Implementation of Embedded SIP-based VoIPv6 System

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1 Implementation of Embedded SIP-based VoIPv6 System Pre Dr. Han-Chieh Chao Department of Electronic Engineering National Ilan University

2 Drivers for IPv6 2

3 Drivers for IPv6 Mobile Networking O.S. and Applications Restoring an Environment for Innovation The Ubiquitous Internet Agriculture/Wildlife Transportation Medical Consumer and Services Manufacturing Services on the Edge Of the Network Government 3

4 Drivers for IPv6 (cont.) Improvement Available address space Multicast Configuration Addressing Aggregation Performance Header structure better organized No fragmentation No checksum Addressing architecture can be more aggregated Security IPv6 integrateds IPSec inside the protocol IPv4 needs to add IPsec IPSec (ESP) and IKE QoS IPv6 improves the performance Header and its processing Reduces jitter and processing delay New specific fields in the header 4

5 Network : IPv6 Let your product get certified IPv6 Ready Logo Phase 1 ( Silver ) IPv6 Ready Logo Phase 2 ( Gold ) 5

6 IPv6 Logo Committee (v6lc) the TAHI project (Japon) the University of New Hampshire (USA) ETSI IPv6 Plugtest (Europe) TTA (Korea) BII (China) NICI v6lab (Taiwan) IRISA/INRIA (France) 6

7 Drivers for VoIP ISP : xdsl providers are also the old incumbent Telo s Revenue streams from the existing PSTN PSTN replacement/upgrades with lifetimes of 20+ years Reluctant to invest money if voice traffic migrates from the PSTN to IP technology Deutsch Telecom Announced to January 2004 that by 2020 all telephone traffic will run over IP Consumer Cost savings Integrated data and voice networks Device interoperability using standards-based protocols Flexibility in deriving new services VoIP + IPv6 is better solution than VoIP + IPv4 Convenience Performance Security QoS 7

8 What is VoIP? Carrying voice or/and video conservations over Internet protocol packet networks IP Network 8

9 Why Use VoIP? Cost savings Integrated data and voice networks Device interoperability using standards-based protocols Flexibility in deriving new services 9

10 Traditional Voice Versus VoIP A traditional T1 can carry 24 telephone calls simultaneously T1 = 1544 kbps, DS0 = 64 kbps, 1544 / 64 = 24 DS0 per T1 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 10

11 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 PC Modem The Internet Modem PC 11

12 A Scenario of Applications IP-to-PSTN Gateway INTERNET PSTN Edge Router Edge Router IP Telephones IP Telephones 12

13 VoIP QoS Issue Delay Algorithmic Delay Processing Delay Network Delay Jitter Lost-Packet Compensation Echo Compensation CODECS 13

14 Delay Processing Delay Actual process of encoding and collecting the encoded samples into a packet for transmission over the packet network. The encoding delay is a function of both the processor execution time and the type of algorithm used Network Delay Physical medium and protocols used to transmit the voice data and by the buffers used to remove packet jitter on the receive side. Network delay is a function of the capacity of the links in the network and the processing. 14

15 Delay Causes Problems Echo Signal reflections of the speaker's voice from the far-end telephone equipment back into the speaker's ear. Round-trip delay becomes greater than 50 milliseconds. (ITU G.131) Talker overlap One talker stepping on the other talker's speech The one-way delay becomes greater than 250 milliseconds. (ITU G.114) 15

16 Jitter Variable delay caused by network traverses Removing jitter : buffering packets and waiting for the slowest packets to arrive in time to be played in the correct sequence. Causes additional delay 16

17 Lost-Packet Compensation Lost packets can be an even more severe problem, depending on the type of packet network that is being used. Interpolate for lost speech packets by replaying the last packet received during the interval. Send redundant information. Use a hybrid approach with a much lower bandwidth voice coder to provide redundant information. Avoiding and managing network congestion 17

18 Echo Compensation For network, usually Round-trip delay > 50 milliseconds ITU standard G.165 Defines performance requirements required for echo cancellers 18

19 CODEC 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 19

20 CODEC Characteristics 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 Compression Delay (ms) g711alaw PCM ms g711ulaw PCM ms g723ar53 ACELP ms g723ar63 MP-MLQ ms g723r53 ACELP ms g723r63 MP-MLQ ms g726r16 ADPCM ms g726r24 ADPCM ms g726r32 ADPCM ms g728 LD-CELP ms g729r8 CS-ACELP ms g729br8 CS-ACELP ms

21 Streaming Video Video Challenges Raw to Compression 21

22 Network Challenges Bandwidth : depend on Codec H.261 H.263 Display size 640 x x 240 Real-time Quality of Service (QoS) Packet loss Jitter Latency : Video needs much more computing latency then audio 22

23 Network Issues 23

24 Real-time Transport Protocol (RTP) Disadvantage of UDP Packets may be lost or out-of-sequence RTP: A Transport Protocol for Real-Time Application RFC 3550 RTP Real-Time Transport Protocol RTCP- RTP Control Protocol RTP over UDP A sequence number A time stamp ofr synchronized play-out Does not solve the problems: simply provides additional information 24

25 Real-time Transport Protocol (RTP) Media content type Talk spurts Payload identification Synchronization Loss detection Segmentation and reassembly V P X M Payload Timestamp Synchronization Source Identifier (SSRC) Payload Sequence Number 25

26 The RTP Header Sequence number A random number generated by the sender at the beginning of a se A random number generated by the sender at the beginning of a session ssion Incremented by one for each RTP packet Timestamp The receiver The receiver Synchronized play Synchronized play-out out Calculate the jitter Calculate the jitter Support silence suppression Support silence suppression The initial timestamp is a random number chosen by the sending application. Payload Type (PT) In general, a single RTP In general, a single RTP only one payload format. only one payload format. RED is an exception. RED is an exception. 26

27 RTP Payload Formats RTP carries the actual digitally encoded voice RTP header + a payload of voice/video samples UDP and IP headers are attached Many voice- and video standards coding standards A payload type identifier in the RTP header A payload type identifier in the RTP header Specified in RFC 1890 and RFC New coding schemes have become available A sender has no idea what coding schemes a receiver could handle Negotiated by signaling protocols like SIP. 27

28 RTP Payload Formats (Cont.) Separate signaling systems Capability negotiation during the call setup SIP and SDP A dynamic payload type may be used Support new coding scheme in the future The encoding name is also significant RED, Redundant payload type RED, Redundant payload type Voice samples + previous samples Cope with packet loss May use different encoding schemes 28

29 SIP signaling : Introduction 3GPP 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:voic @mycompany.com?subject=callme sip: @gateway.mycompany.com tel: DTMFs carried in signaling message 29

30 SIP Components SIP UA Hardware or software running SIP protocols SIP Proxy Ordinary, service provided by ISP Media Gateway Connects different networks SIP-to-PSTN SIP-to-{other VoIP signaling protocol} 30

31 SIP Call Setup 1INVITE From: Proxy for sip.com gets location information for called party. SIP Proxy 2INVITE From: 3OK 200 From: IPv6 Network IP Phone 6Voice/Video packets flow between IP telephones 5ACK IP Phone 4OK 200 From: 31

32 SIP UA Choice 32

33 SIP UA must behave like an ordinary telephone? PC + SIP UA software Windows Messenger CCL + PCA X-Lite It is upset whenever making a phone call after waiting for a computer booted. For a consumer, SIP UA should be easy to use zero-configuration : plug & dial to dial a number to make a phone call as cheaper as possible 33

34 Embedded System for SIP UA CPU & Hardward SoC : ethernet on chip, VGA on chip Easy to extension : PCI, USB Operating System Embedded Linux VxWork Efficiency and system requirement Performance Time-to-prototype and time-to-market Size : easy to be packed into a standalong phone Maintainability 34

35 Protocol stack for embedded system Signaling SIP-SDP CODECs Voice G.711, G.723, G.729 Video H.261, H.263 Transport RTP Application Transport Network Link and Physical User Interface CODECs SIP-SDP RTP TCP/UDP UDP IPv6 2G/3G WIFI XDSL Ethernet 35

36 Application architecture for Embedded system Signaling Network Management Voice Video Voice CODECs Module Video CODECs Module Telephony Signaling Module Module Network Protocol Video, Voice, Signal Packets DSP Module Embedded System 36

37 Application Architecture (cont.) Voice/Video Module Digital-signal processor (DSP) Telephony-Signaling Module Translating signaling into state changes used by the network protocol module to set up connections Network-Management Module Management interface to configure and maintain the other modules Network Protocol Module Processes signaling information and converts it. 37

38 Reference Embedded System Provided by vendor Targeted at potential customers Usually includes Hardware testbed Development software Simulator or emulator Download and bootstrap software Reference implementations 38

39 Reference Software Known as tool chains : installed into host PC Software Purpose C compiler Compile programs for the ADM5120 Assembler Assemble programs for ADM5120 and DSP Downloader Load software into the Embedded System Monitor Communicate with the Embedded system and interact with running software Bootstrap Start the embedded system running Reference Code Example programs for the ADM5120 that show how to implement basic functions 39

40 Reference Hardware Quantity or Size Item 1 Infenion-ADM5120 (175 MHz) 8 Mbytes of SDRAM memory 4 Mbytes of Flash ROM memory 5 10/ 100 Ethernet ports 1 Serial interface (console) 2 MINI-PCI bus interface 1 DSP 40

41 Host Functionalities of host Compiling software for embedded system Software includes OS kernel, device drivers and applications Console of embedded system normally through serial port Show debug messages 41

42 Development Flow Let the OS kernel support IPv6 Configure the OS kernel source to support IPv6 IPv6 Ready Logo Test If failed for IPv6 Ready Logo Test, the network protocol stack in OS kernel should be modified. Let the SIP UP support IPv6 Configure your C library for your embedded system the SIP UA software programming Voice/Video Module Telephony-Signaling Module Network-Management Module Network Protocol Module Put them together Integration test 42

43 Reference System In Our Lab Operating System: Linux SIP, RTP for IPv6 Video Codec H.261, H.263 Audio Codec G.711, G723, G.728, G.729a IPv6 protocol stack IPv6 Phase 1 Ready Logo : Logo ID :

44 Microphone Video Cam LCD Display Signaling 44

45 Future way of communication SIP proxy SIP phone SIP software phone SIP phone IPv6 SIP phone SIP software phone SIP phone SIP proxy 45

46 Documentation References H.323 ITU Standard for Voice/Video over IP SIP Session Initialization Protocol, IETF RFC 2543 RTP Real-time Transport Protocol, IETF RFC 1889 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 46

47 Thank You 47

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