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IIT101 – Introduction to Voice Over IP Technology © Internation Institute of Telecommunications inc., 2000-2004 www.iitelecom.com.

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Presentation on theme: "IIT101 – Introduction to Voice Over IP Technology © Internation Institute of Telecommunications inc., 2000-2004 www.iitelecom.com."— Presentation transcript:

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2 IIT101 – Introduction to Voice Over IP Technology © Internation Institute of Telecommunications inc., 2000-2004 www.iitelecom.com

3 © IITelecom, 2004 2 IIT 101 Logistics Schedule : from 8h30 to 16h30 Break: 10h00 to 10h15 Meal: 11h45 to 13h00 Break: 14h45 to 15h00 Directions Toilets Telephones Concessions Instructions In the event of emergency Food, drink, smoking

4 © IITelecom, 2004 3 IIT 101 Course content Objectives: The participant will be able to: –Describe the routing of a voice signal in an IP network; –Define various compression and sampling methods; –Define the protocols used with IP; –Describe the architectures and the components of a VoIP network; –Describe various services and applications. Contents: LU 1: Traffic and networks LU 2: Signal processing LU 3: QoS Protocols LU 4: Network architectures and components LU 5: Implementation considerations LU 6: Applications

5 © IITelecom, 2004 4 IIT 101 Round table Your name Your employer Your roles and functions Your expertise and your knowledge in telecommunications OSI, LAN, IP, Voice Networks Your expectation about the course

6 © IITelecom, 2004 5 IIT 101 LU 1: Traffic and networks Training objectives: The participant will be able to: –Characterize each traffic type –Characterize the network types –Describe the operation and the components of a PSTN –Describe the operation and the components and of an IP network –Identify the performance factors influencing a VoIP network Contents: Isochronous traffic and the PSTN Networks and the IP protocol The integration of Voice over IP

7 © IITelecom, 2004 6 IIT 101 Activity 1.1 - Integration of VoIP Why integrate voice in an IP network? Constraints Challenges Market trends

8 © IITelecom, 2004 7 IIT 101 Why integrate voice over IP? To reduce the costs? To maximize bandwidth usage To use a single network? Telephony on Intranet and Internet To create new commercial applications? To integrate voice mail, email and fax Call Centers –CTI (Computer Telephony Integration) –Telephony on Internet Integration of voice-data-images?

9 © IITelecom, 2004 8 IIT 101 VoIP traffic on the network Traffic: voice, video… Isochronous traffic –Fixed intervals Throughput –Constant (voice) –Variable (video) Short delays The IP network: Asynchronous mode of transfer –Process by which the data can be transmitted at unspecified intervals. –Generate variable and unforeseeable delays –Designed for non isochronous traffic TTTTT

10 © IITelecom, 2004 9 IIT 101 The traffic adaptation to the network

11 © IITelecom, 2004 10 IIT 101 3 performance factors Name three factors which will influence the ambulance delay :

12 © IITelecom, 2004 11 IIT 101 3 performance factors Transpose these 3 factors in a network environment and relate them to packet latencies at a LAN exit point Serial Link Router LAN Router LAN

13 © IITelecom, 2004 12 IIT 101 Constraints / Solutions Delays Delay variation/jitter Echo Compatibility/interworking Inter local area networks (QoS) With the PSTN –Signaling and supervision –Added value services Information security Sampling Define packets Signal compression RTP/UDP QoS Priority –RSVP –ToS/Diffserv Field –IPv6 Dimensioning/capacity Equipment Gateway, gatekeeper

14 © IITelecom, 2004 13 IIT 101 What are the challenges? The signal must be transported via various network types –A great number of technologies and standards are used –Interworking and interconnections between various technologies and equipment The applications, the services, the technologies and the standards evolve quickly –Proprietary solutions versus standards Maintain the quality of the voice signal –Toll Quality or equivalent To preserve the current telephone services –Call transfer, call on hold, caller ID, etc. Interworking between the traditional voice network and the IP network –Seamless services –Addressing, classification –Quality of service, MTBF, MTTR –Security and confidentiality Billing Operations and management

15 © IITelecom, 2004 14 IIT 101 Equipment vendors Industry standards (ANSI, ITU, IEEE, IETF) Technology (platform, chip, software) Users (residential, commercial) Carriers Internet service providers Who are the stakeholders?

16 © IITelecom, 2004 15 IIT 101 Market trends (U.S.) Source: Phillips InfoTech 2002 2001 2002 2003 2004 2005 2006 2000 4000 6000 8000 10000 12000 14000 Million lines Overall market Traditional market IP Telephony

17 © IITelecom, 2004 16 IIT 101 In the year 2000, some figures… * Frost & Sullivan ** Synergy research *** Allied Business Intelligence **** Phillips Group ***** Ovum VoIP will account for 75 % of the world voice services in 2007. * The number of IP Centrex lines should grow from 13,000 in 2001 to 10 million in 2008. The number of standard Centrex lines in 2001 was estimated at 16,5 million. * The market for IP PBX should reach 3,9 billion dollars in 2005, which represents 20 % of the traditional PBX market. ** The companies will migrate their voice service from traditional networks towards IP networks at a rate which will generate a world market for IP PBX of 16,5 billion dollars from here to 2006. *** 90 % of the companies operating in multiple sites will migrate towards IP systems for the transport of the voice in the next 5 years. **** 25 % of Internet users will adopt IP Telephony PC-TO-TELEPHONE in 2006. ****

18 © IITelecom, 2004 17 IIT 101 In the year 2000, some figures… (cont’d) Unified messaging applications will bloom in 2004 and 2005. The expenditure related to these applications should reach 3.5 billion dollars in 2005 with an annual growth rate of 32.1 % * Call center systems will constitute nearly 30 % of the market for VoIP systems from here to 2003 ** * TIA (Telecommunication Industry Association) ** IDC (International Dated Corporation)

19 © IITelecom, 2004 18 IIT 101 Questions? ?

20 © IITelecom, 2004 19 IIT 101 LU 2: Signal processing Training objectives: The participant will be able to: –Identify and enumerate the protocols used in the transport of voice in a IP network –Enumerate the various sampling procedures and compression standards –Describe the various delay sources and describe their effect on signal quality –Describe the role and the operation of RTP and RTCP protocols Contents: The protocol suite Sampling and compression Delay sources Bandwidth usage Delay variation (Jitter) RTP/RTCP

21 © IITelecom, 2004 20 IIT 101 Activity 2.1 The protocol suite Laboratory activity IP Telephony experimentation Discovering the protocol suite

22 © IITelecom, 2004 21 IIT 101 Laboratory activity 2.1 IP telephony experimentation; Using a protocol analyzer; Discovering the protocol suite.

23 © IITelecom, 2004 22 IIT 101 Equipment configuration

24 © IITelecom, 2004 23 IIT 101 Laboratory activity2.1 (cont'd) Experimenting with IP telephony; Using a protocol analyzer; Discovering the protocol suite.

25 © IITelecom, 2004 24 IIT 101 Activity 2.1/IP Telephony and protocol suite Your Observations Discussion

26 © IITelecom, 2004 25 IIT 101 The protocol suite PCM compression sampling packetization Ethernet Checksum (FCS) Voice sample N O 1 Voice sample N O 2 Voice sample N O 3 RTP Header UDP Header IP Header Ethernet/LLC header 4 bytes n bytes 12 bytes 8 bytes 20 bytes 18 bytes

27 © IITelecom, 2004 26 IIT 101 Activity 2.2 Sampling and compression Sampling Delay Echo Compression Bandwidth usage

28 © IITelecom, 2004 27 IIT 101 Packets construction Sampling Continuous signal conversion into individual samples Size of a sample Which will be the size (in bytes) of a 20ms voice signal sample when using a 64 Kb/s PCM? – The packet size has a significant effect on the quality of service –Delay –Echo –Delay variation sampling

29 © IITelecom, 2004 28 IIT 101 Packet size versus tolerable delay 200 - 250 ms is tolerable Overlaps if higher > 250ms Did you see the storm yesterday? How is your sister? He cuts me all the time

30 © IITelecom, 2004 29 IIT 101 Delay Sources Serialization time Propagation time Processing time “Processing delay” Time to sit in memory “Queuing delay”

31 © IITelecom, 2004 30 IIT 101 Serialization time Time necessary for the transmission of the first to the last bit of a given packet. Called transmission delay Delay = L/C 1 0 1 0 0 1 1 0 1 1 0 0 0 1 0 L [ bits ] C [ bps ] Time (ms) Packet Length (Bytes) Circuit speed (kbps) 56 128 256 1 536 60 8,6 3,8 1,9 0,3 500 71,4 31,3 15,6 2,6 1 500 214 93,8 46,9 7,8 4 000 571 250 125 20,8

32 © IITelecom, 2004 31 IIT 101 Propagation time T 1 T2T2 Delay = T 1 + T 2 [ms]

33 © IITelecom, 2004 32 IIT 101 Processing time Synchronization Sampling, compression Packetization/depacketization Decision of route choice (routing) Multiplexing … PayloadHeader Processor PayloadHeader

34 © IITelecom, 2004 33 IIT 101 Queuing / Time to sit in memory PayloadHeader PayloadHeader PayloadHeader PayloadHeader Memory Queuing delay

35 © IITelecom, 2004 34 IIT 101 End-to-end Delays IP network PBX Gateway PBX Gateway Queuing, Processing and Serialization Delays Propagation Delay Queuing, Processing and Serialization Delays

36 © IITelecom, 2004 35 IIT 101 What is echo? One way delay; if higher than 20 - 30 ms, a person distinguishes the echo. How can we cancel echo? Hello! Hybrid junction Hello! Do I hear echo ? PSTN

37 © IITelecom, 2004 36 IIT 101 What causes echo? Residential sector Telephone exchange Telephone Local loop 2 wire to 4 wire conversion Bad impedance matching Hybrid Junction Reception Transmission

38 © IITelecom, 2004 37 IIT 101 Echo is always present Echo varies according to the delay and the power of the return signal. (dB) Return signal, reduction of power Delay (ms) Echo poses a problem Echo is undetectable

39 © IITelecom, 2004 38 IIT 101 How to eliminate/minimize echo? Decrease the return signal –Advantage: An easy solution –Disadvantages: Human intervention Reduction of the signal power for the person who speaks Installation of echo cancellers –Advantage: Eliminate the return signal (echo) –Disadvantage: Costs Reduction of the delays

40 © IITelecom, 2004 39 IIT 101 Voice compression Objectives: –To reduce bandwidth usage. –To reduce packet size Compression algorithms are used for video and voice. Disadvantages: –Reduction in voice quality –Introduction of delay (echo) 64 32 24 16 8 0 Bandwidth(kbit/s) Quality Not acceptable Quality commercial Toll Quality PCM (G.711) Cellular ADPCM 32 (G.726) ADPCM 24 (G.726) ADPCM 16 (G.726) LD-CELP 16 (G.728) CS-ACELP 8 (G.729) MPC-MLQ 5,3 (G.723.1)

41 © IITelecom, 2004 40 IIT 101 Mean Opinion Score (MOS) Evaluation of the sound quality with various compression methods on a scale of 1 (bad) to 5 (excellent) A result of 4,0 is considered “Toll Quality” Test conditions: –With background noise, several people discussing at the same time; –Individuals speaking various languages, men and women; –Etc. G.711PCM 64 kbit/s 64 kbit/s G.726ADPCM 32 kbit/s 32 kbit/s G.728LD-CELP 16 kbit/s 16 kbit/s G.729CS-ACELP 8 kbit/s G.729aCS-ACELP 8 kbit/s 8 kbit/s G.723.1 MPC-MLQ MPC-MLQ 5,3 kbit/s

42 © IITelecom, 2004 41 IIT 101 Silence suppression Automatically deactivated for fax/modem Background noise generated at the destination Time -31 dbm -54 dbm Voice signal power power Meter Bandwidth Economy Economy Voice Signal Silence

43 © IITelecom, 2004 42 IIT 101 Large DATA packets affect voice WAN 56 kbit/s PBX Gateway PBX Gateway Voice packets Voice packets 60 bytes every 20 ms 1 500 bytes data Voice Voice packets Voice packets 60 bytes every 214 ms Data Packet Serialization time: 214 ms Voice 1 500 bytes data VoiceVoice Voice Voice

44 © IITelecom, 2004 43 IIT 101 Solution - Fragmentation WAN 56 kbit/s PBX Gateway PBX Gateway Voice packets Voice packets 60 bytes every 71,4 ms 500 Data packet Serialization time 71,4 ms Voice packets Voice packets 60 bytes every 20 ms 1 500 bytes data Voice Voice Voice500Voice500 Voice Voice

45 © IITelecom, 2004 44 IIT 101 Packet construction The packetization The sample RTP UDP IP LLC/802.3 12 bytes 8 bytes 8 bytes 20 bytes 22 bytes

46 © IITelecom, 2004 45 IIT 101 VersionIHLType of ServiceLength Total IdentificationFlagsFragment Offset Header ChecksumProtocolTime to Live Address Source Address Destination PaddingOptions Port SourceDestination Port ChecksumLength PtMDCXPV=2Number Sequence Timestamp Synchronization Source (SSRC) To identify Header compression CRTP - Compressed Real-time Protocol - RFC2508 Used on link with low bandwidth G.729: 20ms@8kb/s gives 20 bytes of information 40 bytes per packet: header IP 20; header UDP 8; header RTP 12 The 40 bytes of header are compressed to 2-4 bytes

47 © IITelecom, 2004 46 IIT 101 Quiz Associate the various methods of coding and compression to the bandwidth used: G.711 PCM ___ a) 32, 24, 16 kbit/s G.726 ADPCM ___ b) 5,3, 6,4 kbit/s G.728 LD-CELP ___ c) 16 kbit/s G.729 CS-ACELP ___ d) 64 kbit/s G.723.1 MPC-MLQ ___ e) 8 kbit/s

48 © IITelecom, 2004 47 IIT 101 Quiz Which of these statements is responsible for echo? a) Too great distances between two telephones b) Delay between two telephones superior > 30 ms c) Return Signal too high in the hybrid junction d) Bad impedance matching in hybrid junction e) All these answers

49 © IITelecom, 2004 48 IIT 101 Laboratory activity 2.2 Evaluation of voice quality with IP in relation with packet size.

50 © IITelecom, 2004 49 IIT 101 Activity 2.3 Delay variation, RTP/RTCP and UDP Delay variation causes RTP and RTCP The use of UDP

51 © IITelecom, 2004 50 IIT 101 Delay variation/ Jitter IP network PBX Gateway PBX Gateway D2 = D1 ABC D1 ABC D1 Congestion

52 © IITelecom, 2004 51 IIT 101 Delay variation causes Serialization time Dynamic route Variable number of routers Variable link bandwidth Propagation time Dynamic route Variable link bandwidth Variable distance Processing time Dynamic route Variable number of routers Queuing delay Dynamic route A variable number of routers Variable load on the networkSourceDestination 3 2 1 3 3 3 3 2 2 2 1 1 1 1

53 © IITelecom, 2004 52 IIT 101 Real Time Transport Protocol (RTP) Provides end-to-end transport functions for the applications that need real time video and audio. –Identification of the compression type –Packet sequencing –Packet lost detection –Synchronization RTP uses UDP for transport RFC 1889, January 1996 January 1996 Netscape Live Media based on RTP Microsoft announced that NetMeeting supported RTP

54 © IITelecom, 2004 53 IIT 101 RTP does not offer network resource reservation guarantees of delivery within the required delay guarantees of quality of service guarantees of packets delivery

55 © IITelecom, 2004 54 IIT 101 RTP header description 4 bytes 161718192021222324252627282930310123456789101112131415 V PT DC SEQUENCE NUMBER SYNCHRONIZATION SOURCE (SSRC) IDENTIFIER CONTRIBUTING SOURCE (CSRC) IDENTIFIERS (1 …) PXM TIMESTAMP V: Version P: Padding X: Extension CC: CSRC count PT :Payload Type CONTRIBUTING SOURCE (CSRC) IDENTIFIERS (… 15) … Voice Samples … Usefulpayload

56 © IITelecom, 2004 55 IIT 101 Payload type Type Encoding Audio/Video Clock (Hz) 2 G.721 A 8 000 4 G.723 A 8 000 7 LPC A 8 000 9 G.722 A 8 000 15 G.728 A 8 000 26 JPEG V 90 000 31 _ H.261 V 90 000 34 _ H.263 V 90 000

57 © IITelecom, 2004 56 IIT 101 RTP Timestamp Inter-packet delay - 20ms C C B B A A 10 30 50 20ms NetworkIP RTP Timestamp Delay variation/Jitter C C B B A A 10 30 50 20ms80ms RTP Timestamp Delay variations elimination C C B B A A 10 30 50 20ms Controlling delay variation/Jitter router A router B

58 © IITelecom, 2004 57 IIT 101 Real-time Transport Control Protocol (RTCP) Control protocol intended to work jointly with RTP Provides information for a RTP session in progress. –Number of packets received –Number of packets lost –Delay between each packet (Jitter) –Timestamps for end-to-end delay calculation Services offered –Monitors the quality of service –Controls congestion –Source identification –Inter-media synchronization (sound and image)

59 © IITelecom, 2004 58 IIT 101 RTCP header description 1 2 3 4 5 6 7 8 bits Reception report number P Length Ver Packet Type

60 © IITelecom, 2004 59 IIT 101 RTCP control packet types Value Standard Reports 200 Sender Report (SR) (Synchronization, quantity of bytes transmitted) 201 Receiver Report (RR) (Packets receive/lost, Jitter, Timestamps) 202 Source Description (SDES) (name, telephone number, email address) 203 Greeting (BYE) 204 Application Definition (APP) (Future Use)

61 © IITelecom, 2004 60 IIT 101 TCP/UDP TCP: Logical connection Deliveries acknowledgement Checks for data errors Retransmission of the lost or erroneous segments Sequence check Flow control  Used for the transport of applications sensitive to errors but less sensitive to delays UDP: Without connection Check for data errors Do not offer: –delivery acknowledgement –flow control –retransmission –sequence check  Used for the transport of real-time applications sensitive to delays but where errors are less important than delays

62 © IITelecom, 2004 61 IIT 101 Quiz Associate the following protocols to the function: IP ___ TCP ___ UDP ___ RTP ___ RTCP ___ RSVP ___ a) Bandwidth reservation and quality of service b) Control the quality of service (time, packet received) c) Sequencing, synchronization, detection of the lost packets d) Packet routing on the network e) Retransmission, flow control, delivery acknowledgement f) Identification of connections without delivery acknowledgement and any retransmission

63 © IITelecom, 2004 62 IIT 101 Real-time Protocol Transport (RTP) Real-time Transport Protocol Control (RTCP) Laboratory Activity 2.3 Demonstration of RTP and RTCP protocols.

64 © IITelecom, 2004 63 IIT 101 Questions? ?

65 © IITelecom, 2004 64 IIT 101 LU 3 - Quality of service (QoS) Training objectives : The participant will be able to describe the protocols used to support QoS in IP networks, in order to offer a better voice quality Contents: Congestion causes QoS role QoS Protocols –IPv4 TOS Field –DiffServ (Differentiated Services) –RSVP (Resource Reservation Protocol) –MPLS (Multi-Protocol Label Switching) –IPv6

66 © IITelecom, 2004 65 IIT 101 What is QoS? Methods (i.e protocols, dimensioning, architectures…) used to fulfill an application transmission requirements (delays/ errors rate). A service allowing to fulfill an application congestion requirements without affecting its performance A set of traffic parameters to be managed –Bandwidth –Delay –Delay variation (Jitter) –Packet Loss Can be associated to business priorities through an administration tool containing rules The base for SLAs between service operators and clients

67 © IITelecom, 2004 66 IIT 101 QoS role In the corporate network To ensure the priority of Real time traffic (voice/video) To ensure an appropriate level of service to corporate critical applications In the service provider network QoS allows service providers to offer services with SLA

68 © IITelecom, 2004 67 IIT 101 How can we increase QoS in a network? By Reducing the delay Decrease the packet size Increase the bandwidth … By compensating for delay variation Use RTP … By reducing the number of lost packets Increase the capacity of the buffer Increase the bandwidth …

69 © IITelecom, 2004 68 IIT 101 Priority establishment VoiceVoiceDataVoiceVoiceData VoiceVoice Data Classification By: IP address ProtocolPort/Socketetc Buffer Transmission QoS function Weighted Fair Queuing (WFQ) Router

70 © IITelecom, 2004 69 IIT 101 DiffServ (differentiated services) Principle Grants a particular treatment to packets requiring it Assigns various classes of services to packets Use of 6 bits in the IP header (IPv4 TOS fields and IPv6 Traffic Class) Differentiated Services Delay and delivery guaranteed Best Effort Delivery Guaranteed delivery Voice Email, Web Browsing E-Commerce ERP (Enterprise Resource Planning) Platinum Service low delay Silver Bronze Gold Voice Classification of traffic

71 © IITelecom, 2004 70 IIT 101 DS Field (DiffServ) VersionLengthLen IPv4 IDOffsetTTLProtoCSIP-ITSIP-DADated ToS 1 Byte 70 6 5 4 321 01234567 TOS Field Bits RFC 1122 RFC 1349 Must Be Zero Type of Service MBZ 01234567 DSCPCU IP Precedence DS Field Differentiated Services Codes Points (DSCP) - RFC 2474 Currently Unused

72 © IITelecom, 2004 71 IIT 101 Packet Classification and Code Points EF AF12 AF22 AF32 AF42 AF13 AF23 AF33 AF43 Per-Hop Behaviours (PHB)/DiffServ Codepoints (DSCP) Expedited Forwarding Assured Forwarding Best Effort Classify 1 Classify 2 Classify 3 Classify 4 Rejection Priority WEAK Rejection Priority AVERAGE Rejection Priority STRONG 001 01 0 001 10 0 001 11 0 010 01 0 010 10 0 010 11 0 011 01 0 011 10 0 011 11 0 100 01 0 100 10 0 100 11 0 000000 101110 AF11 AF21 AF31 AF41

73 © IITelecom, 2004 72 IIT 101 DiffServ Operating modes Classifier –Sorts the packets in traffic classes (per hop behavior) Example: all the VoIP packets between UDP ports 16384 and 16484 belong to the Premium Class Marker –Marks (or colors) the packets by assigning them a DSCP value Example: the VoIP packet Premium Class will be marked with DSCP value - 101110 Meter (Optional) –Checks the conformity with the traffic profile and gives the non-conforming and conforming packets to the Marker or the Shaper/Dropper for processing Shaper/Dropper –Accepts the traffic but with a lower bandwidth (a few packet are put in queue to conform to traffic profiles) (regulating) –Rejects the excess of packets in the event of congestion

74 © IITelecom, 2004 73 IIT 101 Resource Reservation Protocol (RSVP) Control protocol between two network equipments –station and router –router and router The receiving station specifies the quality of service (QoS) necessary in the network before receiving information. The routers accept or refuse the request, depending on the network status. The routers give priority to the traffic having the greatest quality of service specified. Priority Class

75 © IITelecom, 2004 74 IIT 101 RTP, RTCP and RSVP in a multi-media session Real time applications RTP and RTCP UDP RTP, RTCP and RSVP Applications RSVP RSVP RouterRouter RTPRTP RTPRTP RSVP RSVP Real Time Server RTP RTCP RTCP Each media (voice, video, data) are transported in a different RTP session, with its own RTCP packets controlling the quality of the session. The routers communicate via RSVP to reserve and control the bandwidth of each session

76 © IITelecom, 2004 75 IIT 101 IP Switching Ipsilon 95/96 CSR Toshiba 94 ARIS IBM 94 Tag Switching Cisco 96/97 IP Navigator Ascend/Lucent 96 MPLS IETF 97/98 MPLS (Multi-Protocol Label Switching) Solution developed by IETF (RFC 3036 and 3037) To improve the performance of IP networks by introducing a switching mechanism based on the packet label To introduce traffic management (Traffic Engineering) by selecting a route based on QoS and by managing the traffic Combination of several proposals IP Switching (Ipsilon/Nokia) Tag Switching (Cisco) IP Navigator (Cascade/Ascend/Lucent) ARIS (IBM) CSR (Toshiba)

77 © IITelecom, 2004 76 IIT 101 MPLS Terminology Label Label being used to identify the packet and its routing LER (Label Edge Router) A router at the edge of the MPLS network which assigns the first label to the packets at the entry of the MPLS network and removes it at the exit LSR (Label Switching Router) A router or an ATM switch which processes the packets according to the label Forwarding Equivalence Class (FEC) Flow of IP packets transmitted using the same mechanism, processed in the same manner and identified by the same label Port 1 Port 3 Port 2 Port 4 Switching Table In (port, label) Out (port, label) (1, 2) (1, 4) (1, 5) (2, 3) (2, 7) (3, 7) (4, 9) (3, 2) 5IP 9

78 © IITelecom, 2004 77 IIT 101 MPLS header TTLLabelExp S IP Packet 32-bits MPLS Header IPv6 Flow Label Field LAN MAC Header PPP Header (Packet over SONET/SDH) ATM Cell Header VersionTC Label Header Flow Label … MAC HeaderLabel HeaderLayer 3 HeaderPPP HeaderLabel HeaderLayer 3 HeaderVPIVCIGFC Label Header … MPLS header Packetization 20 bits 3 bits1 bit8 bits Exp - Experimental (CoS) S - Bottom of Stack TTL - Time To Live

79 © IITelecom, 2004 78 IIT 101 MPLS Routing in networks 134.5.1.5 200.3.2.7 1 2 26 3 5 200.3.2.1 134.5.6.1 Routing Table DestinationNext Hop 134.5/16 200.3.2/24 (2, 84) (3, 99) MPLS Table InOut (1, 99)(2, 56) MPLS Table InOut (3, 56)(5, 3) Table MPLS InOut (2, 84)(6, 3) Destination Routing Table Next Hop 134.5/16 200.3.2/24 134.5.6.1 200.3.2.1 2 3 134.5.1.5 84 3 Label Edge Router (LER) Label Switching Router (LSR)

80 © IITelecom, 2004 79 IIT 101 IPv6 - Motivations Motivations IP address shortage Context of Always-on (cell phone, Palm, ADSL, etc.) NAT unidirectional nature for the VoIP applications Functionalities 128 bits Addressing (greater address space) –340282366920938463463374607431768211455 IP addresses –4 million unique addresses per square meter of the earth's surface Simplified header Header extension support for options Integrated security Better mobility QoS support

81 © IITelecom, 2004 80 IIT 101 Questions? ?

82 © IITelecom, 2004 81 IIT 101 LU 4 - Architectures and network components Training objectives : The participant will be able to: –Identify the various VoIP standards and specifications –Describe the components associated with various architectures –Describe the protocols associated with various architectures Contents: H.323 SIP (Session Initiation Protocol) MGCP (Media Gateway Control Protocol) and Megaco

83 © IITelecom, 2004 82 IIT 101 Activity 4.1 H.323, components and architecture The H.323 standard Network Components H.323 Protocols VoIP network architecture

84 © IITelecom, 2004 83 IIT 101 ITU H.323 To be able to transport real time voice and video data on a packet switched network. Allow the interworking between applications and equipment from various manufacturers. Defines the components and services used. H.323 v1 approved in May 1996 by ITU H.323 v2 approved in January 1998 by ITU H.323 v3 approved in September 1999 by ITU H.323 v4 approved in November 2000 by ITU

85 © IITelecom, 2004 84 IIT 101 H.323 Specific Protocols TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP G.711G.723.1G.726G.728G.729 H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931Signaling H.245 of control signaling Control and management of the calls Data Voice and Video Compression Standards

86 © IITelecom, 2004 85 IIT 101 TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931Signaling H.245 of control signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 Transport Protocols H.323 Specific Protocols (cont’d)

87 © IITelecom, 2004 86 IIT 101 H.323 Specific Protocols (cont’d) TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931Signaling H.245 of control signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 Data Transmission Protocols

88 © IITelecom, 2004 87 IIT 101 H.323 Specific Protocols (cont’d) TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931Signaling H.245 of control signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 Signaling and Call set-up Protocols

89 © IITelecom, 2004 88 IIT 101 H.323 environment and components NetworkIP H.323 terminal remote access Access Server GatekeeperPBX Gateway PSTN IP telephone H.323 terminal MCU

90 © IITelecom, 2004 89 IIT 101 MCU Gatekeeper Gateway Terminal Gatekeeper: the brain of the H.323 network.

91 © IITelecom, 2004 90 IIT 101  Admission Control which determines if a terminal can receive or initiate a call  Translation of the phone numbers by determining the H.323 terminal address to establish the call 514-841-3250 172.31.16.254 E.164 Number IP Addresses  Bandwidth Control  Billing  Zone Management (Terminals, Gateway, MCU) H.323 Gatekeeper Functions

92 © IITelecom, 2004 91 IIT 101 H.323 Gatekeeper Optional Functions Signaling Control –Establishes connection between two terminals or simply lets the terminals communicate between themselves Authorization of the calls Management of the bandwidth –Issue requests for additional bandwidth. Management of the calls –Determine if the terminal called is busy. –Call transfer Bandwidth reservation –For the terminals unable to make the reservation

93 © IITelecom, 2004 92 IIT 101 H.323 zone Gatekeeper 1 Zone 2 Zone 1 Gatekeeper 2

94 © IITelecom, 2004 93 IIT 101 H.323 terminals Must support: –H.225 Protocol (recording, admission control, signaling and call set-up with a gatekeeper) –H.245 Protocol (exchange of functionalities between the terminals and creation of transfer channels) –Audio compression G.711 Optional: –Audio compression G.723, G.729 –Video compression H.261 must be used if video is supported –Multipoint multimedia conference T.120 –Multipoint Control Unit (MCU) Real-time Protocol (RTP) is used for audio and video packets transmission.

95 © IITelecom, 2004 94 IIT 101 GW PSTN IP network Gateways Allows connectivity between a H.323 network and a non- H.323 network (e.g. PSTN) Provides translation functions : –Converts the transmission formats –Converts the signaling protocols

96 © IITelecom, 2004 95 IIT 101 H.323 interworking TerminalH.323 TerminalV.70 PSTN MCUH.323 TerminalH.323TerminalH.323GatewayH.323GatekeeperH.323 TerminalH.324TerminalvoiceTerminalH.322TerminalvoiceTerminalH.320TerminalH.321 LANQoSISDN(N-ISDN)ATM(B-ISDN) Local area IP network ZoneH.323

97 © IITelecom, 2004 96 IIT 101 Telecommuter PSTN Private IP Network VoIP network StructureGatekeeperGateway Internet GatekeeperGatewayGatekeeperGateway V

98 © IITelecom, 2004 97 IIT 101 Questions? ?

99 © IITelecom, 2004 98 IIT 101 Activity 4.2 H.323, Terminal configuration and RAS The H.323 standard –Signaling and control –H.225/RAS –Terminal configuration

100 © IITelecom, 2004 99 IIT 101 Laboratory activity 4.2.1 IP telephone autoconfiguration.

101 © IITelecom, 2004 100 IIT 101 Laboratory activity 4.2.2 IP telephone configuration and its functionalities.

102 © IITelecom, 2004 101 IIT 101 H.225 (RAS) - Registration Admission Status TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931Signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 H.245 of control signaling

103 © IITelecom, 2004 102 IIT 101 Call set-up stages Discovery Registration Call Initialization Call Negotiation Channel establishment for the transfer Exchange of information Call termination To which address must I call? I am Bob, do I have the permission to call? I would like to speak to Joe. Here are my reception and transmission capacity Joe was reached. Hello! How are you? Hello, see you soon!

104 © IITelecom, 2004 103 IIT 101 H.225 Messages (RAS) Remote access terminal Access Server PSTN GatekeeperGateway PSTN H.225 messages RAS Process Seek for a gatekeeper Registration to a gatekeeper Call control admission Localization of the called terminal Change of bandwidth IP network

105 © IITelecom, 2004 104 IIT 101 Questions? ?

106 © IITelecom, 2004 105 IIT 101 Activity 4.3 - H.323, Signaling and Control The H.323 standard Signaling and Control –H.225/Q.931 –H.245 –TCP/UDP ports Attribution

107 © IITelecom, 2004 106 IIT 101 H.225 Signaling derived from Q.931 and H.245 TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225 Q.931 Signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 H.245 of control signaling

108 © IITelecom, 2004 107 IIT 101 Laboratory activity 4.3 H.225 and H.245 protocol analysis TCP/UDP ports attribution

109 © IITelecom, 2004 108 IIT 101 Laboratory activity 4.3 (cont'd) H.225 and H.245 protocol analysis TCP/UDP port attribution

110 © IITelecom, 2004 109 IIT 101 Laboratory activity 4.3 (cont'd) H.225 and H.245 protocol analysis TCP/UDP port attribution

111 © IITelecom, 2004 110 IIT 101 Laboratory activity 4.3 Packet IP address source IP address destination TCP/UDP port source TCP/UDP port destination Transport protocol Type of message 1 2 3 4 6 7 8 9 10 11 12 13

112 © IITelecom, 2004 111 IIT 101 Laboratory activity 4.3 Packet Source IP address Destination IP address TCP/UDP port source TCP/UDP port destination Transported protocol Type of message 14 15 16 17 18 19 20 21 22 23 24 25

113 © IITelecom, 2004 112 IIT 101 Laboratory activity 4.3 Packet IP address source IP address destination TCP/UDP port source TCP/UDP port destination Transported protocol Type of message 26 27 28 29 30 31 32 33 34 35 36 37

114 © IITelecom, 2004 113 IIT 101 Laboratory activity 4.3 Packet IP address source IP address destination TCP/UDP port source TCP/UDP port destination Transported protocol Type of message n - 11 n - 10 n - 9 n - 8 n - 7 n - 6 n - 5 n - 4 n - 3 n - 2 n - 1 n n = last packet

115 © IITelecom, 2004 114 IIT 101 Laboratory activity 4.3 (cont'd) H.225 and H.245 protocol analysis TCP/UDP port attribution

116 © IITelecom, 2004 115 IIT 101 H.225 signaling TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931 Signaling Signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 H.245 of control signaling

117 © IITelecom, 2004 116 IIT 101 H.225 signaling derived from Q.931 IP network Terminal remote access Access Server PSTN GatekeeperGateway PSTN H.225 signaling Initialization Call set-up Warning Connection (ports attribution for H.245) Command for end of session Release of connection

118 © IITelecom, 2004 117 IIT 101 Signaling Model Determine which protocols go trough the gatekeeper and which ones pass directly between the two termination points. Missing gatekeeper from the H.323 network –exchanged directly between the terminals or gateways Gatekeeper present in the H.323 network –exchanged directly between the terminals or gateways (Direct Call Signaling) –exchanged between the terminals or gateways after having passed trough the gatekeeper (Gatekeeper-Routed Call Signaling) The method is chosen by the gatekeeper during the admission.

119 © IITelecom, 2004 118 IIT 101 Direct call signaling Signaling H.225 (Q.931) Address translation Admission control Bandwidthontrol Bandwidth control GK TerminalTerminal Address translation Admission control Bandwidthontrol Bandwidth control Gateway

120 © IITelecom, 2004 119 IIT 101 H.245 Signaling of control TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225Q.931Signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 H.245 of control signaling

121 © IITelecom, 2004 120 IIT 101 H.245 messages IP network Remote access terminal Access Server PSTN GatekeeperRouter Gateway LAN PSTN H.245 messages Exchange of capacities Opening and closing the logical channels used for the transfer of information, voice, video and data. –Attribution of the RTP and RTCP ports

122 © IITelecom, 2004 121 IIT 101 Logical channels Each logical channel uses a different socket (IP address IP + logical port = socket) Terminal 1 H.323 Terminal 2 H.323 Control H.245 Audio Video 0 5 6 7 8 0 1 2 3 4 Logics Ports

123 © IITelecom, 2004 122 IIT 101 Transport of information with RTP and RTCP TCP UDP IP IP Connection (IEEE 802.3) RTP/RTCP H.261H.263 AudioVideo T.120 H.225RAS H.225 Q.931 Signaling Control and management of the calls Data G.711G.723.1G.726G.728G.729 H.245 of control signaling

124 © IITelecom, 2004 123 IIT 101 H.323v2 New functionalities added to the Gatekeeper –Security, authentification, encryption –Establishment of faster call (Fast call setup) QoS Improvement, thanks to RSVP Additional services –Call transfer –Call forwarding

125 © IITelecom, 2004 124 IIT 101 Set-up of a H.323v1 call Gatekeeper Terminal 1 H.323 Terminal 2 H.323 (1) ARQ (2) ACF (3) Initiation (4) Establishment of the call RASH.225H.245 (6) ACF (5) ARQ (7) Warning (8) Connection (9) Acc. logical opening channel (10) Acc. logical opening channel Logical channel for flows of information

126 © IITelecom, 2004 125 IIT 101 Set-up of a H.323v2 call Gatekeeper Terminal 1 H.323 Terminal 2 H.323 (1) ARQ (2) ACF (3) Initialization with faststart OLC * RASH.225 (5) ACF (4) ARQ (6) Connection with faststart OLC * Logical channel for flows of information Faststart OLC: Faster establishment of call with direct opening of the logical channel

127 © IITelecom, 2004 126 IIT 101 Station management for the network GatekeeperGateway MCU and others SNMP/ CMIP H.323v2: network management The Gatekeeper can provide centralized management.

128 © IITelecom, 2004 127 IIT 101 H.323 v3 Signaling (H.225, H.245) using UDP rather than TCP Address resolution for inter and intra domains Addition of supplemental services –Call on hold –Call park and pickup –Message Waiting Signaling –Call waiting

129 © IITelecom, 2004 128 IIT 101 H.323 v4 Possibility for the terminal to select a gateway H.323 URL (h323:xyz@iitelecom.com) Identification of the caller ID

130 © IITelecom, 2004 129 IIT 101 Quiz Associate the following functions with the suitable component: Support multimedia applications ___ Translation of the telephone number to IP address ___ Conversion of the transmission format ___ Admission control ___ Conversion of the signaling protocols ___ Support multimedia conferences ___ a) Terminal b) Gateway c) Gatekeeper d) MCU

131 © IITelecom, 2004 130 IIT 101 Quiz Associate the following functions the suitable protocol: Establish connection between two terminals ___ Control registration___ Opening of the logical channels (video, voice) ___ Admission control ___ Functionalities exchange ___ Termination of connection ___ a) H.225 (RAS) b) H.225 (Q.931) c) H.245

132 © IITelecom, 2004 131 IIT 101 Questions? ?

133 © IITelecom, 2004 132 IIT 101 Activity 4.4 SIP – Session Initiation Protocol SIP Standard To describe the SIP standard, addressing and components To explain the various stages carried out during the call set-up in a SIP network

134 © IITelecom, 2004 133 IIT 101 Session Initiation Protocol (SIP) Signaling protocol for multimedia applications Independent of sub layer protocols (TCP, UDP) Standard developed by the IETF (MMUSIC working group) - RFC 2543 SIP works in various phases of the call –Localization of the corresponding terminal –Analyze recipient profile and resources –Negotiation of the media type and of the communication parameters –Availability of the correspondent –Call set-up and call follow-up SIP uses several existing protocols –Message format (HTTP 1.1) –Media negotiation (SDP - Session Description Protocol), –Media (RTP) –Name resolution and mobility (DNS and DHCP) –Applications encoding (MIME)

135 © IITelecom, 2004 134 IIT 101 SIP Specific Protocols Physical IP TCP/UDP RTP/RTCP G.711 G.729 G.723.1 … H.261 H.263 Audio Video SIP SDP Signaling

136 © IITelecom, 2004 135 IIT 101 SIP Addressing SIP Addresses are identified by URL, in the form user@host user = name or telephone number host = domain name or IP addresses Examples sip:xyz@iitelecom.com sip:xyz@192.168.10.1 sip:5141234567@iitelecom.com; user=phone

137 © IITelecom, 2004 136 IIT 101 SIP Components User Agent An end user application initiating, receiving and terminating a call Proxy Server An application server conveying the requests on behalf of the end user application The request is processed and sent to the destination (called person) or to another server Redirect Server An application server determining the destination address (To:) and returning it to the end user application

138 © IITelecom, 2004 137 IIT 101 SIP Components (cont'd) Localization Server Used by the Proxy Server and Redirect Server to obtain the location of the called user (one or more addresses) Registration Server Accept registration requests from the client applications Generally, the service is offered by the Proxy Server or Redirect Server DNS Server Used to locate the Proxy Server or Redirect Server

139 © IITelecom, 2004 138 IIT 101 SIP components and services SIP Servers and services Proxy SIP Server RegistrarRedirect Location Database Register I am here Redirect Here is the address Locate Where this name is or tel. number… INVITE I want to speak with another agent. Proxy INVITES I will call it for you. SIP User Agents GW SIP SIP User Agents

140 © IITelecom, 2004 139 IIT 101 INVITE sip:pierre@192.168.1.31 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060 Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 INVITES Content-type: application/sdp Content-Length: 98 v = (protocol version) O = (owner/creator and session to identify) C = (session information) T = (time the session is active) m = (media name and address transport) Example: INVITE SDP - Session Description Protocol SDP defines the conversation parameters on the client application (User Agent) SDP transmits information required to establish a multimedia session SDP is similar to H.245 in H.323 functions SDP contains the following parameters: –Medium to be used (codec, sampling rate) –Destination (IP address and port number) –Session name –Session duration –Contact –etc…

141 © IITelecom, 2004 140 IIT 101 SIP session set-up Contents Signaling INVITE 100 Trying 180 Ringing 200 OK ACK Logical opening of RTP channel Bye 200 OK Signaling Logical opening of RTCP channel Logical opening of RTP channel Logical opening of RTCP channel Each end knows the other one IP address Media (UDP)

142 © IITelecom, 2004 141 IIT 101 SDP Messages in a SIP session INVITE 100 Trying 180 Ringing 200 OK ACK INVITE sip:pierre@192.168.1.31 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060 Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 INVITES Content-type: application/sdp Content-Length: 98 v=0 o=marie 3123 121231 IN IP4 192.190.132.20 c=IN IP4 192.190.132.20 m=audio 5004 RTP/AVP 0 SIP/2.0 200 OK Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 INVITES Content-type: application/sdp Content-Length: 98 v=0 o=pierre 5664 456456 IP IP4 192.190.132.31 c=IN IP4 192.190.132.31 m=audio5004 RTP/AVP 0 ACK sip:pierre@192.190.132.31 SIP/2.0 Via: SIP/2.0/UDP 192.190.132.20:5060 Call-ID: 141710@192.190.132.20 From: sip: marie@192.190.132.20 To: sip:pierre@192.190.132.31 Cseq 1 ACK Marie 192.168.1.20 Pierre 192.168.1.31 Each end knows the other one IP address

143 © IITelecom, 2004 142 IIT 101 SIP message types SIP is modeled on HTTP Use same syntax and semantics as HTTP –Request Method (INVITE, ACK, BYE, etc.) Header (Accept, Contact, etc.) –Answer Status code (200 OK, 180 Ringing, etc.) Header (Content-type, Content-encoding, etc.) SIP Methods INVITE Initiate a call by inviting a user to take part in a session. ACK Confirm that the client received a final response to a request INVITES. BYE Indicate the end of the call. CANCEL Cancel a request. REGISTER To register the User Agent. OPTIONS Used to know the capacities of the server. SIP Answers 1xx - Informational Messages. 2xx - Successful Responses. 3xx - Redirection Responses. 4xx - Request Failure Responses. 5xx - Server Failure Responses. 6xx - Global Failure Responses.

144 © IITelecom, 2004 143 IIT 101 SIP in Proxy mode marie@iitelecom.com INVITE pierre@uqam.ca From: marie@iitelecom.com 1 Pierre?Pierre@stanford.edu Location Server pierre@stanford.edu INVITE pierre@stanford.edu From: marie@iitelecom.com 200 OK ACK 23 4 5 8 6 7 Established session Proxy Server

145 © IITelecom, 2004 144 IIT 101 SIP in Redirect mode marie@iitelecom.com INVITE pierre@uqam.ca From: marie@iitelecom.com 1 Pierre?Pierre@stanford.edu Location server pierre@stanford.edu INVITE pierre@stanford.edu From: marie@iitelecom.com 302 Moved Contact: pierre@stanford.edu ACK 23 6 200 OK 7 ACK 8 4 5 Established session Redirect Server

146 © IITelecom, 2004 145 IIT 101 SIP call example Call forward busy from B to C INVITE 100 Trying 486 Busy INVITE ACK INVITE 180 Ringing 200 OK 180 Ringing 200 OK ACK Established session ACK Proxy Server UA AUA BUA C

147 © IITelecom, 2004 146 IIT 101 SIP call example Call transfer from A to C Bye (also C) 100 Trying INVITE (req A) 180 Ringing 200 OK ACK Established session Proxy Server Bye (also C) 200 OK INVITE (req A) 200 OK 180 Ringing 200 OK ACK Established session UA A UA B UA C

148 © IITelecom, 2004 147 IIT 101 SIP References Columbia university Web site http://www.cs.columbia.edu/sip/ IETF SIP working group http://ietf.org/html.charters/sip-charter.html SIP forum http://www.sipforum.org Ubiquity Information Center : SIP center http://www.sipcenter.com

149 © IITelecom, 2004 148 IIT 101 Laboratory activity 4.4 Analysis of the SIP protocol.

150 © IITelecom, 2004 149 IIT 101 Activity 4.5 MGCP (Media Gateway Control Protocol) and Megaco (MEdia GAteway COntrol) To describe the MGCP/Megaco standard and its components To explain the various stages carried out during the call set- up in a MGCP/Megaco network To compare H.323, SIP and MGCP/Megaco standards

151 © IITelecom, 2004 150 IIT 101 New voice network architecture Standard-Based Packet Infrastructure Layer Standard-Based Open Call Control Layer Open Service Application Layer Standard Interface Switching Network DIGITAL Trunk Subsystem LineConcentrationLineConcentration Common Channel Signaling Complex Common Channel Signaling Complex AdministrationMaintenanceBillingAdministrationMaintenanceBilling TDM/ Switch Circuit Call Control Connection Control Features Call Control Connection Control Features Separation of the three architecture levels

152 © IITelecom, 2004 151 IIT 101 Protocol evolution SGCP(IETF) July 1998 BellcoreCisco MGCP 1.0 (IETF) October 1999 MEGACO(MGCP+)(IETF) November 2000 MDCP(IETF) December 1998 Lucent IPDC August 1998 Level 3 SGCP - Simple Gateway Control Protocol MGCP - Media Gateway Control Protocol IPDC - IP Device Control MDCP - Media Device Control Protocol MEGACO - Media Gateway Control

153 © IITelecom, 2004 152 IIT 101 MGCP (Media Gateway Control Protocol) Defined by the IETF in document RFC 2705 (version 1.0), in October 1999 Version 0.1 (October 98) is the result of the fusion of SGCP 1.2 (Telcordia) with IPDC (Level 3) Media Gateways (MGs) are controlled by the Media Gateway Controllers (MGCs) in an master/slave architecture –Voice call set-up only (no multimedia) MGCP architecture is divided into three layers –Application (optional): Applications Server –Call control : Media Gateway Controller, Call Agent –Connectivity: Media Gateways, Routers, LAN, Switches Uses the session description protocol (SDP) to describe the media capabilities— like SIP and MEGACO/H.248 Can be deployed in a network in combination with other architectures (H.323, SIP)

154 © IITelecom, 2004 153 IIT 101 MGCP structure and protocols SS7 Network SS7 Network PSTN Media H.323, SIP, ISUP SCTP MGCP SCTP MGC Media gateway Media gateway SCTP: Stream Control Transmission Protocol

155 © IITelecom, 2004 154 IIT 101 MGCP Components Media Gateway (MG) Various types: residential, access, trunking Adapts the content format coming from a network type to another network type format Must be able to convert the audio into full-duplex Media Gateway Controller (MGC) Also called Call Agent or Softswitch Provides a centralized control for the gateways Responsible for call signaling (set-up, modify, terminate) Uses UDP

156 © IITelecom, 2004 155 IIT 101 MGCP Primitives NotificationRequest (RQNT) –Inform the gateway to supervise specific events Notify (NTFY) –Inform the MGC when the required events take place CreateConnection (CRCX) –Create a connection towards an Endpoint inside the gateway ModifyConnection (MDCX) –Change the parameters associated with an already established connection DeleteConnection (DLCX) –Remove an existing connection — an ACK returns the call statistics AuditEndPoint (AUEP) and AuditConnection (AUCX) –Check an endpoint status and any associated connection RestartIngProgresss (RSIP) –Inform the MGC that an endpoint (or a group of endpoints) is out-of- service

157 © IITelecom, 2004 156 IIT 101 MGCP call set-up MGC Trunking gateway Residential gateway IP Network Voice circuits MGCP Q.931/SS7 RTP SS7 Links User B User A STP Switch

158 © IITelecom, 2004 157 IIT 101 Megaco/H.248 Defined by the IETF and ITU (RFC 3015) in November 2000 –The H.248 recommendation was published in February 2001 Called Megaco by the IETF and H.248 by the ITU Megaco is a control protocol between the Media Gateway and the Media Gateway Controller (same as MGCP) Multimedia applications support –MGCP supports only voice

159 © IITelecom, 2004 158 IIT 101 Megaco/H.248 structure IP network PSTN Media Gateway Media Gateway SS7/IP Gateway Gateway Media Controller (Softswitch) Switch PSTN IP telephone SS7Network Megaco/ H.248 Sigtran

160 © IITelecom, 2004 159 IIT 101 MGCP vs. Megaco MGCP is the first developed –Some products already on the market MGCP is simpler –Support only voice –Megaco developed for multimedia application MGCP is the “de facto” standard Megaco is the upcoming standard

161 © IITelecom, 2004 160 IIT 101 Summary of VoIP protocols SIP 0%5%15%24% Megaco 1%8%20%28% MGCP 10%15%20%14% H.323 89%73%45%35% 2001200220032004 Current tendencies according to Insight Research, Jan 2001 H.323 is still the most used protocol MGCP is accepted by the Softswitch Manufacturers SIP is increasingly popular; industry sees much interest there; Windows XP includes SIP client (Messenger) Megaco is increasingly accepted SIP and Megaco are chosen by 3GPP 3GPP: 3rd Generation Partnership Project

162 © IITelecom, 2004 161 IIT 101 SIP vs. H.323 vs. MGCP/Megaco H.323 SIP Source ITU IETF Transport Primarily TCP Primarily UDP Encoding Binary ASN.1 ASCII Text ServicesTelephony Multimedia H.323 SIP Source ITU IETF Transport Primarily TCP Primarily UDP Encoding Binary ASN.1 ASCII Text ServicesTelephony Multimedia Service H.323v1 H323v2 H.323v3 SIP Call Transfer No Yes Yes Yes Call Redirection No Yes Yes Yes Call standby/on-hold No Yes Yes Yes Conference NoYes Yes Yes Click-to-dial No Yes Yes Yes Call set-up 6-7 RT 3-4 RT 2.5 RT 1.5 RT Service H.323v1 H323v2 H.323v3 SIP Call Transfer No Yes Yes Yes Call Redirection No Yes Yes Yes Call standby/on-hold No Yes Yes Yes Conference NoYes Yes Yes Click-to-dial No Yes Yes Yes Call set-up 6-7 RT 3-4 RT 2.5 RT 1.5 RT SIP H.323 MGCP/Megaco Complexity lowhigh high SS7 Compatibility low low high Cost low high moderated SIP H.323 MGCP/Megaco Complexity lowhigh high SS7 Compatibility low low high Cost low high moderated

163 © IITelecom, 2004 162 IIT 101 Other Specifications Related To VoIP PINT (PSTN and Internet Internetworking) –Allow an IP user to have access to the PSTN network (example: Click-to-dial) SPIRITS (Service in the PSTN/IN Requesting Internet Service) –Allow a user of the PSTN network to have access to IP services (example: Internet Call Waiting) ENUM (Telephone Number Mapping) –Translation of the telephone number in URL or IP addresses TRIP (Telephony Routing over IP) –Routing of the VoIP calls Sigtran (Signaling Transport) –Transport of SS7 signaling on IP

164 © IITelecom, 2004 163 IIT 101 References Megaco from the IETF –http://www.ietf.org/html.charters/megaco-charter.html MGCP 1.0 (RFC 2705) –http://ftp. ietf.org/rfc/rfc2705.txt?number=2705 Documentation on Megaco –ftp://ftp.isi.edu/in-notes/rfc3015.txt Archives –http://standards.nortelnetworks.com/archives/megaco.html Softswitch Consortium –http://www.softswitch.org Mailing list –listserv@standards.nortelnetworks.com subscribe megaco

165 © IITelecom, 2004 164 IIT 101 Questions? ?

166 © IITelecom, 2004 165 IIT 101 LU 5 – Implementation considerations Training objectives: The participant will be able to: –Identify the different critical points to consider when considering a VoIP implementation

167 © IITelecom, 2004 166 IIT 101 Corporate network Number of offices and geographical distribution Voice transmission –Number of stations –Centrex, PBX, Keys System Obsolescence Financial amortization –PSTN Links (bandwidth/costs) –Inter branches Links (bandwidth/costs) Data transmission network –Structure and components –Internet Links (bandwidth/costs) –Inter offices Links (bandwidth/costs)

168 © IITelecom, 2004 167 IIT 101 Needs and required functionalities Growth External/internal Call processing Added value services Call centers –Inbound –Outbound Integration of voice and data –Message centralization –Personal assistant –Internet call centers V PSTN V 1 3 4 2 Personal assistant Calendar Softswitch IP network

169 © IITelecom, 2004 168 IIT 101 Potential solutions Traditional voice circuit switch system Voice over IP system Hybrid configuration –Site by site migration –Partial migration/keep existing equipment Manufacturer choice

170 © IITelecom, 2004 169 IIT 101 Decision factors QoS Reliability and robustness Supported functionalities and applications Security Costs

171 © IITelecom, 2004 170 IIT 101 QoS Network scaling Sampling size –Packet size –Bandwidth optimization Compression type –Locally –Tie lines Priority mechanisms –Locally –Tie lines Effects on other decision factors –Bandwidth costs and use –Data transmission quality and effectiveness

172 © IITelecom, 2004 171 IIT 101 Reliability and robustness Redundancy –Power supply unit –Power over Ethernet –CPU redundancy –Fall back on the PSTN Effect on the costs of the equipment MTBF/MTTR –Technology maturity –Difference in technology Effect on the maintenance costs CPU: Central Processing Unit MTBF: Mean time Between Failure (average time between breakdowns) MTTR: Mean time To Repair (Mean repair time)

173 © IITelecom, 2004 172 IIT 101 Supported functionalities and applications Traditional applications Technology Maturity –Call processing functionality –Call center management –… VS. New applications Multimedia/ voice & data integration Personal assistant Internet call centers …

174 © IITelecom, 2004 173 IIT 101 Security/confidentiality Conventional network Point-to-point Circuit switch –Network security –Additional security necessary for specific applications Encryption IP network Broadcast environment Divided bandwidth –Security at risk –Security mechanisms External oFirewall oVPN oEncryptions Internal oEncryption –Effects on costs User-friendliness VPN: Virtual Private Network

175 © IITelecom, 2004 174 IIT 101 Solutions costs Equipment Cost Equipment required for the new solution implementation, considering: –Protection of the investment and amortization –Upgrade of the existing equipment PBX, key systems Routers, switches oQoS Support oInterworking with the WAN (bandwidth, protocol…) –Redundancy –Security Wiring cost –1 cable for the telephone and the PC Effect on reliability

176 © IITelecom, 2004 175 IIT 101 Solutions costs (cont’d) Bandwidth cost Tie lines / inter offices links –Dimensioning voice and data Voice QoS Effects on data QoS Access to the PSTN and long distance calls expenses Maintenance costs Service contract Respect of the MTBF and MTTR Data and voice network

177 © IITelecom, 2004 176 IIT 101 Internet use in our private network Today Best effort network –No control on the bandwidth –No control on the packet size –No QoS mechanism Evolution towards different QoS on the Internet Effects on network architecture and technology Effects on the price –Single tariff –Billing

178 © IITelecom, 2004 177 IIT 101 LU 6 - Applications Training objectives : The participant will be able to: –Describe various services and applications offered in VoIP network and to describe the advantages

179 © IITelecom, 2004 178 IIT 101 Where can we use Voice over IP? Today –Tie line replacement between PBX –Long distance call “Internet Telephony Provider Service” –Off Premise Extension (OPX) –Replacement of key systems by a router “Router Key System” –IP telephone system for small companies (< 100 users) Tomorrow –Call centers accessible by Internet “Virtual call centers” –Integration of voice and data applications –Collaboration platform (fax, electronic mail, voice messages) –Unified messaging

180 © IITelecom, 2004 179 IIT 101 Voice over IP on an Intranet Gateway Gateway MontrealToronto PBXPBX DS1 RouterRouter LAN LAN PSTNPSTN

181 © IITelecom, 2004 180 IIT 101 Internet Telephony Service Provider (ITSP) Private IP network Gateway PSTN Gateway PSTN 1-514 1-416 ITSP Montreal ITSP Toronto RouterRouter

182 © IITelecom, 2004 181 IIT 101 Internet Telephony Service Provider (ITSP) Private IP network

183 © IITelecom, 2004 182 IIT 101 Main office Regional office Telecommuter V V V PSTN Network IP V Regional o ffice Softswitch IP-PBX Only one network for voice, data & video No geographical limit Integration with other Web applications Simplified mobility with DHCP Solution based on the standards vs. solution based on PBX manufacturer

184 © IITelecom, 2004 183 IIT 101 Main office Telecommuter VV Network IP V Regional o ffice PSTN CO Softswitch Media Gateway Centrex IP Same advantages as the IP-PBX No acquisition cost for the technology

185 © IITelecom, 2004 184 IIT 101 Click-to-dial (CTD) Called No yyy CTD User No xxx Navigator WWW Web server 1. CTD requires to call No yyy 2. App checks the coordinates Of the CTD user (IP addresse) 3. App requests the CA to establish the call between No xxx and yyy 4. CA calls xxx 5. When xxx answers, CA calls yyy 6. When yyy answers, CA establishes the call between xxx and yyy User Database CTD (Profiles) Application Server CTD Softswitch

186 © IITelecom, 2004 185 IIT 101 Personal assistant V PSTN IP Network V 1 3 4 2 Personal assistant Calendar Softswitch IP telephone Softphone Composition of the call number Forward call to the personal assistant The personal assistant checks and follows the rules laid down by the user

187 © IITelecom, 2004 186 IIT 101 Call centers and Internet Internet Hello, can I help you?

188 © IITelecom, 2004 187 IIT 101 Centralization of messages (fax, email, voice message) IP network Fax Mail Message Messages Server email Fax PBX

189 © IITelecom, 2004 188 IIT 101 Questions? ?

190 © IITelecom, 2004 189 IIT 101 Appendix - Acronym list

191 © IITelecom, 2004 190 IIT 101 Acronym list 3GPP Third Generation Partnership Project ABR Available Bit Rate ACFAdmission Confirm ACM Address Complete Message ADPCMAdaptive Differential Pulses Code Modulation ADSMAsymmetric Digital Subscriber Line AF Assured Forwarding ALI Automatic Location Identifier ANI Automatic Number Identifier ANM Answer Message ANSIAmerican National Standard Institute ARP Address Resolution Protocol ARQAdmission Request ASCIIAmerican Standard Code for Information Interchange ASNAbstract Symbol Notation ATM Asynchronous Transfer Mode BGP Border Gateway Protocol bpsBits Per Second CACall Agent CAC Connection Admission Control CANCampus Area Network CAR Committed Access Rate CBWFQ Class-Based Weighted Fair Queuing CGICommon Gateway Interface CIC Circuit Identification Code CIR Committed Information Rate CM Cable Modem CMIPCommon Management Information Protocol CMTS Cable Modem Termination System COCentral Office CoPS Common open Policy Server CoS Class of Service CPL Common Programming Language CPUCentral Processing Unit CRTP Compressed Real-Time Protocol CS-ACELPConjugate Structure Adaptive Code Excited Linear Prediction CSRCContributing Source CTDClick-To-Dial CTIComputer Telephony Integration dBDecibel dBmDecibel relative to 1 milliwatt DHCP Dynamic Host Configuration Protocol DiffServ Differentiated Services DNSDomain Name Server DOCSIS Data Over Cable Interface Specifications DS Differentiated Services DSCP Differentiated Services Code Point EF Expedited Forwarding EGP Exterior Gateway Protocol EIGRP Enhanced Interior Gateway Routing Protocol

192 © IITelecom, 2004 191 IIT 101 Acronym list (cont'd) ERP Enterprise Resource Planning ETSI European Telecomm Standards Institute FCSFrame Check Sequence FEC Forward Equivalence Class FRF Frame Relay Forum FRTS Frame Relay Traffic Shaping FTP File Transfer Protocol GKGatekeeper GTS Generic Traffic Shaping HFC Hybrid Fiber Coax HTTP Hypertext Transfer Protocol HzHertz IAM Initial Address Message ICMP Internet Control Message Protocol IDCInternational Dated Corporation IEEEInstitute of Electrical and Electronic Engineers IETF Internet Engineering Task Force IGP Interior Gateway Protocol IHLInternet Header Length INAP Intelligent Network Application Profile IntServ Integrated Services IP Internet Protocol IPv4IP version 4 IPv6IP version 6 IPDC IP Device Control ISDNIntegrated Services Digital Network ISUP Integrated Services Digital Network User Part ITSP Internet Telephony Service Provider ITUInternational Telecommunication Union ITU-TITU - Telecom JPEGJoint Photographic Expert Group kbpsKilobits Per Second LAN Local Area Network LDP Label Distribution Protocol LD-CELPLow-Delay Code Excited Linear Prediction LER Label Edge Router LIB Label Information Base LLCLogical Link Control LLQ Low Latency Queuing LSLocation Server LSR Label Switch Router MACMedia Access Control MANMetropolitan Area Network MAP Mobile Application Part MCR Minimum Cell Rate MCU Multipoint Control Unit MDCP Media Device Control Protocol Megaco Media Gateway Control MGMedia Gateway MGCMedia Gateway Controller

193 © IITelecom, 2004 192 IIT 101 Acronym list (cont'd) MGCP Media Gateway Control protocol MIB Management Information Base MIMEMultipurpose Internet Mail Extension MIPS Million Instruction Per Second MMUSICMultiparty Multimedia Session Control MOSMean Opinion Score MPLS Multi Protocol Label Switching MP-MLQMultipulse Multilevel Quantization msMillisecond MTBFMean Time Between Failure MTP Message Transfer Part MTTRMean Time To Repair NAT Network Address Translation NCS Network-based Control Signaling NFS Network File System NNTP Network News Transfer Protocol OLCOpen Logical Channel OMAP Operational, Management and Admin Process OPXOff Premise Extension OSPF Open Shortest Path First OSIOpen System Interconnection PANPersonal Area Network PBH Per-Hop Behavior PBXPrivate Branch Exchange PCR Peak Cell Rate PCMPulse Code Modulation PHB Per-Hop Behavior POP3 Post Office Protocol version 3 POTS Plain Old Telephone Service PPP Point-to-Point Protocol PQ Priority Queuing PSProxy Server PSTNPublic Switched Telephone Network QoS Quality of Service RARPReverse ARP RAS Registration Admission Status RFC Request For Comment RMON Remote Monitoring RPC Remote Procedure Call RSRegistration Server RSVP Resource Reservation Protocol RTCPReal-time Control Protocol RTP Real-time Protocol SCCPSignalling Connection Control Part SCP Signal Control Point SCTP Stream Control Transmission Protocol SDHSynchronous Digital Hierarchy SDP Session Description Protocol SGCP Simple Gateway Control Protocol SIGTRANSignalling Transport

194 © IITelecom, 2004 193 IIT 101 Acronym list (cont'd) SIP Session Initiation Protocol SLA Service Level Agreement SLIPSerial Line Internet Protocol SMTP Simple Mail Transfer Protocol SNMP Simple Network Management Protocol SONETSynchronous Optical Network SS7Signaling System 7 SSP Switching Service Point SSRCSynchronisation Source STHMLSafe Hypertext Transfer Protocol STP Signal Transfer Point SVC Switched Virtual Circuit TCAP Transaction Capabilities Application Part TCP Transport Control Protocol TDMTime Division Multiplexing TFTP Trivial File Transfer Protocol TIATelecommunication Industry Association ToS Type of Service TTLTime To Live UAUser Agent UDP User Datagram Protocol URL Uniform Resource Locator VAD Voice Activation Detection VBR-rtVariable Bit Rate - Real-time VoIP Voice over IP VPNVirtual Private Network WANWide Area Network WFQ Weighted Fair Queuing WRED Weighted Random Early Drop For additional acronyms, http://www.csrstds.com


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