1 LATA #3 LATA #4 LATA #5 LATA #6 PSTN Gateway

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP  What is VoIP?  Why is it used?  How is it used?  Applications and architectures  How does VoIP work?  Protocols  What do VoIP calls sound like?  QoS  How can I make sure that VoIP deployments will work properly?  Modeling and simulation Agenda

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP IP Phone How Does VoIP Work? Gatekeeper LAN 1. Caller dials Gatekeeper responds with the IP address of the called party 4. Caller sends a call setup message to the called party 5. Called party accepts the call by picking up the telephone receiver. An “accept” message is sent back to the caller. 6. Voice packets flow between IP telephones 2. Gatekeeper performs authentication, call admission control, and address translation

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP IP Phone How Does VoIP Work? (IP-to-PSTN) Gatekeeper LAN 1. Caller dials Gatekeeper responds with the IP address of the gateway 4. Caller sends a call setup message to the gateway 2. Gatekeeper performs authentication, call admission control, and address translation PSTN Ordinary Telephone IP-to-PSTN Gateway 7.Gateway converts the PSTN accept message into VoIP accept message and sends it back to the caller. 5. Gateway converts the VoIP signaling message to PSTN signaling message 6. Called party accepts the call by picking up the telephone receiver. An “accept” message is sent back to the gateway. 8. Voice packets flow between IP telephone and gateway. Gateway converts between packet data and timeslot data.

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Protocol Soup  Signaling  H.323, SIP, MGCP, H.248, SCCP, etc.  CODECs  G.711, G.723, G.729  Transport  RTP, RTCP, CRTP, ECRTP, UDP, IP  Other  RSVP, DiffServ, IntServ, MPLS, DNS, COPS (policy), Radius & Diameter (authentication)

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Signaling Protocols  H.323  Distributed architecture  Used for video conferencing, but also VoIP  SIP – Session Initialization Protocol  Distributed architecture  IETF RFC 2543  MGCP – Media Gateway Control Protocol  Centralized architecture  IETF RFC 2705  H.248  Centralized architecture  Extends MGCP  Collaboration between ITU and IETF  Also known as RFC 2885, Megaco  SCCP – Skinny Client Control Protocol  Cisco proprietary  For use with Cisco CallManager H.323 SIP Distributed MGCP H.248 Centralized

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP H.323 Details  ITU umbrella standard for packet-based multimedia communication systems  Audio CODECs  Video CODECs  H.255 registration, admission, and status (RAS)  H.225 call signaling  H.245 control signaling  Real-time transport protocol (RTP)  Real-time control protocol (RTCP)  Early standard  Complex Transport Protocols RTP RTCPH.255 RAS H.225 Call Signaling H.245 Control Signaling T.120 Data G.711 G.729 G.723 H.261 H.263 Call Manager ApplicationAudioVideo

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP H.323 Components  Terminals  Hardware or software running H.323 protocols  Gateway  Connects different networks  H.323-to-PSTN  H.323-to-{other VoIP signaling protocol}  Gatekeeper (optional)  Address translation  Admission control  Bandwidth control  Zone control

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP H.323 Call Setup Gatekeeper LAN 1. Caller dials Gatekeeper responds with the IP address of the called party 4. Caller sends a call setup message to the called party 6. Voice packets flow between IP telephones 2. Gatekeeper performs authentication, call admission control, and address translation IP Phone IP Phone ( ) 5. Called party accepts the call by picking up the telephone receiver. An “accept” message is sent back to the caller. H.323 Messages

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Session Initiation Protocol (SIP) Details  Recent standard  Simpler then H.323  Also used for video conferencing, network gaming, instant messaging  Similar to HTTP, textual coding  Uses URLs for addressing:     tel:  DTMFs carried in signaling message

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP SIP Call Setup SIP Proxy IP Network INVITE From: INVITE From: ACK Voice packets flow between IP telephones Proxy for sip.com gets location information for called party. IP Phone IP Phone OK 200 From: OK 200 From:

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP MGCP/H.248/Megaco Details  Based on master/slave principal  More palatable to telco's  Easier to rollout new feature since only the servers need to be updated, not the individual telephones Media Gateway Controller Signaling Gateway PSTN H.248 Messages Call Control (SIP, H.323, etc.) SS7, etc. Trunks Media Gateway Controller IP Phone Media Gateway

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Break!!! Break!

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP CODECS  Voice codecs create blocks of data at fixed intervals  Usually 10 ms  Each block contains a fixed number of bytes depending on the coding scheme used  bytes/block  Codecs can typically be parameterized to put a given number of voice data bytes into a single IP packet  10, 20, 30, …, 240 bytes  Bandwidth saving techniques  Silence suppression  Compression  Tradeoffs  Small packets = less delay, but more layer 2/3 overhead  Large packets = more delay, less layer 2/3 overhead

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Typical CODEC Behavior

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP CODEC Characteristics

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Real-time Transport Protocol (RTP)  Media content type  Talk spurts  Sender identification  Synchronization  Loss detection  Segmentation and reassembly  Security (encryption) VPXPayloadMSequence Number Timestamp Synchronization Source Identifier (SSRC) Payload

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP RTP Control Protocol (RTCP)  Used for monitoring the quality of a session  Transferring that information to all of the participants in the session  Provides minimal session control  Sent on different port number from RTP  Messages:  Sender Reports: Information about sent data, synchronization timestamp  Receiver Reports: Information about received data, losses, jitter and delay  Source Description:Name, , Phone, Identification  Bye: Explicit leave indication  Application defined parts: Parts for experimental functions

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Compressed RTP  Technique for reducing the bandwidth requirements for RTP-UDP-IP headers  Reduces all three headers from 40 bytes to 2-4 bytes  RTP Header = 12 bytes  UDP Header = 8 bytes  IP Header = 20 bytes  Utilizes the fact that much the headers’ contents remain the same from packet to packet  Critical for low-speed uplinks  Versions:  RFC 2508, CRTP for low-speed serial links  RFC 3545, Enhanced CRTP for high delay, packet loss, and reordering

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Other Issues  Interoperability between signaling protocols  Gateways can convert between protocols  Handling modem and fax traffic  Detection needed at gateway  T.37/T.38 Fax Delivery of IP  Modems must use G.711 with no echo cancellation and no high pass filter VoIP Network IP-to-PSTN Gateway IP-to-PSTN Gateway Fax Modem Fax Modem PSTN

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP  What is VoIP?  Why is it used?  How is it used?  Applications and architectures  How does VoIP work?  Protocols  What do VoIP calls sound like?  QoS  How can I make sure that VoIP deployments will work properly?  Modeling and simulation Agenda

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP What Do VoIP Calls Sound Like?  Sound quality depends on many factors  Telephone quality  Type of CODEC used  Higher compression leads to lower quality  Network performance  Quality of Service Metrics

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Subjective Versus Objective Quality Scoring  Mean Opinion Score (MOS)  A telephone industry standard for measuring voice quality  Based on users’ perceptions of voice quality  Excellent = 5, Good = 4, Fair = 3, Poor = 2, Bad = 1  MOS should be > 4.0  E-model, ITU G.107  Predicts the MOS based on  CODEC characteristics  Packet loss  Delay  Jitter Excellent Good Poor Bad Fair

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Quality of Service Metrics  Packet Loss – What percentage of the packets are dropped  Should be less than 1%  Delay – How much time elapses between when an utterance is spoken and when it is played back at the receiver  Must be less than 150 ms for real-time conversations  Jitter – The variability in the delay  Must be less than 30 ms  De-jitter buffer helps fix the problem, but adds to the overall delay

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Example of Delay Budget  Delays of less than 150 ms are sought  But the fixed components of delay can be high  Careful control of the variable components (queuing) required Delay ComponentFixed/VariableDelay (msec) Codec-Related g729a Compression Delayfixed5 g729a Sampling Delay (10 ms x 2)fixed20 Queuing Delay on Trunkvariable5 Transmission Delayfixed3 Propagation Delayfixed25 Queuing at Intermediate Hopsvariable20 De-jitter bufferfixed50 Total of Fixed Delays103 Total of Variable (Queuing) Delays25 Total Delay128

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Active Quality Measurement Systems  Use active network to monitor the QoS of a VoIP Network  Examine actual calls check performance  Set up extra calls on real network to test performance  Monitoring software is embedded in gateways and other devices  Use E-model to estimate MOS  Vendors  Psytechnics  RADCOM  Agilent

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP QoS Mechanisms – Queuing  Queuing – Mechanisms for giving different treatment to different types of packets  First In, First Out (FIFO)  Default behavior  Priority queuing (PQ)  Strict ordering of queues  Weighted Fair Queuing (WFQ)  Each queue gets a percentage of the bandwidth during congestion  Combination  A single high-priority queue + WFQ + best-effort queue

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP FIFO Queue Example  Voice packets can get delayed or even dropped due to interaction with data flows Voice Flow Data Flows FIFO Queue Packets lost due to tail drop during congestion As the queue length grows, so does the average delay The varying length of the queue adds to the jitter Multiplexer

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Example of WFQ + Priority Queue  Voice packets are always transmitted first via the “Priority FIFO Queue” Voice Flow Data Flows Priority FIFO Queue Classifier WFQ Queues Scheduler Best-effort Queue …

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP QoS Mechanisms  Ethernet QoS – 802.1p  IntServ – A mechanism for a reserving resources on devices via RSVP signaling  Fine-grained  Not scalable  DiffServ – A static mechanism for marking packets at the edge of the network and giving per-class treatment within the network  Coarse  Scalable  No signaling  MPLS-DiffServ-TE  Using label switched paths to control the paths that packets take through the network as well as the treatment they receive at each hop  Call Admission Control (CAC)  Gatekeeper/Proxy function for limiting number of calls in system

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP  What is VoIP?  Why is it used?  How is it used?  Applications and architectures  How does VoIP work?  Protocols  What do VoIP calls sound like?  QoS  How can I make sure that VoIP deployments will work properly?  Modeling and simulation Agenda

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Deployment Considerations  QoS strategy  Server location  Signaling latency issues  Load balancing  Redundancy  Dial plan  PSTN backup  Electrical power  Connectivity to Voice Mail and other Integration Voice Response (IVR) systems  Cooperation between telecom and data teams

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Modeling and Simulation  Configuration analysis  Process configuration files for errors and security problems  Readiness assessment  Propagation delay prediction  Failure analysis  Capacity planning  Using flow analysis to determine the appropriate link sizes in a VoIP network  Voice traffic conversion: erlangs to bits/sec  QoS configuration planning  Setting queue sizes  Voice quality analysis  Using discrete event simulation (DES) to model packet loss, delay, jitter of voice calls  Protocol modeling  Using ACE or DES to model and verify VoIP signaling protocols and signaling latencies

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Useful VoIP Links  Transition to VoIP in campus   Market research   General information    SIP information   CODEC calculator 

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Documentation References  H.323 ITU Standard for Voice/Video over IP  SIP – Session Initialization Protocol, IETF RFC 2543  MGCP – Media Gateway Control Protocol, IETF RFC 2705  H.248, Megaco, IETF RFC 2885  SCCP – Skinny Client Control Protocol  RTP – Real-time Transport Protocol, IETF RFC 1889  RTCP – RTP Control Protocol, IETF RFC 1889  CRTP for low-speed serial links, RFC 2508  Enhanced CRTP for high delay, packet loss, and reordering, RFC 2508  ITU-T.37 – Procedures for the Transfer of Facsimile Data Via Store-and-forward on the Internet  ITU-T.38 – Procedures for Real-time Group 3 Facsimile Communication over IP Networks

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Related OPNETWORK Sessions  1346 Planning and Analyzing VoIP Deployments  Thursday, 14:00-16:00, Atrium Ballroom B  1352 Case Studies: VoIP and Circuit-to-Packet  Thursday, 14:00-16:00, Continental A  1806 Introduction to QoS Mechanisms  Thursday, 14:00-16:00, Continental C  1337 Case Studies: QoS I  1338 Case Studies: QoS II  Thursday, 16:00-18:00, Polaris C

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties Introduction to VoIP Take-Away Points  VoIP can take many forms  Toll-bypass, PBX, access, trunking  Many signaling protocols and architectures will be deployed  Providing QoS guarantees is critical to VoIP success  Modeling and simulation tools can help address these issues  The VoIP market is growing – Get prepared!