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     .1506

 .FastLane-Voice over IP


  • What is IP ?
  • IP - Internet Protocol, which was originally designed to support data

  • Current version 4 is the most common in use (RFC 791)

  • Other protocols used with the IP :
    -- TCP - Transmission Control Protocol (reliable)
    -- UDP - User Datagram Protocol (non-reliable)
    -- RTP - Real Time Protocol (RFC 1889)
    -- RSVP - Resource reSerVation Protocol)

  • Voice Over IP - Why?

  • Most common telephone system today - analogue

  • A very simple implementation

  • Keeps the end-to-end delay of voice transmission

  • Inexpensive (relatively) when users are not too far apart

BUT

  • Other protocols used with the IP :

  • Requires one pair of wires per active conversation

  • Analogue switches require lot of electromechanical gear which is expensive to buy and maintain

  • Parasitic noise adds up at all of the stages of transmission

  • For all of the previous reasons - many countries use digital telephone networks.

  • In most cases - subscriber line remains analogue. Usually first local exchange converts to digital data stream (64kbps).

  • Many voice channels can be multiplexed by TDM.

  • The switch copies the content of one time slot in the incoming transmission to another time slot in the outgoing transmission.

  • Conclusion - switching can by done by computers!

  • Usually, speak session takes less than half of the time during the conversation (35 percent). 

  • Data can be sent only when Voice Activity is detected, avoiding transmission of silence. Data is delivers as 'chunks' of data - Packets.

  • Advantage - gain of bandwidth.

  • Disadvantage - if line transmission is full, need to wait until there is spare capacity.

  •  The Network delay is called Jitter and must be corrected by the receiving side.

MARKET

  • International VOIP calls could cost 1/5th of normal rates T Big share of $18B US to foreign calls. $15B within Europe.

  • 500,000 IP telephony users at the end of 1995. 

  • 15% of all voice calls on IP/Internet by 2000 ??10M users and $500M in VOIP product sales in 1999 [IDC] 

  • US VOIP service will grow from $30M in 1998 to $2B in 2004 [Forester Research], $2B in 2001 and $16B by 2004 [Frost & Sullivan]

Applications

  • Any voice communication where PC is already used:

  • Helpdesk access

  • On-line order placement

  • International callbacks (many operators use voice over frame relay). 

  • Intranet telephony

  • Internet fax

1. Applications - PC to PC

  • Need a PC with sound card

  • IP Telephony software: Cuseeme, Internet Phone, ...Intranet telephony

  • Video optional

2.Applications 每 PC to Phone

  • Need more gateways that connect IP network to phone networks

  • The IP network could be dedicated intra-net or the Internet.

  • The phone networks could be intra-company PBXs or the carrier switches

  • ※Problem§ - A sequence of packets must be delivered to the recipient in the same sequence as was transmitted by the source

  • Protocol is failed if any packets are lost or damaged or any packet is duplicated or packets are received in a different order

  • ※Basic Solution§ - Packet is transmitted and Recipient acknowledges its receipt

  • Packet has a checksum so recipient can detect if packet arrives Okay.

  • Once transmitter received the acknowledge - goes on with the next packet

  • Defined in RFC 1889

  • Provides an End-to-End delivery service for Real-Time traffic applications (Voice, Video)

  • The service include :
    -- Payload Type identification
    -- Sequence numbering
    -- Time-Stamping
    -- Delivery monitoring

  • RTCP - Real-Time Transport Control Protocol which is used to provide feedback on the quality of the received data on the connection (among other things)

  • Defined in draft RFC 2205

  • Reserve network bandwidth for a flow of information to provide QoS to information flow

  • Uses a ※flow descriptor§ to request a particular QoS from the network

  • Provides the necessary resources in the network to guarantee resources

  • TCP - Transmission Control Protocol (reliable)

  • UDP - User Datagram Protocol (non-reliable)

  • RTP - Real Time Protocol (Voice data stream is carried by it)

  • RSVP - Resource reSerVation Protocol)

  • 64kbps (serial) data stream is converted into data packets by a codec

  • ITU (International Telecommunication Union) is the international organization that specifies codecs

  • ITU evaluates the codec*s performance:

-- Quality ? G.726 @ 32kbps
-- Constant quality for man and women in several languages
-- Ability to consider a background noise and recreate is correctly
-- DTMF transparently
-- Should be in use over non-reliable medium (radio link, etc.)

  • Significant bandwidth saving can be derived using techniques for voice compression and silence suppression

  • Complexity of delays (Voice Coding,

  • Packetization, WAN, Depacketization,

  • Network Jitters)

  • IP does not provide any QoS and other mechanisms are required to provide them.

  • Problem 每 What is a good codec ?

  • Voice quality criteria 每 MOS (Mean Opinion Score) is an average mark given by a panel of listeners

  • Bit Rate effects the voice quality 

  • A trade of between the voice quality and bit rate

  • Are looked after by the Voice over IP Forum, under the auspices of the International Multimedia Teleconferencing Consortium (IMTC)

  • Based upon ITU-T Recommendation H.323.

  • H.323 defines the support of five voice coders:
    -- G.711 每 64kps, MOS=4.2
    -- G.722 每 48, 56, 64kps, MOS=4.5 (audio spectrum 7khz), 10MIPS 
    -- G.723.1每 5.3 / 6.4kps, MOS=3.7 / 3.9, 16MIPS (default for IT)
    -- G.728 每 16kps, MOS=4.3, in use for H.320 (video conferencing) 
    -- G.729 / A 每 8kps, MOS=4.0, 20(3 for decoding / 10.5 (2 for decoding) MIPS

  • H.323 signalling & gatekeeper interoperation
  • G3 FAX modem support
  • T.38 FAX protocol
  • Echo cancellation

  • Enterprise use
    -- Limited as need to have access to copper pairs
  • Carrier/ISP use
    -- Termination of voice and data services (also called an IAD - Integrated Access device)

VHTU Voice HTU Application
  • Voice Ports
    4 x Analog Voice (FXS,FXO,E&M) or
    1 x E1 Digital (4 channels or 8 channels)
  • Data Ports
    1 x Sync 
    No Ethernet (as don*t want to be a router)
  • WAN Port
    1 x DSL (LDSL or MDSL)
  • Management
    Console + Telnet

Carrier IP VPN over xDSL

  • VHTU Specification
    -- Use HTU for
    -- Connecting carrier or enterprise provided router to VPN
  • Use VoIP HTU for
    -- Connecting router and voice services to VPN

VHTU Functions

  • As for Enterprise VoIP
  • Option 1 TDM link
    -- Voice and FAX converted to IP traffic 
    -- IP data over HDLC
    -- Separate Voice and Data TDM timeslots over xDSL
  • Option 2 Frame relay link
    -- Voice/FAX IP packets over FR
    -- Set external router to FR protocol
    -- Internal FR switch 
    -- FR over xDSL

VOIP Jabiru Cards

  • SPC 每 Synch Port Card (6 X V35)
  • JPP Packet processor card 
    -- FR switch to high speed serial port (4M)

     


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