VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet

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1 VOICE over IP H.323 Advanced Computer Network SS2005 Presenter : Vu Thi Anh Nguyet 1

2 Outlines 1. Introduction 2. QoS in VoIP 3. H Signalling in VoIP 5. Conclusions 2

3 1. Introduction to VoIP Voice over IP the transmission of digitalized voice over packet-switched IP networks PSTN Class 5 City B City A V IP Network V Class 5 PSTN 3

4 VoIP Advantages Lower costs per call Lower infrastructure costs New advanced features 4

5 VoIP Packet Format Link layer size vary per media Using UDP protocol without TCP Voice carried using the RTP protocol Payload size depend on codec type 5

6 2. Quality of Service (QoS ) QoS in a packet network is characterized by the main parameters as: - Bandwidth - Delay - Packet loss 6

7 VoIP Bandwidth Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size) PPS = (codec bit rate) / (voice payload size) Bandwidth = total packet size * PPS 7

8 Example: VoIP Bandwidth (cont.) A G.729 call (8 Kbps codec bit rate) with crtp and the default 20 bytes of voice payload requires: Total packet size (bytes) = (MP header of 6 bytes) + (compressed header of 2 bytes) + (voice payload of 20 bytes) = 28 bytes Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps (160 bits = 20 bytes (default voice payload) * 8 bits per byte Bandwidth per call IP/UDP/RTP = voice packet size (224 bits) * 50 pps = 11.2 Kbps 8

9 Delay Voice Path Loss + Delay CODEC Packetization Output queuing Access (up) link transmission Backbone network transmission Access (down) link transmission Input queuing Jitter buffer CODEC 9

10 Fixed Delay Components (cont.) Propagation Delay Serialization Delay Buffer to Serial Link Processing Delay Propagation 6 microseconds per kilometer Processing - Coding / compression - Decoding / decompression - Packetization Serialization 10

11 Variable Delay Components (cont.) Queuing Delay Queuing Delay Queuing Delay Jitter Buffer Queuing delay Jitter buffer 11

12 Jitter Sender Network Receiver Variation of interpacket arrival time A B C A B C D 1 D 2 = D 1 D 3 = D 2 Sender t Receives t 12

13 Total Delay Time Total delay for above example : 167 ms ITU-T: <150ms : not detectable = ms : Acceptatble quality > ms : unacceptable quality 13

14 Packet Loss missing packet G.729 vocoder algorithm The total of number of lost packets can be accepted 5% 14

15 QoS Remarks VoIP frames have to traverse an IP network which is unreliable. Frames may be dropped as a result of network congestion or data corruption. For real-time traffic like voice, retransmission of lost frames at the transport layer is not practical because of the additional delays. Voice terminals have to deal with missing voice samples, also referred to as frame erasures. 15

16 3. H.323 Standards H.323 is a standard that defines how voice and video devices can communicate. It specifies both signaling characteristics and host-to-host communication protocols 16

17 H.323 Standards (cont.) The H.323 standard consists of the following components and protocols: Protocol: Feature: H.225 Call Signalling H.245 Media Control G.711,G.722, G.723,G.728,G.729 Audio Codes H.261, H.263 Video Codes T.120 Data Sharing RTP/RTCP Media Transport 17

18 H.323 Components GK H.323 MCU e H.323 Gatekeeper Packet Network H.323 Terminal V H.323 Gateway PSTN ISDN V.70 Terminal H.324 Terminal Speech Terminal H.320 Terminal Speech Terminal 18

19 Gateway The H.323 gateway reflects the characteristics of a Switches Circuit Network (SCN) endpoint and H.323 endpoint. It converts voice and fax calls, in real time, between the PSTN and an IP network. Gateways work as an H323 terminal. Gateways are not needed unless the interconnection with the PSTN is required. 19

20 Gatekeeper An optional H.323 Component Defines H.323 Zone Provides Centralized Call Control Mandatory and Optional Services 20

21 Gatekeeper Mandatory Services (cont.) Address Translation Translates H.323 aliases (e.g. or E.164 addresses (standard phone numbers) into IP transport addresses (e.g port 1720) Admissions Control Authorizes access to the H.323 network Bandwidth Control Manages endpoint bandwidth requirements Zone Management Provides the above functions to all terminals, gateways, and MCUs that register to it 21

22 Gatekeeper Optional Services (cont.) Call control signaling Gatekeeper Routed Call Signaling (GKRCS) Call authorization Restrict certain terminals, gateways, time of day Bandwidth management Reject admission if bandwidth is not available Call management Services include maintaining an active call list that use to indicate busy terminals. 22

23 4. H. 323 Signaling Admission Request Admission Confirm H.225 (UDP) RAS V Gatekeeper V Setup Alerting / Connect H.225 (TCP) Q.931 H.323 Gateway A Capabilities Exchange Open Logical Channel H.245 (TCP) V Open Logical Channel Acknowledge H.323 Gateway B RTP Stream RTP Stream RTCP Stream Media (UDP) 23

24 RAS Messages RAS channel is established between endpoints and Gatekeeper across an IP network. RAS channel is opend before any other channels which are established. RAS messages are carried by the UDP connection, perform registration, admission, bandwidth changes, etc. 24

25 RAS Messages (cont.) GRQ/GCF/GRJ (Discovery) GRQ : A multicast message sent by a GW looking for the GK GCF: The reply to a GW with it s transport address RRQ/RCF/RRJ (Registration) RRQ : sent from GW to GK RAS channel address RCF : sent from GK to GW to confirm a GW registration GRQ GCF/GRJ 25

26 RAS Messages (cont.) ARQ/ACF/ARJ (Admission) ARQ: The GK assigned terminal identifier The type of call (point to point) The call model that the terminal is willing to use (direct or GK routed) The destination address (Ex: E.164 address) ACF: The call model in use The transport address and port to use for Q.931 call signalling The allowed bandwidth for the call 26

27 RAS Messages (cont.)( DRQ/DCF/DRJ (Disconnect) Get rid of call state LRQ/LCF/LRJ (Location) Stateless name - IP address resolution Inter gatekeeper communication IRQ/IRR (Information Request) Ping during active calls Resource information for gateways BRQ/BCF/BRJ (Bandwidth) Ask for more/less bandwidth during call URQ/UCF/URJ (Unregistration) Get rid of registration state 27

28 RAS Message Exchange (cont.) Gatekeeper A LRQ Gatekeeper B LCF ACF IP Network ACF ARQ H.225 (Q.931) Setup ARQ H.225 (Q.931) Connect V Gateway A H.245 RTP V Gateway B Phone B Phone A 28

29 H.225 Call Control (ISDN Q.931) Setup Incoming call Call Proceeding Alerting Phone is ringing Connect Media cut through (used for billing) Release/Release Complete Tear down call 29

30 H.245 System Control Capabilities Exchange Exchange the capabilities between two entpoints entpoint s transmit and receive capabilities for audio, video, data. Master/ Slave Determination Open Logical Channel/Ack The channel is set up before the actual transmission to ensure the entpoints are ready and capable of receiving and decoding information. 30

31 SS7 Interconnect for Voice Gateway Call Setup PSTN/SS7 SC A GW A GK A GK B GW B SC B PSTN/SS7 1. IAM 2. Setup H.323 Phone A 3. Call proc 4. ARQ 5. LRQ 7. ACF 6. LCF 8. H225 Setup 9. ARQ SS7 19. ACM Q Alerting 14. H225 Call proc 17. H225 Alert 10. ACF 11. Setup 13. Call proc 16. Alerting 12. IAM 15. ACM 24. Connect 23. H225 Connect 21. Connect 22. Con. ACK 20. ANM Phone B 26. ANM 25. Con. ACK Connection Established 31

32 VoIP Configuration SC POP A GK PSTN N x E1 N x E1 outbound V VV GW E1 POP B PSTN N x E1 V V V GW SLT Router E1 N x E1 outbound SC GK PSTN N x E1 V V V GW GW HNI POP POP C N x E1 outbound 32

33 5. Conclusions One of the major motivations of developing VoIP networks is the cost benefit. QoS provides reduced delay and fewer dropped packets of voice traffic to ensure the good voice quality to customers. H.323 is probably the most important standard supporting packetized voice technology. However it is also the complex standard with many protocols. 33

34 References [1] J. Davidson, J. Peters, Voice over IP Fundamentals, Cisco Press, [2] O. Hersent, D. Gurle, J. P. Petit, IP Telephony Packet-based multimedia communications systems, Addison-Wesley, [3] L. L. Peterson, B. S. Davie, Computer Networks - A Systems Approach, 2nd Edition, Morgan Kaufmann, [4] J. Walrand, P. Varaiya, High-Performance Communication Networks, 2nd Edition, Morgan Kaufmann, [5] [6] [7] [8] Training documents, Cisco Advance Services, [9] Training documents, Cisco Voice over IP (CVOICE), [10] Paul J. Fong, Eric Knipp, Charles Riley, Configuring Cisco Voice over IP, Syngress,

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