.About Newave
.Products
.PCM CODEC: NW10XX
series
.1034
.1068
.DigitalSwitch Matrix:
NW15XX series
.1500 & 1502
.1506
.FastLane-Voice over IP
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IP - Internet Protocol, which was originally designed to support data
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Current version 4 is the most common in use (RFC 791)
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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)
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Most common telephone system today - analogue
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A very simple implementation
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Keeps the end-to-end delay of voice transmission
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Inexpensive (relatively) when users are not too far apart
BUT
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Other protocols used with the IP :
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Requires one pair of wires per active conversation
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Analogue switches require lot of electromechanical gear which is expensive to buy and maintain
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Parasitic noise adds up at all of the stages of transmission
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For all of the previous reasons - many countries use digital telephone networks.
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In most cases - subscriber line remains analogue. Usually first local exchange converts to digital data stream (64kbps).
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Many voice channels can be multiplexed by
TDM.
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The switch copies the content of one time slot in the incoming transmission to another time slot in the outgoing transmission.
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Conclusion - switching can by done by computers!
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Usually, speak session takes less than half of the time during the conversation (35 percent).
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Data can be sent only when Voice Activity is detected, avoiding transmission of silence. Data is delivers as 'chunks' of data - Packets.
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Advantage - gain of bandwidth.
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Disadvantage - if line transmission is full, need to wait until there is spare capacity.
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The Network delay is called Jitter and must be corrected by the receiving side.
MARKET
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International VOIP calls could cost 1/5th of normal rates T Big share of $18B US to foreign calls. $15B within Europe.
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500,000 IP telephony users at the end of 1995.
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15% of all voice calls on IP/Internet by 2000 ??10M users and $500M in VOIP product sales in 1999
[IDC]
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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
1. Applications - PC to PC
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Need a PC with sound card
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IP Telephony software: Cuseeme, Internet Phone, ...Intranet telephony
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Video optional
2.Applications 每 PC to Phone
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Need more gateways that connect IP network to phone networks
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The IP network could be dedicated intra-net or the Internet.
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The phone networks could be intra-company PBXs or the carrier switches
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※Problem§ - A sequence of packets must be delivered to the recipient in the same sequence as was transmitted by the source
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Protocol is failed if any packets are lost or damaged or any packet is duplicated or packets are received in a different order
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※Basic Solution§ - Packet is transmitted and Recipient acknowledges its receipt
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Packet has a checksum so recipient can detect if packet arrives Okay.
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Once transmitter received the acknowledge - goes on with the next packet
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Defined in RFC 1889
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Provides an End-to-End delivery service for Real-Time traffic applications (Voice, Video)
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The service include :
-- Payload Type identification
-- Sequence numbering
-- Time-Stamping
-- Delivery monitoring
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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)
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Defined in draft RFC 2205
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Reserve network bandwidth for a flow of information to provide QoS to information flow
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Uses a ※flow descriptor§ to request a particular QoS from the network
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Provides the necessary resources in the network to guarantee resources
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TCP - Transmission Control Protocol
(reliable)
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UDP - User Datagram Protocol (non-reliable)
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RTP - Real Time Protocol (Voice data stream
is carried by it)
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RSVP - Resource reSerVation Protocol)
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64kbps (serial) data stream is converted into data packets by a codec
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ITU (International Telecommunication Union) is the international organization that specifies codecs
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ITU evaluates the codec*s performance:
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-- 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.) |
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Significant bandwidth saving can be derived using techniques for voice compression and
silence suppression
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Complexity of delays (Voice Coding,
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Packetization, WAN, Depacketization,
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Network Jitters)
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IP does not provide any QoS and other mechanisms are required to provide them.
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Problem 每 What is a good codec ?
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Voice quality criteria 每 MOS (Mean Opinion Score) is an average mark given by a panel of listeners
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Bit Rate effects the voice quality
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A trade of between the voice quality and bit rate
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Are looked after by the Voice over IP Forum, under the auspices of the International Multimedia Teleconferencing Consortium
(IMTC)
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Based upon ITU-T Recommendation H.323.
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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
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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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