1 / 48

Introduction to VoIP Technology Tutorials

Session 1819. Introduction to VoIP Technology Tutorials. Agenda. 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?

baby
Télécharger la présentation

Introduction to VoIP Technology Tutorials

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Session 1819 Introduction to VoIP Technology Tutorials

  2. Agenda • 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

  3. IP Network What is VoIP? • Carrying voice conservations over Internet protocol packet networks • Private • Public • There are other flavors of packetized voice: • VoATM • VoFR

  4. Why Use VoIP? • Cost savings • Integrated data and voice networks • Device interoperability using standards-based protocols • Flexibility in deriving new services

  5. Traditional Voice Versus VoIP • A traditional T1 can carry 24 telephone calls simultaneously • With VoIP, a T1 can carry 64 calls simultaneously! T1 = 1544 kbps, DS0 = 64 kbps, 1544 / 64 = 24 DS0 per T1 G.729 8kbps compression, 20 msec frame size = 24 kbps 1544 / 24 = 64 calls per T1

  6. 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.

  7. How is VoIP Used? • Applications and architectures • Consumer • Campus • Enterprise • Service Provider

  8. The Internet PC Modem Modem PC Consumer: IP-to-IP • Uses PC software to make calls over public and private internets • Free!! • But, no quality of service guarantees • Examples: • Microsoft NetMeetingTM • SkypeTM • Hybrids • PC2PhoneTM

  9. PSTN Campus Applications – IP PBX • Connecting office telephones to PBX with VoIP links • Vendors • Cisco • Nortel LAN Switches Call Manager & IP-to-PSTN Gateway IP Telephones

  10. Public Switched Telephone Network (PSTN) Private Data Network or VPN Router Router PBX PBX Enterprise Applications – Toll Bypass • Connecting enterprise PBXs with VoIP links to avoid paying for long distance charges • Vendors: • Nortel • NEC • Avaya • Toshiba • Ericsson • Cisco

  11. PSTN Access Provider Broadband Service Provider Ordinary Telephone ISP Broadband Modem Splitter Service Provider Applications – Local Access • Using broadband access to provide local and long distance telephone service • Example Services: • Vonage • ATT CallVantageTM • Packet8 • Broadvox • Time Warner Cable PC

  12. LATA #6 LATA #2 LATA #3 LATA #5 LATA #1 LATA #4 PSTN Private Data Network Gateway Gateway 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

  13. Agenda • 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

  14. 1. Caller dials 555-1234 3. Gatekeeper responds with the IP address of the called party Gatekeeper 4. Caller sends a call setup message to the called party IP Phone IP Phone LAN 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 How Does VoIP Work? 2. Gatekeeper performs authentication, call admission control, and address translation

  15. 3. Gatekeeper responds with the IP address of the gateway 1. Caller dials 555-1234 5. Gateway converts the VoIP signaling message to PSTN signaling message 4. Caller sends a call setup message to the gateway PSTN Ordinary Telephone IP Phone IP-to-PSTN Gateway LAN 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. 7.Gateway converts the PSTN accept message into VoIP accept message and sends it back to the caller. How Does VoIP Work? (IP-to-PSTN) 2. Gatekeeper performs authentication, call admission control, and address translation Gatekeeper

  16. 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)

  17. Distributed Centralized H.323 SIP MGCP H.248 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

  18. Audio Video Call Manager Application G.711 G.729 G.723 H.261 H.263 RTCP H.255 RAS H.245 Control Signaling H.225 Call Signaling T.120 Data RTP Transport Protocols 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

  19. 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

  20. 1. Caller dials 555-1234 3. Gatekeeper responds with the IP address of the called party Gatekeeper 4. Caller sends a call setup message to the called party LAN 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 H.323 Call Setup 2. Gatekeeper performs authentication, call admission control, and address translation H.323 Messages IP Phone IP Phone (555-1234)

  21. 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: • sip:bobsmith@mycompany.com • sip:voicemail@mycompany.com?subject=callme • sip:+1-919-555-1234@gateway.mycompany.com • tel:+1-919-555-1234 • DTMFs carried in signaling message

  22. INVITE sip:joe@sip.com From: bob@opnet.com To:joe@sip.com Call-ID:12345@opnet.com INVITE sip:joe@123.23.44.3 From: bob@opnet.com To:joe@sip.com Call-ID:12345@opnet.com SIP Proxy OK 200 From: bob@opnet.com To:joe@sip.com Call-ID:12345@opnet.com IP Network OK 200 From: bob@opnet.com To:joe@sip.com Call-ID:12345@opnet.com Voice packets flow between IP telephones ACK joe@sip.com SIP Call Setup Proxy for sip.com gets location information for called party. IP Phone (bob@opnet.com) IP Phone (joe@sip.com)

  23. Signaling Gateway Media Gateway Controller Media Gateway Controller IP Phone PSTN Media Gateway 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 SS7, etc. Call Control (SIP, H.323, etc.) H.248 Messages Trunks

  24. Break! Break!!!

  25. 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 • 10-80 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

  26. Typical CODEC Behavior

  27. Codec Compression Method Codec Bit Rate Block Length Block Size (bytes) Blocks per Packet Voice Call Bandwidth Required(Excl. L2 o/h) Mean Opinion Score g711alaw PCM 64000 10 ms 80 2 80000 4.1 0.75 Compression Delay (ms) g711ulaw PCM 64000 10 ms 80 2 80000 g723ar53 ACELP 5300 10 ms 7 2 22000 3.65 30 g723ar63 MP-MLQ 6300 10 ms 8 2 23000 3.9 30 g723r53 ACELP 5300 10 ms 7 2 22000 g723r63 MP-MLQ 6300 10 ms 8 2 23000 g726r16 ADPCM 16000 10 ms 20 2 32000 g726r24 ADPCM 24000 10 ms 30 2 40000 g726r32 ADPCM 32000 10 ms 40 2 48000 3.85 1 g728 LD-CELP 16000 10 ms 20 2 32000 3-5 g729r8 CS-ACELP 8000 10 ms 10 2 24000 3.92 10 g729br8 CS-ACELP 8000 10 ms 10 2 24000 3.7 10 CODEC Characteristics

  28. V P X M Payload Sequence Number Timestamp Synchronization Source Identifier (SSRC) Payload Real-time Transport Protocol (RTP) • Media content type • Talk spurts • Sender identification • Synchronization • Loss detection • Segmentation and reassembly • Security (encryption)

  29. 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, Email, Phone, Identification • Bye: Explicit leave indication • Application defined parts: Parts for experimental functions

  30. 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

  31. PSTN PSTN VoIP Network Fax Fax IP-to-PSTN Gateway IP-to-PSTN Gateway Modem Modem 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 SIP to H.323 SIP to Megaco SIP to PSTN H.323 to PSTN Megaco to PSTN

  32. Agenda • 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

  33. 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

  34. Excellent Good Fair Poor Bad 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

  35. 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

  36. 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 Component Fixed/Variable Delay (msec) Codec-Related g729a Compression Delay fixed 5 g729a Sampling Delay (10 ms x 2) fixed 20 Queuing Delay on Trunk variable 5 Transmission Delay fixed 3 Propagation Delay fixed 25 Queuing at Intermediate Hops variable 20 De-jitter buffer fixed 50 Total of Fixed Delays 103 Total of Variable (Queuing) Delays 25 Total Delay 128

  37. 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

  38. 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

  39. Multiplexer FIFO Queue Example • Voice packets can get delayed or even dropped due to interaction with data flows Voice Flow FIFO Queue Data Flows As the queue length grows, so does the average delay The varying length of the queue adds to the jitter Packets lost due to tail drop during congestion

  40. Classifier Scheduler Example of WFQ + Priority Queue • Voice packets are always transmitted first via the “Priority FIFO Queue” Priority FIFO Queue Voice Flow WFQ Queues Data Flows … Best-effort Queue

  41. 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

  42. Agenda • 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

  43. 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

  44. 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

  45. Useful VoIP Links • Transition to VoIP in campus • http://www.cisco.com/warp/public/cc/so/neso/vvda/avvid/ttpnp_bc.pdf • Market research • http://www.sonusnetworks.com/contents/brochures/solutions/Market_Impact.pdf • General information • http://www.voip-news.com/ • http://www.voip-info.org/ • SIP information • http://www.cs.columbia.edu/sip/ • CODEC calculator • http://www.voip-calculator.com/calculator/lipb/

  46. 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

  47. 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

  48. 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!

More Related